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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for MPEG-4 Audio/Visual Streams
	Author(s)       : M. Schmidt, et al.
	Filename        : draft-ietf-avt-rfc3016bis-01.txt
	Pages           : 32
	Date            : 2010-10-01

This document describes Real-Time Transport Protocol (RTP) payload
formats for carrying each of MPEG-4 Audio and MPEG-4 Visual
bitstreams without using MPEG-4 Systems.  For the purpose of directly
mapping MPEG-4 Audio/Visual bitstreams onto RTP packets, it provides
specifications for the use of RTP header fields and also specifies
fragmentation rules.  It also provides specifications for Media Type
registration and the use of Session Description Protocol (SDP).

Comments are solicited and should be addressed to the working group's
mailing list at avt@ietf.org and/or the author(s).

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http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc3016bis-01.txt

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From frans.de.bont@philips.com  Fri Oct  1 01:54:42 2010
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From: "Bont, Frans de" <frans.de.bont@philips.com>
To: "avt@ietf.org" <avt@ietf.org>
Date: Fri, 1 Oct 2010 10:55:23 +0200
Thread-Topic: [AVT] I-D Action:draft-ietf-avt-rfc3016bis-01.txt
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Subject: Re: [AVT] I-D Action:draft-ietf-avt-rfc3016bis-01.txt
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Hi all,

We have submitted a slightly updated version of the draft as it would expir=
e soon.

Below a summary of the changes.

http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc3016bis-01.txt

1. The media subtype registration was not in line in accordance with
RFC4855 and was not using the template of RFC4288. This has been corrected.

2. The registration section had examples, they have been moved to the mappi=
ng
to SDP section.

3. The IANA consideration was updating the registering of the media subtype
definition from RFC 3016, but since it is the intention to obsolete 3016,
RFC 3016 has been replaced by RFC XXXX in the related sections.

4. The draft listed the differences with respect to RFC 3016, but an
interoperability consideration with systems that support RFC 3016 was missi=
ng.
Therefore, a section has been added to describe that no interoperability is=
sues
are foreseen because existing systems that support RFC 3016 already comply =
with
the specification of the 3GPP PSS service and the draft is compatible with =
it.

5. In section 3.3 (f) an example of fragmentation was mentioned but the las=
t
sentence of that example was confusing. This last sentence has been removed=
.

6. The change controller in the registration is changed to:
"IETF Audio/Video Transport working group delegated from the IESG."

Your comments are welcome!

Best regards,
Frans

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Internet-Drafts@ietf.org
> Sent: 2010 Oct 01 10:45 AM
> To: i-d-announce@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] I-D Action:draft-ietf-avt-rfc3016bis-01.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Audio/Video Transport Working Group of t=
he
> IETF.
>
>
>       Title           : RTP Payload Format for MPEG-4 Audio/Visual Stream=
s
>       Author(s)       : M. Schmidt, et al.
>       Filename        : draft-ietf-avt-rfc3016bis-01.txt
>       Pages           : 32
>       Date            : 2010-10-01
>
> This document describes Real-Time Transport Protocol (RTP) payload format=
s for
> carrying each of MPEG-4 Audio and MPEG-4 Visual bitstreams without using =
MPEG-
> 4 Systems.  For the purpose of directly mapping MPEG-4 Audio/Visual bitst=
reams
> onto RTP packets, it provides specifications for the use of RTP header fi=
elds
> and also specifies fragmentation rules.  It also provides specifications =
for
> Media Type registration and the use of Session Description Protocol (SDP)=
.
>
> Comments are solicited and should be addressed to the working group's mai=
ling
> list at avt@ietf.org and/or the author(s).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc3016bis-01.txt
>
> Internet-Drafts are also available by anonymous FTP at:
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>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the Interne=
t-
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From Even.roni@huawei.com  Fri Oct  1 13:51:03 2010
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Date: Fri, 01 Oct 2010 22:49:25 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] rfc3984bis - max-smbps limit
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Hi,

We got the following comment as a private email (bellow) . The draft has =
not
be published yet and we can do this change before publication.
Please review and send comments to the list since it changes the current
semantics. If there are no objections we will do this change.
Since this parameter has been added in the bis draft and was not part of =
RFC
3984 we do not see any interoperability issues.

Thanks
Roni Even
As a co-author of the draft


Email from Tom Kristensen

I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps =
as
defined in draft-ietf-avt-rtp-rfc3984bis.

 From Table 8-9, H.241:
---
The encoder should recompute this value for each picture.
The value of (MaxStaticMBPS =AA 500) shall not be less than the value =
MaxMBPS
for the Level given in Table A-1/H.264, and if CustomMaxMBPS is =
signalled,
shall not be less than the value (CustomMaxMBPS =AA 500).
---

On the other hand in the SIP world, rfc3984bis says:
---
The value of max-smbps MUST be greater than the value of MaxMBPS given =
in
Table A-1 of [1] for the highest level.
---
http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44


H.241 says      max-smbps >=3D max-mbps.
rfc3984bis says max-smbps >  max-mbps.
We are missing "or equal".

Of course it doesn't really make much meaning to include a max-smbps=20
equal to max-mbps, but we have a mismatch between H.241 and the SDP=20
counterpart.

Is it possible and feasible to alter the text given the current state of =

soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?



From stephen.botzko@gmail.com  Fri Oct  1 14:02:45 2010
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Roni Even <Even.roni@huawei.com>
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Cc: Randell Jesup <rjesup@wgate.com>, keith.drage@alcatel-lucent.com, Tom Kristensen <tom.kristensen@tandberg.com>, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] rfc3984bis - max-smbps limit
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--0016364ee23cfc25ef049194868c
Content-Type: text/plain; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I think reconciling the text with H.241 is a good idea, as it removes a
corner-case from fully compliant non-transcoding gateways.

I also agree with Tom's observation that it doesn't make sense to signal
max-smbps unless it is greater than max_mpbs.

Stephen Botzko

2010/10/1 Roni Even <Even.roni@huawei.com>

> Hi,
>
> We got the following comment as a private email (bellow) . The draft has
> not
> be published yet and we can do this change before publication.
> Please review and send comments to the list since it changes the current
> semantics. If there are no objections we will do this change.
> Since this parameter has been added in the bis draft and was not part of
> RFC
> 3984 we do not see any interoperability issues.
>
> Thanks
> Roni Even
> As a co-author of the draft
>
>
> Email from Tom Kristensen
>
> I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps =
as
> defined in draft-ietf-avt-rtp-rfc3984bis.
>
>  From Table 8-9, H.241:
> ---
> The encoder should recompute this value for each picture.
> The value of (MaxStaticMBPS =D7 500) shall not be less than the value Max=
MBPS
> for the Level given in Table A-1/H.264, and if CustomMaxMBPS is signalled=
,
> shall not be less than the value (CustomMaxMBPS =D7 500).
> ---
>
> On the other hand in the SIP world, rfc3984bis says:
> ---
> The value of max-smbps MUST be greater than the value of MaxMBPS given in
> Table A-1 of [1] for the highest level.
> ---
> http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44
>
>
> H.241 says      max-smbps >=3D max-mbps.
> rfc3984bis says max-smbps >  max-mbps.
> We are missing "or equal".
>
> Of course it doesn't really make much meaning to include a max-smbps
> equal to max-mbps, but we have a mismatch between H.241 and the SDP
> counterpart.
>
> Is it possible and feasible to alter the text given the current state of
> soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--0016364ee23cfc25ef049194868c
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I think reconciling the text with H.241 is a good idea, as it removes a cor=
ner-case from fully compliant non-transcoding gateways.<br><br>I also agree=
 with Tom&#39;s observation that it doesn&#39;t make sense to signal max-sm=
bps unless it is greater than max_mpbs.<br>
<br>Stephen Botzko<br><br><div class=3D"gmail_quote">2010/10/1 Roni Even <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:Even.roni@huawei.com">Even.roni@huawe=
i.com</a>&gt;</span><br><blockquote class=3D"gmail_quote" style=3D"margin: =
0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left:=
 1ex;">
Hi,<br>
<br>
We got the following comment as a private email (bellow) . The draft has no=
t<br>
be published yet and we can do this change before publication.<br>
Please review and send comments to the list since it changes the current<br=
>
semantics. If there are no objections we will do this change.<br>
Since this parameter has been added in the bis draft and was not part of RF=
C<br>
3984 we do not see any interoperability issues.<br>
<br>
Thanks<br>
Roni Even<br>
As a co-author of the draft<br>
<br>
<br>
Email from Tom Kristensen<br>
<br>
I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps as=
<br>
defined in draft-ietf-avt-rtp-rfc3984bis.<br>
<br>
=A0From Table 8-9, H.241:<br>
---<br>
The encoder should recompute this value for each picture.<br>
The value of (MaxStaticMBPS =D7 500) shall not be less than the value MaxMB=
PS<br>
for the Level given in Table A-1/H.264, and if CustomMaxMBPS is signalled,<=
br>
shall not be less than the value (CustomMaxMBPS =D7 500).<br>
---<br>
<br>
On the other hand in the SIP world, rfc3984bis says:<br>
---<br>
The value of max-smbps MUST be greater than the value of MaxMBPS given in<b=
r>
Table A-1 of [1] for the highest level.<br>
---<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page=
-44" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc398=
4bis-11#page-44</a><br>
<br>
<br>
H.241 says =A0 =A0 =A0max-smbps &gt;=3D max-mbps.<br>
rfc3984bis says max-smbps &gt; =A0max-mbps.<br>
We are missing &quot;or equal&quot;.<br>
<br>
Of course it doesn&#39;t really make much meaning to include a max-smbps<br=
>
equal to max-mbps, but we have a mismatch between H.241 and the SDP<br>
counterpart.<br>
<br>
Is it possible and feasible to alter the text given the current state of<br=
>
soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?<br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br>

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From yekuiwang@huawei.com  Fri Oct  1 14:05:49 2010
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Date: Fri, 01 Oct 2010 17:06:25 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
In-reply-to: <AANLkTi=ASeg-7srNyTMmqik_j=jCcYZ9qk-mkqo3LeBb@mail.gmail.com>
To: 'Stephen Botzko' <stephen.botzko@gmail.com>, 'Roni Even' <Even.roni@huawei.com>
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Subject: Re: [AVT] rfc3984bis - max-smbps limit
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+1


  _____ =20

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Stephen Botzko
Sent: Friday, October 01, 2010 5:04 PM
To: Roni Even
Cc: Randell Jesup; keith.drage@alcatel-lucent.com; Tom Kristensen; IETF =
AVT
WG
Subject: Re: [AVT] rfc3984bis - max-smbps limit


I think reconciling the text with H.241 is a good idea, as it removes a
corner-case from fully compliant non-transcoding gateways.

I also agree with Tom's observation that it doesn't make sense to signal
max-smbps unless it is greater than max_mpbs.

Stephen Botzko


2010/10/1 Roni Even <Even.roni@huawei.com>


Hi,

We got the following comment as a private email (bellow) . The draft has =
not
be published yet and we can do this change before publication.
Please review and send comments to the list since it changes the current
semantics. If there are no objections we will do this change.
Since this parameter has been added in the bis draft and was not part of =
RFC
3984 we do not see any interoperability issues.

Thanks
Roni Even
As a co-author of the draft


Email from Tom Kristensen

I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps =
as
defined in draft-ietf-avt-rtp-rfc3984bis.

 From Table 8-9, H.241:
---
The encoder should recompute this value for each picture.
The value of (MaxStaticMBPS =D7 500) shall not be less than the value =
MaxMBPS
for the Level given in Table A-1/H.264, and if CustomMaxMBPS is =
signalled,
shall not be less than the value (CustomMaxMBPS =D7 500).
---

On the other hand in the SIP world, rfc3984bis says:
---
The value of max-smbps MUST be greater than the value of MaxMBPS given =
in
Table A-1 of [1] for the highest level.
---
http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44


H.241 says      max-smbps >=3D max-mbps.
rfc3984bis says max-smbps >  max-mbps.
We are missing "or equal".

Of course it doesn't really make much meaning to include a max-smbps
equal to max-mbps, but we have a mismatch between H.241 and the SDP
counterpart.

Is it possible and feasible to alter the text given the current state of
soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt




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<DIV dir=3Dltr align=3Dleft><SPAN class=3D910030621-01102010><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>+1</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Stephen=20
  Botzko<BR><B>Sent:</B> Friday, October 01, 2010 5:04 PM<BR><B>To:</B> =
Roni=20
  Even<BR><B>Cc:</B> Randell Jesup; keith.drage@alcatel-lucent.com; Tom=20
  Kristensen; IETF AVT WG<BR><B>Subject:</B> Re: [AVT] rfc3984bis - =
max-smbps=20
  limit<BR></FONT><BR></DIV>
  <DIV></DIV>I think reconciling the text with H.241 is a good idea, as =
it=20
  removes a corner-case from fully compliant non-transcoding =
gateways.<BR><BR>I=20
  also agree with Tom's observation that it doesn't make sense to signal =

  max-smbps unless it is greater than max_mpbs.<BR><BR>Stephen =
Botzko<BR><BR>
  <DIV class=3Dgmail_quote>2010/10/1 Roni Even <SPAN dir=3Dltr>&lt;<A=20
  =
href=3D"mailto:Even.roni@huawei.com">Even.roni@huawei.com</A>&gt;</SPAN><=
BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">Hi,<BR><BR>We=20
    got the following comment as a private email (bellow) . The draft =
has=20
    not<BR>be published yet and we can do this change before=20
    publication.<BR>Please review and send comments to the list since it =
changes=20
    the current<BR>semantics. If there are no objections we will do this =

    change.<BR>Since this parameter has been added in the bis draft and =
was not=20
    part of RFC<BR>3984 we do not see any interoperability=20
    issues.<BR><BR>Thanks<BR>Roni Even<BR>As a co-author of the=20
    draft<BR><BR><BR>Email from Tom Kristensen<BR><BR>I discovered =
missing=20
    alignment between H.241 MaxStaticMBPS and max-smbps as<BR>defined in =

    draft-ietf-avt-rtp-rfc3984bis.<BR><BR>&nbsp;From Table 8-9,=20
    H.241:<BR>---<BR>The encoder should recompute this value for each=20
    picture.<BR>The value of (MaxStaticMBPS =D7 500) shall not be less =
than the=20
    value MaxMBPS<BR>for the Level given in Table A-1/H.264, and if=20
    CustomMaxMBPS is signalled,<BR>shall not be less than the value=20
    (CustomMaxMBPS =D7 500).<BR>---<BR><BR>On the other hand in the SIP =
world,=20
    rfc3984bis says:<BR>---<BR>The value of max-smbps MUST be greater =
than the=20
    value of MaxMBPS given in<BR>Table A-1 of [1] for the highest=20
    level.<BR>---<BR><A=20
    =
href=3D"http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-=
44"=20
    =
target=3D_blank>http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-=
11#page-44</A><BR><BR><BR>H.241=20
    says &nbsp; &nbsp; &nbsp;max-smbps &gt;=3D max-mbps.<BR>rfc3984bis =
says=20
    max-smbps &gt; &nbsp;max-mbps.<BR>We are missing "or =
equal".<BR><BR>Of=20
    course it doesn't really make much meaning to include a =
max-smbps<BR>equal=20
    to max-mbps, but we have a mismatch between H.241 and the=20
    SDP<BR>counterpart.<BR><BR>Is it possible and feasible to alter the =
text=20
    given the current state of<BR>soon-to-be-RFC=20
    =
draft-ietf-avt-rtp-rfc3984bis?<BR><BR><BR>_______________________________=
________________<BR>Audio/Video=20
    Transport Working Group<BR><A=20
    href=3D"mailto:avt@ietf.org">avt@ietf.org</A><BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/avt"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/avt</A><BR></BLOCKQ=
UOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>

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From eckelcu@cisco.com  Fri Oct  1 14:26:31 2010
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References: <4CA5DCCC.3020105@tandberg.com><015f01cb61aa$29ef8a10$7dce9e30$%roni@huawei.com><AANLkTi=ASeg-7srNyTMmqik_j=jCcYZ9qk-mkqo3LeBb@mail.gmail.com> <0E1BEE7BB93D42C9AD14BD04A8D6A9D6@china.huawei.com>
From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Ye-Kui Wang" <yekuiwang@huawei.com>, "Stephen Botzko" <stephen.botzko@gmail.com>, "Roni Even" <Even.roni@huawei.com>
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Subject: Re: [AVT] rfc3984bis - max-smbps limit
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I agree, and have seen this come up with H.323/SIP interworking =
gateways. I think the change is worthwhile.

Charles

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Ye-Kui Wang
> Sent: Friday, October 01, 2010 2:06 PM
> To: 'Stephen Botzko'; 'Roni Even'
> Cc: 'Randell Jesup'; keith.drage@alcatel-lucent.com; 'Tom Kristensen'; =
'IETF AVT WG'
> Subject: Re: [AVT] rfc3984bis - max-smbps limit
>=20
> +1
>=20
>=20
> ________________________________
>=20
> 	From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Stephen Botzko
> 	Sent: Friday, October 01, 2010 5:04 PM
> 	To: Roni Even
> 	Cc: Randell Jesup; keith.drage@alcatel-lucent.com; Tom Kristensen; =
IETF AVT WG
> 	Subject: Re: [AVT] rfc3984bis - max-smbps limit
>=20
>=20
> 	I think reconciling the text with H.241 is a good idea, as it removes =
a corner-case from fully
> compliant non-transcoding gateways.
>=20
> 	I also agree with Tom's observation that it doesn't make sense to =
signal max-smbps unless it is
> greater than max_mpbs.
>=20
> 	Stephen Botzko
>=20
>=20
> 	2010/10/1 Roni Even <Even.roni@huawei.com>
>=20
>=20
> 		Hi,
>=20
> 		We got the following comment as a private email (bellow) . The draft =
has not
> 		be published yet and we can do this change before publication.
> 		Please review and send comments to the list since it changes the =
current
> 		semantics. If there are no objections we will do this change.
> 		Since this parameter has been added in the bis draft and was not =
part of RFC
> 		3984 we do not see any interoperability issues.
>=20
> 		Thanks
> 		Roni Even
> 		As a co-author of the draft
>=20
>=20
> 		Email from Tom Kristensen
>=20
> 		I discovered missing alignment between H.241 MaxStaticMBPS and =
max-smbps as
> 		defined in draft-ietf-avt-rtp-rfc3984bis.
>=20
> 		 From Table 8-9, H.241:
> 		---
> 		The encoder should recompute this value for each picture.
> 		The value of (MaxStaticMBPS =D7 500) shall not be less than the =
value MaxMBPS
> 		for the Level given in Table A-1/H.264, and if CustomMaxMBPS is =
signalled,
> 		shall not be less than the value (CustomMaxMBPS =D7 500).
> 		---
>=20
> 		On the other hand in the SIP world, rfc3984bis says:
> 		---
> 		The value of max-smbps MUST be greater than the value of MaxMBPS =
given in
> 		Table A-1 of [1] for the highest level.
> 		---
> 		http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44
>=20
>=20
> 		H.241 says      max-smbps >=3D max-mbps.
> 		rfc3984bis says max-smbps >  max-mbps.
> 		We are missing "or equal".
>=20
> 		Of course it doesn't really make much meaning to include a max-smbps
> 		equal to max-mbps, but we have a mismatch between H.241 and the SDP
> 		counterpart.
>=20
> 		Is it possible and feasible to alter the text given the current =
state of
> 		soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?
>=20
>=20
> 		_______________________________________________
> 		Audio/Video Transport Working Group
> 		avt@ietf.org
> 		https://www.ietf.org/mailman/listinfo/avt
>=20
>=20


From mzanaty@cisco.com  Sat Oct  2 06:08:46 2010
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Thread-Topic: [AVT] rfc3984bis - max-smbps limit
Thread-Index: ActhaXmd2eNVMAcqToWvhL7+ejvpbgAPaDZAACFimyA=
From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Roni Even" <Even.roni@huawei.com>, "IETF AVT WG" <avt@ietf.org>
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Cc: Randell Jesup <rjesup@wgate.com>, Tom Kristensen <tom.kristensen@tandberg.com>, keith.drage@alcatel-lucent.com
Subject: Re: [AVT] rfc3984bis - max-smbps limit
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The draft is missing more than just "or equal to".

> The value of max-smbps MUST be greater than the value of MaxMBPS given =
in
> Table A-1 of [1] for the highest level.

It should also take max-mbps into account. And the reference to "highest =
level" has no antecedent in this section; it is in the prior section on =
max-mbps.

I suggest this [new] text:

"The value of max-smbps MUST be greater than [or equal to] the value of =
MaxMBPS given [explicitly in max-mbps or implicitly] in Table A-1 of [1] =
for the highest level [given in profile-level-id or max-recv-level]."

Regards,
Mo

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Roni Even
Sent: Friday, October 01, 2010 4:49 PM
To: 'IETF AVT WG'
Cc: 'Randell Jesup'; 'Tom Kristensen'; keith.drage@alcatel-lucent.com
Subject: Re: [AVT] rfc3984bis - max-smbps limit

Hi,

We got the following comment as a private email (bellow) . The draft has =
not
be published yet and we can do this change before publication.
Please review and send comments to the list since it changes the current
semantics. If there are no objections we will do this change.
Since this parameter has been added in the bis draft and was not part of =
RFC
3984 we do not see any interoperability issues.

Thanks
Roni Even
As a co-author of the draft


Email from Tom Kristensen

I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps =
as
defined in draft-ietf-avt-rtp-rfc3984bis.

 From Table 8-9, H.241:
---
The encoder should recompute this value for each picture.
The value of (MaxStaticMBPS =D7 500) shall not be less than the value =
MaxMBPS
for the Level given in Table A-1/H.264, and if CustomMaxMBPS is =
signalled,
shall not be less than the value (CustomMaxMBPS =D7 500).
---

On the other hand in the SIP world, rfc3984bis says:
---
The value of max-smbps MUST be greater than the value of MaxMBPS given =
in
Table A-1 of [1] for the highest level.
---
http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44


H.241 says      max-smbps >=3D max-mbps.
rfc3984bis says max-smbps >  max-mbps.
We are missing "or equal".

Of course it doesn't really make much meaning to include a max-smbps=20
equal to max-mbps, but we have a mismatch between H.241 and the SDP=20
counterpart.

Is it possible and feasible to alter the text given the current state of =

soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

From dworley@avaya.com  Fri Oct  1 08:53:21 2010
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: =?iso-8859-1?Q?Gunnar_Hellstr=F6m?= <gunnar.hellstrom@omnitor.se>, "avt@ietf.org" <avt@ietf.org>, Robert Sparks <rjsparks@nostrum.com>
Date: Fri, 1 Oct 2010 11:54:05 -0400
Thread-Topic: [AVT] Report on Errata ID: 1793, RFC4103, "RTP Payload for Text Conversation"
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________________________________________
From: Gunnar Hellstr=F6m [gunnar.hellstrom@omnitor.se]

You say " if one is carefully examining the example, one can see that it
is incorrect based on RFC 2198."
Can you please explain what inconsistency you have found?
________________________________________

In the scenario discussed in RFC 4103 section 7.1, the example message on p=
age 12 contains two T140blocks, one "primary" (P), that is, the most recent=
, containing zero bytes, and one "redundant" (R), which is a copy of the im=
mediately preceding T140block to be sent, containing some bytes.  The examp=
le message on page 13 (which is the subject of the erratum) is the next RTP=
 packet sent, and is to contain a primary T140block "P", the next-previous =
T140block "R1", and the one before that, "R2".

Thus, the various contained T140blocks are duplicates in the following way:

(must be seen in fixed-width)

    RTP packet, p. 13    RTP packet, p. 12

    P

    R1 ----------------- P

    R2 ----------------- R

The p. 13 P is the newest T140block and is stated to be non-empty.  The p. =
13 R1 is the same as the p. 12 P and is stated on p. 12 to be empty.  The p=
. 13 R2 is the same as the p. 12 R and is stated on p. 12 to be non-empty. =
 But the RFC's diagram on p. 13 says that R2 is empty and R1 is not.  Corre=
cting that statement is the substance of the erratum.

Dale


Worley, Dale R (Dale) skrev 2010-09-30 20:54:
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Errata ID: 1793
> RFC4103, "RTP Payload for Text Conversation"
> Source of RFC: avt (rai)
>
> Status: Reported
> Type: Technical
>
> Reported By: Gunnar Hellstrom
> Date Reported: 2009-06-16
>
> Section 7.1 says:
>
> The value of the "R2 block length" would be set to zero in order to
> represent the empty T140block.
>
>    0                   1                   2                   3
>      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |V=3D2|P|X| CC=3D0  |M|  "RED" PT   |   sequence number of primary  |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |               timestamp of primary encoding "P"               |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |           synchronization source (SSRC) identifier            |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |1|   T140 PT   |  timestamp offset of "R2" | "R2" block length |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |1|   T140 PT   |  timestamp offset of "R1" | "R1" block length |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |0|   T140 PT   | "R1" T.140 encoded redundant data             |
>     +-+-+-+-+-+-+-+-+                               +---------------+
>     |                                               |               |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+         +-+-+-+
>     |              "P" T.140 encoded primary data             |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>
> It should say:
>
> The value of the "R1 block length" would be set to zero in order to
> represent the empty T140block.
>
>    0                   1                   2                   3
>      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |V=3D2|P|X| CC=3D0  |M|  "RED" PT   |   sequence number of primary  |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |               timestamp of primary encoding "P"               |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |           synchronization source (SSRC) identifier            |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |1|   T140 PT   |  timestamp offset of "R2" | "R2" block length |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |1|   T140 PT   |  timestamp offset of "R1" | "R1" block length |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>     |0|   T140 PT   | "R2" T.140 encoded redundant data             |
>     +-+-+-+-+-+-+-+-+                               +---------------+
>     |                                               |               |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+         +-+-+-+
>     |              "P" T.140 encoded primary data             |
>     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
> ----------------------------------------------------------------------
> Recommended status:  Verified, hold for document update
>
> The erratum is correct.  In the example in question (on page 13), the
> three T140blocks (in time order) are R2, R1, and P, the first two of
> which correspond to R and P of the previous example (on page 12).  The
> latter P has zero length.  So in the example in question, R2 is
> non-empty, R1 is empty, and P is non-empty.
>
> However, if one is carefully examining the example, one can see that
> it is incorrect based on RFC 2198.  So I recommend that this erratum
> be held for document update.
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
> Dale
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From tom.kristensen@tandberg.com  Sun Oct  3 02:48:28 2010
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Cc: Randell Jesup <rjesup@wgate.com>, keith.drage@alcatel-lucent.com, IETF AVT WG <avt@ietf.org>, Roni Even <Even.roni@huawei.com>
Subject: Re: [AVT] rfc3984bis - max-smbps limit
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Indeed, Mo, we have to take a potential max-mbps into account in the 
description and the clarification of "highest level" also in the 
max-smbps text part is just fine.

Even though "highest level" is described under a common heading for 
these parameters, it is repeated for the other parameters.
Cf.:
        max-mbps, max-smbps, max-fs, max-cpb, max-dpb, and max-br:
           These parameters MAY be used to signal the capabilities of a
           receiver implementation. These parameters MUST NOT be used
           for any other purpose.  The highest level conveyed in the
           value of the profile-level-id parameter or the max-recv-level
           parameter MUST be such that the receiver is fully capable of
           supporting.

-- Tom

On 10/02/2010 03:08 PM, Mo Zanaty (mzanaty) wrote:
> The draft is missing more than just "or equal to".
>
>> The value of max-smbps MUST be greater than the value of MaxMBPS given in
>> Table A-1 of [1] for the highest level.
>
> It should also take max-mbps into account. And the reference to "highest level" has no antecedent in this section; it is in the prior section on max-mbps.
>
> I suggest this [new] text:
>
> "The value of max-smbps MUST be greater than [or equal to] the value of MaxMBPS given [explicitly in max-mbps or implicitly] in Table A-1 of [1] for the highest level [given in profile-level-id or max-recv-level]."
>
> Regards,
> Mo
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Roni Even
> Sent: Friday, October 01, 2010 4:49 PM
> To: 'IETF AVT WG'
> Cc: 'Randell Jesup'; 'Tom Kristensen'; keith.drage@alcatel-lucent.com
> Subject: Re: [AVT] rfc3984bis - max-smbps limit
>
> Hi,
>
> We got the following comment as a private email (bellow) . The draft has not
> be published yet and we can do this change before publication.
> Please review and send comments to the list since it changes the current
> semantics. If there are no objections we will do this change.
> Since this parameter has been added in the bis draft and was not part of RFC
> 3984 we do not see any interoperability issues.
>
> Thanks
> Roni Even
> As a co-author of the draft
>
>
> Email from Tom Kristensen
>
> I discovered missing alignment between H.241 MaxStaticMBPS and max-smbps as
> defined in draft-ietf-avt-rtp-rfc3984bis.
>
>   From Table 8-9, H.241:
> ---
> The encoder should recompute this value for each picture.
> The value of (MaxStaticMBPS × 500) shall not be less than the value MaxMBPS
> for the Level given in Table A-1/H.264, and if CustomMaxMBPS is signalled,
> shall not be less than the value (CustomMaxMBPS × 500).
> ---
>
> On the other hand in the SIP world, rfc3984bis says:
> ---
> The value of max-smbps MUST be greater than the value of MaxMBPS given in
> Table A-1 of [1] for the highest level.
> ---
> http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44
>
>
> H.241 says      max-smbps>= max-mbps.
> rfc3984bis says max-smbps>   max-mbps.
> We are missing "or equal".
>
> Of course it doesn't really make much meaning to include a max-smbps
> equal to max-mbps, but we have a mismatch between H.241 and the SDP
> counterpart.
>
> Is it possible and feasible to alter the text given the current state of
> soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From Even.roni@huawei.com  Sun Oct  3 05:48:10 2010
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Date: Sun, 03 Oct 2010 14:46:32 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'Tom Kristensen' <tom.kristensen@tandberg.com>, "'Mo Zanaty (mzanaty)'" <mzanaty@cisco.com>
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Cc: 'Randell Jesup' <rjesup@wgate.com>, keith.drage@alcatel-lucent.com, 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] rfc3984bis - max-smbps limit
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Tom,
Can you suggest a final text so that we can ask the AD what to do
Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Tom Kristensen
> Sent: Sunday, October 03, 2010 11:49 AM
> To: Mo Zanaty (mzanaty)
> Cc: Randell Jesup; keith.drage@alcatel-lucent.com; IETF AVT WG; Roni
> Even
> Subject: Re: [AVT] rfc3984bis - max-smbps limit
>=20
> Indeed, Mo, we have to take a potential max-mbps into account in the
> description and the clarification of "highest level" also in the
> max-smbps text part is just fine.
>=20
> Even though "highest level" is described under a common heading for
> these parameters, it is repeated for the other parameters.
> Cf.:
>         max-mbps, max-smbps, max-fs, max-cpb, max-dpb, and max-br:
>            These parameters MAY be used to signal the capabilities of =
a
>            receiver implementation. These parameters MUST NOT be used
>            for any other purpose.  The highest level conveyed in the
>            value of the profile-level-id parameter or the max-recv-
> level
>            parameter MUST be such that the receiver is fully capable =
of
>            supporting.
>=20
> -- Tom
>=20
> On 10/02/2010 03:08 PM, Mo Zanaty (mzanaty) wrote:
> > The draft is missing more than just "or equal to".
> >
> >> The value of max-smbps MUST be greater than the value of MaxMBPS
> given in
> >> Table A-1 of [1] for the highest level.
> >
> > It should also take max-mbps into account. And the reference to
> "highest level" has no antecedent in this section; it is in the prior
> section on max-mbps.
> >
> > I suggest this [new] text:
> >
> > "The value of max-smbps MUST be greater than [or equal to] the value
> of MaxMBPS given [explicitly in max-mbps or implicitly] in Table A-1 =
of
> [1] for the highest level [given in profile-level-id or max-recv-
> level]."
> >
> > Regards,
> > Mo
> >
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =
Of
> Roni Even
> > Sent: Friday, October 01, 2010 4:49 PM
> > To: 'IETF AVT WG'
> > Cc: 'Randell Jesup'; 'Tom Kristensen'; =
keith.drage@alcatel-lucent.com
> > Subject: Re: [AVT] rfc3984bis - max-smbps limit
> >
> > Hi,
> >
> > We got the following comment as a private email (bellow) . The draft
> has not
> > be published yet and we can do this change before publication.
> > Please review and send comments to the list since it changes the
> current
> > semantics. If there are no objections we will do this change.
> > Since this parameter has been added in the bis draft and was not =
part
> of RFC
> > 3984 we do not see any interoperability issues.
> >
> > Thanks
> > Roni Even
> > As a co-author of the draft
> >
> >
> > Email from Tom Kristensen
> >
> > I discovered missing alignment between H.241 MaxStaticMBPS and max-
> smbps as
> > defined in draft-ietf-avt-rtp-rfc3984bis.
> >
> >   From Table 8-9, H.241:
> > ---
> > The encoder should recompute this value for each picture.
> > The value of (MaxStaticMBPS =AA 500) shall not be less than the =
value
> MaxMBPS
> > for the Level given in Table A-1/H.264, and if CustomMaxMBPS is
> signalled,
> > shall not be less than the value (CustomMaxMBPS =AA 500).
> > ---
> >
> > On the other hand in the SIP world, rfc3984bis says:
> > ---
> > The value of max-smbps MUST be greater than the value of MaxMBPS
> given in
> > Table A-1 of [1] for the highest level.
> > ---
> > http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44
> >
> >
> > H.241 says      max-smbps>=3D max-mbps.
> > rfc3984bis says max-smbps>   max-mbps.
> > We are missing "or equal".
> >
> > Of course it doesn't really make much meaning to include a max-smbps
> > equal to max-mbps, but we have a mismatch between H.241 and the SDP
> > counterpart.
> >
> > Is it possible and feasible to alter the text given the current =
state
> of
> > soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From gonzalo.camarillo@ericsson.com  Mon Oct  4 03:18:17 2010
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Date: Mon, 04 Oct 2010 13:19:09 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [AVT] AD review: draft-ietf-avt-rtp-h264-rcdo-06
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Hi,

as part of a load balancing exercise between Robert and I, I will be
acting as the responsible AD for this draft.

https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-h264-rcdo/

Below you can find my AD review. As soon as the authors address these
comments (it should be trivial to address them), I will IETF Last Call
this draft.

Thanks,

Gonzalo


The acronyms RTP and RCDO need to be expanded in the title of the
draft. In general, all acronyms need to be expanded on their first
use.

The document refers to [3] as RFC3984bis. Given that [3] is a
normative reference, this document will not be published until [3] is
published as well. So, referring to it as, say, RFCYYYY and then get
the RFC Editor to update the text seems like a better option than
calling it RFC3984bis.

Page 19: section X should be section 9.

Section 9: "although if suitable the usage of SRTP [11] is
recommended". Should that be a normative RECOMMENDED instead?

The draft should have a reference to the SDP spec: RFC 4566.



From magnus.westerlund@ericsson.com  Mon Oct  4 04:58:42 2010
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Subject: Re: [AVT] Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to	Proposed Standard
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Hi,

Responses inline.

Ali C. Begen (abegen) skrev 2010-09-29 17:32:
> Hi Magnus,
> 
> We are all glad to have you back. Comments are inline. We will get these into the revision. See inline.
> 
>> I have the following comments on this document, I think version 7 or 8
>> was the last version I read before my parental leave.
>>
>>
>> 1. Applicability statement: I think this document should have an
>> applicability statement making clear that this is only for engineered
>> environments due to the traffic bursts.
> 
> We already put a lot of effort to explain the precautions for using this in BE networks. I don't think there is a strong argument here that this will only work in engineered networks. So, personally I don't think we should put such a statement.

I can agree that it works in certain type of best-effort networks.
However, I don't think at all it is suitable for any best-effort
network. However, as this is mostly bound by the requirement to also
support SSM in the same network the deployment cases are usually meeting
the assumptions about network capacity. However, if one runs an
multicast overlay and a client starts up with not congestion or network
capacity information and does a RAMS-R it can severely congest a network
for a few seconds.

There are a number of assumptions in where this will work most of the
time and with very few exceptions have any large scale impact on the
network. I think these should be documented in an applicability statement.

>  
>> 2. Section 6.2, bullet 5: I don't think the document is clear on that
>> the BRS is expected to determine if a RAMS-R is a retransmission or an
>> update based on if any content of the message has been changed.
> 
> The statement over there implies this. And naturally it is the server's job to figure that out. Based on this discussion, we even wrote the new CNAME guidelines document so that the server's job would get easier.

Exactly the document implies rather than being explicit. I am not
certain that I have understood all the mechanism you expect me to use to
determine if a RAMS-R is a retransmission or a new request. Based on
that it is uncertain for a client to know what effect an RAMS-R message
will have. I also think it makes sense to mandate that servers do have
retransmission detection, otherwise you end up in a situation where a
client don't dare retransmit a request for the fear of getting two bursts.

>  
>> 3. I worried that what is intended to be an RAMS-R update from the
>> client will be interpreted as new RAMS-R request. The reason is the only
>> that separates these two cases are timing, does it arrive before the
>> burst ends or not. Relying on this heuristics is quite weak and error
>> prone. I still wished that an sequence number had been added to RAMS-R.
> 
> I gave enough arguments why we should not use a seqnum for the RAMS-R messages earlier. It is not justified IMO. Based on the SSRC of the primary and the requesting client's CNAME, things are pretty straightforward.

Yes, I know I am in the rough here. From my perspective, it makes the
use of update RAMS-R uncertain to use. Maybe not a big issue if most
doesn't implement updates. But if one finds need for it then there will
be issues. Consider the AD and IESG notified about the potential issue.

>  
>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globally
>>         unique CNAME identifier."
>>
>> I don't think this is a statement that can either be fulfilled nor
>> tested. You can mandate a method for selecting CNAMEs that has low risk
>> of producing CNAME collisions, but nothing can guarantee it unless a
>> entity is coordinating the CNAME space for all clients. Can you mandate
>> the method instead?
> 
> The new CNAME draft has ways to ensure they are unique. If the client uses one of them, we are all set.

Yes, but in that case mandate the implementation and usage of one of
these instead of statement that can't be ensured.

>  
>> If I understand the impact of a CNAME collision is that the collision
>> clients request will be mixed up, for example terminating each others
>> request, or update the values in the RAMS-R.
> 
> When they are unique, this won't happen.

Just checking, but if that is the case then I am missing a discussion of
hijacking attacks in the security consideration section by guessing your
targets CNAME.

>  
>> 5. Section 6.4 and others: The draft has a tendency of formulating
>> itself as everything is guaranteed to work as long as one keep within
>> given limits for bandwidth etc. An example from Section 7.2 Max Receive
>> Bitrate TLV:
> 
> We don't assume things will sure work or guarantee that they will work if everything is satisfied. These are simple rules that both sides must comply with.

No, but a certain view point is clear in the text. Lets call it
optimistic that it will work, while I would write from a more
pessimistic view.

>  
>>       If specified, the total bitrate of the unicast burst(s) plus any
>>       payload-specific information MUST NOT be larger than this value.
>>       Otherwise, congestion and packet loss may occur.
>>
>> The last sentence I would formulate as:
>> "Otherwise, congestion and packet loss are much likelier to occur"
> 
> How about "more likely to occur"?

Sure.

>  
>> Even if one stays within this value congestion and packet loss may
>> occur. This is not the only case, but when I actually decided this seems
>> to be an issue, I hadn't taken note of all examples I seen.
>>
>> 6. Section 7:
>>
>> Each feedback message has a fixed-length field for version, padding,
>>    feedback message type (FMT), payload type (PT), length, SSRC of
>>    packet sender, SSRC of media sender as well as a variable-length
>>    field for feedback control information (FCI).
>>
>>
>> What is called "Payload type" is actually "Packet Type" in RTCP packet
>> headers.
> 
> We followed the definition from 4585 but "packet type" makes more sense to me as well.

Well, section 6.1 of RFC 4585 is wrong.

>  
>> 7. Section 7.3:
>> "The MSN value SHALL
>>       be set to zero only when a new RAMS request is received."
>>
>> How is that actually known? And why reset it at all? Why not increase if
>> for each new RAMS-I message with new content, independently if it is an
>> update or a new request.
> 
> If this is relating to a new burst request, then it is reset. Ow, what is the point of having a seqnum? If something has changed compared to the previous RAMS-I, then MSN is incremented. If it is just a re-xmit, MSN stays the same. 

I fully agree with the need for separating retransmissions from updates.
However, I wonder over the reset of the field for each new RAMS-R. It
appears to me to be more robust to simply increment it rather than
reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN = 0.
Then a RAMS-R(2) that is intended to be an update but becomes an new
request results in an RAMS-I with MSN = 0. The client will not know if
this is an retransmission of RAMS-R(1) info. The updated should result
in MSN=1. So without comparing the RAMS-I you can't determine if there
is new information. Going my way (no reset) does not allow a client to
determine if an RAMS-R was treated as update or new request, but it will
correctly know that it is new RAMS-I information.

>  
>> 8. Section 7.3:
>> "When the RTP_Rx receives a RAMS-I message with a response code
>>       that it does not understand, the RTP_Rx MUST send a RAMS-T message
>>       immediately to the BRS."
>>
>> Which media sender SSRC should the client direct this request to if it
>> hasn't been signalled in SDP nor any RTP/RTCP packets received?
> 
> The rules in the RAMS-T section applies. You received a RAMS-I so you know the SSRC.

Yes, you are correct. My mistake.

>  
>> 9. Section 7.3, Media Sender SSRC TLV:
>> " While this
>>       information is already available in the message header, it can be
>>       useful to repeat it in an explicit field."
>>
>> This sentence seems out of place compared to the rest of the text.
> 
> If I recall correctly, this was something you asked for. I am not sure whether there is a better place to put it. Maybe in parentheses? 

I think we can remove that sentence. I think you now have text making
clear when it needs to be included and when it shouldn't.

>  
>> 10. Section 7.3, RTP Seqnum of the First Packet TLV:
>>
>> How is this TLV bound to the SSRC number if is for? This is not
>> explicitly explained.
> 
> What do you mean? The SSRC comes from the RAMS-I's SSRC. There is one (not multiple) stream per RAMS-I.

Yes, something that I apparently have forgotten.
>  
>> 11. Section 7.4:
>>  Each of these RAMS-T messages has
>>    the appropriate media sender SSRC populated in its message header.
>>
>> Instead of writing "appropriate" cant you simply write " Each of these
>> RAMS-T message's media sender SSRC SHALL BE populated with the SSRC of
>> the Media Stream to be terminated."
> 
> Good suggestion.
>  
>> 12. Section 8.3:
>> "This section provides a declarative SDP [RFC4566] example for
>>    enabling rapid acquisition of multicast RTP sessions."
>>
>> Can you make clear that this an example of an SDP to be provided to a
>> client?
> 
> OK.
>  
>> 13. Section 9, bullet b:
>> Be configured to forward certain ports (e.g., using HTML
>>        configuration and UPnP IGD [UPnP-IGD]).
>>
>> Shouldn't the "and" be an "or"?
> 
> Yes, good catch.
>  
>> 14. Section 10:
>>
>> Shouldn't the security consideration make it clear that RAMS-R are
>> especially suspectible to Replay attacks as there is no information in
>> the packet that one can use to detect that it is out of sequence.
> 
> There is a wording about this in that section (which simply refers the reader to 5760). Are you considering a RAMS-specific replay attack here? 


Yes, I am considering that it is easy to target RAMS-R specifically for
an replay attack. Especially when sent in a reduced size RTCP packet
only containing RAMS-R and SDES CNAME. That has no time specific
information and all replay detection must happen in the security protocol.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From gonzalo.camarillo@ericsson.com  Mon Oct  4 05:55:22 2010
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Hi,

as part of a load balancing exercise between Robert and I, I will be
acting as the responsible AD for this draft.

https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-svc/

Below you can find my AD review. As soon as the authors address these
comments (it should be trivial to address them), I will IETF Last Call
this draft.

Thanks,

Gonzalo


The acronyms RTP and SVC need to be expanded in the title. All acronyms
in the draft need to be expanded on their first use.

Section 1.2.2 has a few non-normative statements such as "SST should
be used...". This normatively specified at a later point in the
document (Section 4.4). If Section 1 is meant to be a non-normative
overview of the specification, the draft should say so somewhere.

Page 80. When specifying behavior beyond RFC 3264, the draft should
use normative language. However, when specifying behavior that is
already specified in RFC 3264, the draft should not use normative
language. For example: "the offer SHOULD also be used in the answer,
as specified in [RFC3264]". This issue also appears in other sections
in the document (e.g., in page 85). Simply s/SHOULD/should/ would
resolve the issue, for instance.


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Hi Magnus,

> >> 1. Applicability statement: I think this document should have an
> >> applicability statement making clear that this is only for =
engineered
> >> environments due to the traffic bursts.
> >
> > We already put a lot of effort to explain the precautions for using =
this in BE networks. I don't think there is a strong
> argument here that this will only work in engineered networks. So, =
personally I don't think we should put such a statement.
>=20
> I can agree that it works in certain type of best-effort networks.
> However, I don't think at all it is suitable for any best-effort
> network. However, as this is mostly bound by the requirement to also
> support SSM in the same network the deployment cases are usually =
meeting
> the assumptions about network capacity. However, if one runs an
> multicast overlay and a client starts up with not congestion or =
network

That is a bit stretch :)

> capacity information and does a RAMS-R it can severely congest a =
network
> for a few seconds.

Well we have rules how the server is supposed to send the burst (rate, =
timing, etc.) documented in the text. It explains what the server does.
=20
> There are a number of assumptions in where this will work most of the
> time and with very few exceptions have any large scale impact on the
> network. I think these should be documented in an applicability =
statement.

At the end of intro, I think we can put a few sentences about this.
=20
> >> 2. Section 6.2, bullet 5: I don't think the document is clear on =
that
> >> the BRS is expected to determine if a RAMS-R is a retransmission or =
an
> >> update based on if any content of the message has been changed.
> >
> > The statement over there implies this. And naturally it is the =
server's job to figure that out. Based on this discussion, we
> even wrote the new CNAME guidelines document so that the server's job =
would get easier.
>=20
> Exactly the document implies rather than being explicit. I am not

Actually, I re-read it and it is pretty explicit. The text says it =
checks client's CNAME (which must be unique). If the server is already =
processing a request from the same client for the same SSRC, then it is =
a update. Ow, it is a new request.

I can repeat the above in the text if you think it is necessary.

> certain that I have understood all the mechanism you expect me to use =
to
> determine if a RAMS-R is a retransmission or a new request. Based on
> that it is uncertain for a client to know what effect an RAMS-R =
message
> will have. I also think it makes sense to mandate that servers do have
> retransmission detection, otherwise you end up in a situation where a
> client don't dare retransmit a request for the fear of getting two =
bursts.

OK what if the text said:
=20
        Updated Request:  The RTP_Rx MAY send an updated RAMS-R message
        (as unicast feedback in the primary multicast RTP session) with
        a different value for one or more fields of an earlier RAMS-R
        message. The RS MUST be able to detect whether a burst is =
already planned for or being transmitted to this particular RTP_Rx for =
this particular media source SSRC. If there is an existing burst planned =
for or being transmitted, the newly arriving RAMS-R is an updated =
request; otherwise it is a new request...
    =20
> >> 3. I worried that what is intended to be an RAMS-R update from the
> >> client will be interpreted as new RAMS-R request. The reason is the =
only
> >> that separates these two cases are timing, does it arrive before =
the
> >> burst ends or not. Relying on this heuristics is quite weak and =
error
> >> prone. I still wished that an sequence number had been added to =
RAMS-R.
> >
> > I gave enough arguments why we should not use a seqnum for the =
RAMS-R messages earlier. It is not justified IMO. Based
> on the SSRC of the primary and the requesting client's CNAME, things =
are pretty straightforward.
>=20
> Yes, I know I am in the rough here. From my perspective, it makes the
> use of update RAMS-R uncertain to use. Maybe not a big issue if most
> doesn't implement updates. But if one finds need for it then there =
will
> be issues. Consider the AD and IESG notified about the potential =
issue.

Is the updated text above good for you?
=20
> >
> >> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globally
> >>         unique CNAME identifier."
> >>
> >> I don't think this is a statement that can either be fulfilled nor
> >> tested. You can mandate a method for selecting CNAMEs that has low =
risk
> >> of producing CNAME collisions, but nothing can guarantee it unless =
a
> >> entity is coordinating the CNAME space for all clients. Can you =
mandate
> >> the method instead?
> >
> > The new CNAME draft has ways to ensure they are unique. If the =
client uses one of them, we are all set.
>=20
> Yes, but in that case mandate the implementation and usage of one of
> these instead of statement that can't be ensured.

Personally, this is one what I did initially but then we agreed that one =
could come up with an alternative method that would produce unique =
CNAMEs. In other words, methods for producing unique CNAMEs are not =
unique. So, we rather put the "MUST" in the sentence above not for the =
specific implementation.

> >> If I understand the impact of a CNAME collision is that the =
collision
> >> clients request will be mixed up, for example terminating each =
others
> >> request, or update the values in the RAMS-R.
> >
> > When they are unique, this won't happen.
>=20
> Just checking, but if that is the case then I am missing a discussion =
of
> hijacking attacks in the security consideration section by guessing =
your
> targets CNAME.

We should probably mention this but I am not sure how the server can =
deal with this. Hijacking is not easy since the attack must take place =
at the same instant (more or less) with the request from the authentic =
client. One of your family members can probably do this :)
=20
> >> 5. Section 6.4 and others: The draft has a tendency of formulating
> >> itself as everything is guaranteed to work as long as one keep =
within
> >> given limits for bandwidth etc. An example from Section 7.2 Max =
Receive
> >> Bitrate TLV:
> >
> > We don't assume things will sure work or guarantee that they will =
work if everything is satisfied. These are simple rules
> that both sides must comply with.
>=20
> No, but a certain view point is clear in the text. Lets call it
> optimistic that it will work, while I would write from a more
> pessimistic view.

Things may go wrong anytime, and we do ack that in the paper and have =
solutions. But if one does not comply with the rules, more things will =
likely go wrong.
=20
> >> 6. Section 7:
> >>
> >> Each feedback message has a fixed-length field for version, =
padding,
> >>    feedback message type (FMT), payload type (PT), length, SSRC of
> >>    packet sender, SSRC of media sender as well as a variable-length
> >>    field for feedback control information (FCI).
> >>
> >>
> >> What is called "Payload type" is actually "Packet Type" in RTCP =
packet
> >> headers.
> >
> > We followed the definition from 4585 but "packet type" makes more =
sense to me as well.
>=20
> Well, section 6.1 of RFC 4585 is wrong.

OK, "packet type" it is then.
=20
> >
> >> 7. Section 7.3:
> >> "The MSN value SHALL
> >>       be set to zero only when a new RAMS request is received."
> >>
> >> How is that actually known? And why reset it at all? Why not =
increase if
> >> for each new RAMS-I message with new content, independently if it =
is an
> >> update or a new request.
> >
> > If this is relating to a new burst request, then it is reset. Ow, =
what is the point of having a seqnum? If something has
> changed compared to the previous RAMS-I, then MSN is incremented. If =
it is just a re-xmit, MSN stays the same.
>=20
> I fully agree with the need for separating retransmissions from =
updates.
> However, I wonder over the reset of the field for each new RAMS-R. It
> appears to me to be more robust to simply increment it rather than
> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN =3D 0.

I think we discussed this before. The RAMS-R numbers are no way =
correlated with the RAMS-I numbers. You are still trying to correlate =
them.

> Then a RAMS-R(2) that is intended to be an update but becomes an new
> request results in an RAMS-I with MSN =3D 0. The client will not know =
if
> this is an retransmission of RAMS-R(1) info. The updated should result
> in MSN=3D1. So without comparing the RAMS-I you can't determine if =
there

The client checks the RAMS-I seqnum to see whether it is a repeat or new =
info. If RAMS-I MSN is zero, that is the first RAMS-I anyway so it must =
be fully parsed. Does not matter which RAMS-R actually generated it =
since that is the info from the server until an updated RAMS-I is =
received. This is how the protocol works.

> is new information. Going my way (no reset) does not allow a client to
> determine if an RAMS-R was treated as update or new request, but it =
will
> correctly know that it is new RAMS-I information.

The server cannot keep state information (i.e. tracking RAMS-R numbers) =
across different sessions. Furthermore, requests for different streams =
can go to different servers.
=20
> >> 9. Section 7.3, Media Sender SSRC TLV:
> >> " While this
> >>       information is already available in the message header, it =
can be
> >>       useful to repeat it in an explicit field."
> >>
> >> This sentence seems out of place compared to the rest of the text.
> >
> > If I recall correctly, this was something you asked for. I am not =
sure whether there is a better place to put it. Maybe in
> parentheses?
>=20
> I think we can remove that sentence. I think you now have text making
> clear when it needs to be included and when it shouldn't.

Ok.
=20
> >> 14. Section 10:
> >>
> >> Shouldn't the security consideration make it clear that RAMS-R are
> >> especially suspectible to Replay attacks as there is no information =
in
> >> the packet that one can use to detect that it is out of sequence.
> >
> > There is a wording about this in that section (which simply refers =
the reader to 5760). Are you considering a RAMS-specific
> replay attack here?
>=20
>=20
> Yes, I am considering that it is easy to target RAMS-R specifically =
for
> an replay attack. Especially when sent in a reduced size RTCP packet
> only containing RAMS-R and SDES CNAME. That has no time specific
> information and all replay detection must happen in the security =
protocol.

The Token stuff (or the cookie stuff) will avoid request messages from =
being sent by others (it will do at least a reverse path check). Beyond =
that is not SRTCP a good solution to avoid this?

-acbegen

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On 4 Oct 2010, at 13:53, Gonzalo Camarillo wrote:
> as part of a load balancing exercise between Robert and I, I will be
> acting as the responsible AD for this draft.
>
> https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-svc/
>
> Below you can find my AD review. As soon as the authors address these
> comments (it should be trivial to address them), I will IETF Last Call
> this draft.
>
> Thanks,
>
> Gonzalo
>
>
> The acronyms RTP and SVC need to be expanded in the title. All  
> acronyms
> in the draft need to be expanded on their first use.

Unless their policy has changed recently, RTP is one of the few well- 
known acronyms the RFC editor allows to be used without expansion in  
document titles.

> Section 1.2.2 has a few non-normative statements such as "SST should
> be used...". This normatively specified at a later point in the
> document (Section 4.4). If Section 1 is meant to be a non-normative
> overview of the specification, the draft should say so somewhere.
>
> Page 80. When specifying behavior beyond RFC 3264, the draft should
> use normative language. However, when specifying behavior that is
> already specified in RFC 3264, the draft should not use normative
> language. For example: "the offer SHOULD also be used in the answer,
> as specified in [RFC3264]". This issue also appears in other sections
> in the document (e.g., in page 85). Simply s/SHOULD/should/ would
> resolve the issue, for instance.


-- 
Colin Perkins
http://csperkins.org/




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Ali C. Begen (abegen) skrev 2010-10-04 14:57:
> Hi Magnus,
> 
>>>> 1. Applicability statement: I think this document should have an
>>>> applicability statement making clear that this is only for engineered
>>>> environments due to the traffic bursts.
>>>
>>> We already put a lot of effort to explain the precautions for using this in BE networks. I don't think there is a strong
>> argument here that this will only work in engineered networks. So, personally I don't think we should put such a statement.
>>
>> I can agree that it works in certain type of best-effort networks.
>> However, I don't think at all it is suitable for any best-effort
>> network. However, as this is mostly bound by the requirement to also
>> support SSM in the same network the deployment cases are usually meeting
>> the assumptions about network capacity. However, if one runs an
>> multicast overlay and a client starts up with not congestion or network
> 
> That is a bit stretch :)
> 
>> capacity information and does a RAMS-R it can severely congest a network
>> for a few seconds.
> 
> Well we have rules how the server is supposed to send the burst (rate, timing, etc.) documented in the text. It explains what the server does.
>  
>> There are a number of assumptions in where this will work most of the
>> time and with very few exceptions have any large scale impact on the
>> network. I think these should be documented in an applicability statement.
> 
> At the end of intro, I think we can put a few sentences about this.

Ok, I guess I have to live with that.

>  
>>>> 2. Section 6.2, bullet 5: I don't think the document is clear on that
>>>> the BRS is expected to determine if a RAMS-R is a retransmission or an
>>>> update based on if any content of the message has been changed.
>>>
>>> The statement over there implies this. And naturally it is the server's job to figure that out. Based on this discussion, we
>> even wrote the new CNAME guidelines document so that the server's job would get easier.
>>
>> Exactly the document implies rather than being explicit. I am not
> 
> Actually, I re-read it and it is pretty explicit. The text says it checks client's CNAME (which must be unique). If the server is already processing a request from the same client for the same SSRC, then it is a update. Ow, it is a new request.
> 
> I can repeat the above in the text if you think it is necessary.
> 
>> certain that I have understood all the mechanism you expect me to use to
>> determine if a RAMS-R is a retransmission or a new request. Based on
>> that it is uncertain for a client to know what effect an RAMS-R message
>> will have. I also think it makes sense to mandate that servers do have
>> retransmission detection, otherwise you end up in a situation where a
>> client don't dare retransmit a request for the fear of getting two bursts.
> 
> OK what if the text said:
>  
>         Updated Request:  The RTP_Rx MAY send an updated RAMS-R message
>         (as unicast feedback in the primary multicast RTP session) with
>         a different value for one or more fields of an earlier RAMS-R
>         message. The RS MUST be able to detect whether a burst is already planned for or being transmitted to this particular RTP_Rx for this particular media source SSRC. If there is an existing burst planned for or being transmitted, the newly arriving RAMS-R is an updated request; otherwise it is a new request...
>      

Better.

>>>> 3. I worried that what is intended to be an RAMS-R update from the
>>>> client will be interpreted as new RAMS-R request. The reason is the only
>>>> that separates these two cases are timing, does it arrive before the
>>>> burst ends or not. Relying on this heuristics is quite weak and error
>>>> prone. I still wished that an sequence number had been added to RAMS-R.
>>>
>>> I gave enough arguments why we should not use a seqnum for the RAMS-R messages earlier. It is not justified IMO. Based
>> on the SSRC of the primary and the requesting client's CNAME, things are pretty straightforward.
>>
>> Yes, I know I am in the rough here. From my perspective, it makes the
>> use of update RAMS-R uncertain to use. Maybe not a big issue if most
>> doesn't implement updates. But if one finds need for it then there will
>> be issues. Consider the AD and IESG notified about the potential issue.
> 
> Is the updated text above good for you?

Not, really, because it has only clarified the issue. The issue is
really the uncertainty for the client how its request will be
interpreted, either as updated request or a new request, all depending
on timing. The biggest issue is the updated request which might be
interpreted as new request, when the client is likely satisfied and
might even have gone one to join the multicast group when the
RAMS-R(update) is processed as a new RAMS-R. Then causing congestion and
unwanted traffic. The client that detects this can surely terminate the
burst, but that will be after some delay, and traffic has arrived.

>  
>>>
>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globally
>>>>         unique CNAME identifier."
>>>>
>>>> I don't think this is a statement that can either be fulfilled nor
>>>> tested. You can mandate a method for selecting CNAMEs that has low risk
>>>> of producing CNAME collisions, but nothing can guarantee it unless a
>>>> entity is coordinating the CNAME space for all clients. Can you mandate
>>>> the method instead?
>>>
>>> The new CNAME draft has ways to ensure they are unique. If the client uses one of them, we are all set.
>>
>> Yes, but in that case mandate the implementation and usage of one of
>> these instead of statement that can't be ensured.
> 
> Personally, this is one what I did initially but then we agreed that one could come up with an alternative method that would produce unique CNAMEs. In other words, methods for producing unique CNAMEs are not unique. So, we rather put the "MUST" in the sentence above not for the specific implementation.

Okay, I might be nit picking then to request that the text is changed to
"Thus, the RTP_Rx MUST use a method that ensure it uses an unique CNAME
identifier." I don't think globally is the correct scope, it is
unnecessary big, when only needs to be unique within this particular
instance of a system.

> 
>>>> If I understand the impact of a CNAME collision is that the collision
>>>> clients request will be mixed up, for example terminating each others
>>>> request, or update the values in the RAMS-R.
>>>
>>> When they are unique, this won't happen.
>>
>> Just checking, but if that is the case then I am missing a discussion of
>> hijacking attacks in the security consideration section by guessing your
>> targets CNAME.
> 
> We should probably mention this but I am not sure how the server can deal with this. Hijacking is not easy since the attack must take place at the same instant (more or less) with the request from the authentic client. One of your family members can probably do this :)

The real solution is where you have an more permanent id system in place
that you can connect through source authentication of the requests.

In an SSM session that uses simple feedback model the RTP_Rx cname may
leak as they are redistributed.

Based on that you could bombard a BRS with RAMS-T for example for all
known CNAMES and do that in a round-robin fashion across channels and
time. Depending on source address spoofing you will more or less easy to
find. But I do agree that it becomes a little bit more a brute force
attack, but an attacker could gain knowledge about an important piece of
information to mount the attack at all.

>  
>>>> 5. Section 6.4 and others: The draft has a tendency of formulating
>>>> itself as everything is guaranteed to work as long as one keep within
>>>> given limits for bandwidth etc. An example from Section 7.2 Max Receive
>>>> Bitrate TLV:
>>>
>>> We don't assume things will sure work or guarantee that they will work if everything is satisfied. These are simple rules
>> that both sides must comply with.
>>
>> No, but a certain view point is clear in the text. Lets call it
>> optimistic that it will work, while I would write from a more
>> pessimistic view.
> 
> Things may go wrong anytime, and we do ack that in the paper and have solutions. But if one does not comply with the rules, more things will likely go wrong.
>  

It is a writing style thing and I think we can stop the discussion.

>>>> 6. Section 7:
>>>>
>>>> Each feedback message has a fixed-length field for version, padding,
>>>>    feedback message type (FMT), payload type (PT), length, SSRC of
>>>>    packet sender, SSRC of media sender as well as a variable-length
>>>>    field for feedback control information (FCI).
>>>>
>>>>
>>>> What is called "Payload type" is actually "Packet Type" in RTCP packet
>>>> headers.
>>>
>>> We followed the definition from 4585 but "packet type" makes more sense to me as well.
>>
>> Well, section 6.1 of RFC 4585 is wrong.
> 
> OK, "packet type" it is then.
>  

Great.

>>>
>>>> 7. Section 7.3:
>>>> "The MSN value SHALL
>>>>       be set to zero only when a new RAMS request is received."
>>>>
>>>> How is that actually known? And why reset it at all? Why not increase if
>>>> for each new RAMS-I message with new content, independently if it is an
>>>> update or a new request.
>>>
>>> If this is relating to a new burst request, then it is reset. Ow, what is the point of having a seqnum? If something has
>> changed compared to the previous RAMS-I, then MSN is incremented. If it is just a re-xmit, MSN stays the same.
>>
>> I fully agree with the need for separating retransmissions from updates.
>> However, I wonder over the reset of the field for each new RAMS-R. It
>> appears to me to be more robust to simply increment it rather than
>> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN = 0.
> 
> I think we discussed this before. The RAMS-R numbers are no way correlated with the RAMS-I numbers. You are still trying to correlate them.

Nope, the number here are still only to indicate that they are different
to get the sequence right. My point is that the client can determine
based on MSN if it is an repeat or a new RAMS-I based on a new request.

> 
>> Then a RAMS-R(2) that is intended to be an update but becomes an new
>> request results in an RAMS-I with MSN = 0. The client will not know if
>> this is an retransmission of RAMS-R(1) info. The updated should result
>> in MSN=1. So without comparing the RAMS-I you can't determine if there
> 
> The client checks the RAMS-I seqnum to see whether it is a repeat or new info. If RAMS-I MSN is zero, that is the first RAMS-I anyway so it must be fully parsed. Does not matter which RAMS-R actually generated it since that is the info from the server until an updated RAMS-I is received. This is how the protocol works.

As I try to explain there is a case where you can get two RAMS-I with
MSN=0 in a row with different information. Thus not providing any
relieve for the client in the need to compare the whole request with the
previous one.

> 
>> is new information. Going my way (no reset) does not allow a client to
>> determine if an RAMS-R was treated as update or new request, but it will
>> correctly know that it is new RAMS-I information.
> 
> The server cannot keep state information (i.e. tracking RAMS-R numbers) across different sessions. Furthermore, requests for different streams can go to different servers.
>  

No, I am not talking about RAMS-R number here. I am talking about RAMS-I
MSN numbers, that they shouldn't be kept. But I see your point in that
you don't want to keep CNAME,SSRC => MSN counter mappings.

But in that case, I think we should make it clear that a client that has
sent a RAMS-R (update) must check if the RAMS-I information is same or
different to determine if the server interpreted the update as a new
request.

>>>> 14. Section 10:
>>>>
>>>> Shouldn't the security consideration make it clear that RAMS-R are
>>>> especially suspectible to Replay attacks as there is no information in
>>>> the packet that one can use to detect that it is out of sequence.
>>>
>>> There is a wording about this in that section (which simply refers the reader to 5760). Are you considering a RAMS-specific
>> replay attack here?
>>
>>
>> Yes, I am considering that it is easy to target RAMS-R specifically for
>> an replay attack. Especially when sent in a reduced size RTCP packet
>> only containing RAMS-R and SDES CNAME. That has no time specific
>> information and all replay detection must happen in the security protocol.
> 
> The Token stuff (or the cookie stuff) will avoid request messages from being sent by others (it will do at least a reverse path check). Beyond that is not SRTCP a good solution to avoid this?

SRTCP is if keyed correctly a good solution yes.

 My point is that one normally document the vulnerability and then
ensure that one has a remedy for it. When it comes to the token stuff I
am not certain that it helps against a replay attack. Unless the token
has expired on the server you still will get traffic to the target. That
is why I think a reverse path check when you start delivering should
happen. Because first in that case if there is no listener at the
indicated address and port will you find out that you are attempting a
delivery not intended.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Subject: Re: [AVT] Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to	Proposed Standard
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I am glad that we are converging on something.

> >>>> 3. I worried that what is intended to be an RAMS-R update from =
the
> >>>> client will be interpreted as new RAMS-R request. The reason is =
the only
> >>>> that separates these two cases are timing, does it arrive before =
the
> >>>> burst ends or not. Relying on this heuristics is quite weak and =
error
> >>>> prone. I still wished that an sequence number had been added to =
RAMS-R.
> >>>
> >>> I gave enough arguments why we should not use a seqnum for the =
RAMS-R messages earlier. It is not justified IMO.
> Based
> >> on the SSRC of the primary and the requesting client's CNAME, =
things are pretty straightforward.
> >>
> >> Yes, I know I am in the rough here. From my perspective, it makes =
the
> >> use of update RAMS-R uncertain to use. Maybe not a big issue if =
most
> >> doesn't implement updates. But if one finds need for it then there =
will
> >> be issues. Consider the AD and IESG notified about the potential =
issue.
> >
> > Is the updated text above good for you?
>=20
> Not, really, because it has only clarified the issue. The issue is
> really the uncertainty for the client how its request will be
> interpreted, either as updated request or a new request, all depending
> on timing. The biggest issue is the updated request which might be
> interpreted as new request, when the client is likely satisfied and

Why is this a problem? For this to happen, the initial request must have =
been lost. And if it is lost and the second makes it, the client still =
gets what it wants. I don't see a problem here.

> might even have gone one to join the multicast group when the
> RAMS-R(update) is processed as a new RAMS-R. Then causing congestion =
and

Are you saying this:
1- The client makes a request but it gets lost. In the mean time, the =
client sends an update and the server thinks it is a new request and =
starts sending the burst
2- In the mean time, client gives up and joins the multicast anyway.
3- Burst + mcast creates a congestion.

> unwanted traffic. The client that detects this can surely terminate =
the
> burst, but that will be after some delay, and traffic has arrived.

OK, so you are saying what I wrote above. Well, the client has to send a =
RAMS-T and upon receiving it, the server stops the burst. So that is not =
a big deal. If you are concerned about RAMS-T being lost (which is =
repeated if the bust is not stopped), then I will just remind you that =
this is a protocol where control messages are not fully reliable (they =
are just repeated for redundancy as frequently as 4585 allows). If you =
are really this unlucky, you will have a problem but it will be only =
temporary.=20

I don't think there is anything here that requires for us to burn our =
energy. FWIW, your proposal would not solve this problem entirely, =
either.
=20
> >
> >>>
> >>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a =
globally
> >>>>         unique CNAME identifier."
> >>>>
> >>>> I don't think this is a statement that can either be fulfilled =
nor
> >>>> tested. You can mandate a method for selecting CNAMEs that has =
low risk
> >>>> of producing CNAME collisions, but nothing can guarantee it =
unless a
> >>>> entity is coordinating the CNAME space for all clients. Can you =
mandate
> >>>> the method instead?
> >>>
> >>> The new CNAME draft has ways to ensure they are unique. If the =
client uses one of them, we are all set.
> >>
> >> Yes, but in that case mandate the implementation and usage of one =
of
> >> these instead of statement that can't be ensured.
> >
> > Personally, this is one what I did initially but then we agreed that =
one could come up with an alternative method that would
> produce unique CNAMEs. In other words, methods for producing unique =
CNAMEs are not unique. So, we rather put the
> "MUST" in the sentence above not for the specific implementation.
>=20
> Okay, I might be nit picking then to request that the text is changed =
to
> "Thus, the RTP_Rx MUST use a method that ensure it uses an unique =
CNAME
> identifier." I don't think globally is the correct scope, it is
> unnecessary big, when only needs to be unique within this particular
> instance of a system.

Fine with me.
=20
> >
> >>>> If I understand the impact of a CNAME collision is that the =
collision
> >>>> clients request will be mixed up, for example terminating each =
others
> >>>> request, or update the values in the RAMS-R.
> >>>
> >>> When they are unique, this won't happen.
> >>
> >> Just checking, but if that is the case then I am missing a =
discussion of
> >> hijacking attacks in the security consideration section by guessing =
your
> >> targets CNAME.
> >
> > We should probably mention this but I am not sure how the server can =
deal with this. Hijacking is not easy since the attack
> must take place at the same instant (more or less) with the request =
from the authentic client. One of your family members
> can probably do this :)
>=20
> The real solution is where you have an more permanent id system in =
place
> that you can connect through source authentication of the requests.
>=20
> In an SSM session that uses simple feedback model the RTP_Rx cname may
> leak as they are redistributed.
>=20
> Based on that you could bombard a BRS with RAMS-T for example for all
> known CNAMES and do that in a round-robin fashion across channels and
> time. Depending on source address spoofing you will more or less easy =
to
> find. But I do agree that it becomes a little bit more a brute force
> attack, but an attacker could gain knowledge about an important piece =
of
> information to mount the attack at all.

SRTCP?
=20
> >>>
> >>>> 7. Section 7.3:
> >>>> "The MSN value SHALL
> >>>>       be set to zero only when a new RAMS request is received."
> >>>>
> >>>> How is that actually known? And why reset it at all? Why not =
increase if
> >>>> for each new RAMS-I message with new content, independently if it =
is an
> >>>> update or a new request.
> >>>
> >>> If this is relating to a new burst request, then it is reset. Ow, =
what is the point of having a seqnum? If something has
> >> changed compared to the previous RAMS-I, then MSN is incremented. =
If it is just a re-xmit, MSN stays the same.
> >>
> >> I fully agree with the need for separating retransmissions from =
updates.
> >> However, I wonder over the reset of the field for each new RAMS-R. =
It
> >> appears to me to be more robust to simply increment it rather than
> >> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN =3D =
0.
> >
> > I think we discussed this before. The RAMS-R numbers are no way =
correlated with the RAMS-I numbers. You are still trying
> to correlate them.
>=20
> Nope, the number here are still only to indicate that they are =
different
> to get the sequence right. My point is that the client can determine
> based on MSN if it is an repeat or a new RAMS-I based on a new =
request.

When the client receives a RAMS-I with MSN=3D0, it knows that RAMS-I was =
sent for a RAMS-R message that was received the first time by the server =
or an identical repeat of the initial RAMS-I message.=20

But even if the client sends an updated request, the server may ignore =
it, may ignore the changes and subsequently repeat the earlier RAMS-I =
with no changes in it, which will have the previous MSN. The server may =
not send anything at all or the message may get lost. The client cannot =
assume the changes it requested were honored by the server UNLESS there =
was an updated RAMS-I from the server. Even in that case, the RAMS-I =
changes may be due to other things the server has observed - not the =
changes the client asked.

So, the client should not really read too much in to the MSN values =
received. That is what I have been trying to explain in this discussion.
=20
> >
> >> Then a RAMS-R(2) that is intended to be an update but becomes an =
new
> >> request results in an RAMS-I with MSN =3D 0. The client will not =
know if
> >> this is an retransmission of RAMS-R(1) info. The updated should =
result
> >> in MSN=3D1. So without comparing the RAMS-I you can't determine if =
there
> >
> > The client checks the RAMS-I seqnum to see whether it is a repeat or =
new info. If RAMS-I MSN is zero, that is the first
> RAMS-I anyway so it must be fully parsed. Does not matter which RAMS-R =
actually generated it since that is the info from the
> server until an updated RAMS-I is received. This is how the protocol =
works.
>=20
> As I try to explain there is a case where you can get two RAMS-I with
> MSN=3D0 in a row with different information. Thus not providing any
> relieve for the client in the need to compare the whole request with =
the
> previous one.

So what? If you made a single request and received two RAMS-I messages =
with MSN=3D0, that means they are identical. No need to compare them. If =
you made two requests and received two RAMS-I messages with MSN=3D0, =
they are different messages and you need to fully read them anyway.
=20
> >
> >> is new information. Going my way (no reset) does not allow a client =
to
> >> determine if an RAMS-R was treated as update or new request, but it =
will
> >> correctly know that it is new RAMS-I information.
> >
> > The server cannot keep state information (i.e. tracking RAMS-R =
numbers) across different sessions. Furthermore, requests
> for different streams can go to different servers.
> >
>=20
> No, I am not talking about RAMS-R number here. I am talking about =
RAMS-I
> MSN numbers, that they shouldn't be kept. But I see your point in that
> you don't want to keep CNAME,SSRC =3D> MSN counter mappings.
>=20
> But in that case, I think we should make it clear that a client that =
has
> sent a RAMS-R (update) must check if the RAMS-I information is same or
> different to determine if the server interpreted the update as a new
> request.
>=20
> >>>> 14. Section 10:
> >>>>
> >>>> Shouldn't the security consideration make it clear that RAMS-R =
are
> >>>> especially suspectible to Replay attacks as there is no =
information in
> >>>> the packet that one can use to detect that it is out of sequence.
> >>>
> >>> There is a wording about this in that section (which simply refers =
the reader to 5760). Are you considering a RAMS-
> specific
> >> replay attack here?
> >>
> >>
> >> Yes, I am considering that it is easy to target RAMS-R specifically =
for
> >> an replay attack. Especially when sent in a reduced size RTCP =
packet
> >> only containing RAMS-R and SDES CNAME. That has no time specific
> >> information and all replay detection must happen in the security =
protocol.
> >
> > The Token stuff (or the cookie stuff) will avoid request messages =
from being sent by others (it will do at least a reverse path
> check). Beyond that is not SRTCP a good solution to avoid this?
>=20
> SRTCP is if keyed correctly a good solution yes.
>=20
>  My point is that one normally document the vulnerability and then
> ensure that one has a remedy for it. When it comes to the token stuff =
I
> am not certain that it helps against a replay attack. Unless the token
> has expired on the server you still will get traffic to the target. =
That
> is why I think a reverse path check when you start delivering should
> happen. Because first in that case if there is no listener at the
> indicated address and port will you find out that you are attempting a
> delivery not intended.

Token provides RPF check. But, let's work on this Sec. section together. =
I could use some help.

Thanks.
-acbegen

From tom.kristensen@tandberg.com  Mon Oct  4 14:16:02 2010
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Cc: 'Randell Jesup' <rjesup@wgate.com>, keith.drage@alcatel-lucent.com, 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] rfc3984bis - max-smbps limit
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Roni,
I propose clarifying "highest level" first time the term is used for 
max-smbps, that is in the first bullet point at page 44. New text in []:
----
   o If the parameter max-mbps is signalled, set a variable
       MaxMacroblocksPerSecond to the value of max-mbps.
       Otherwise, set MaxMacroblocksPerSecond equal to the value
       of MaxMBPS in Table A-1 [1] for the [signaled] highest level
       [conveyed in the value of the profile-level-id parameter or the
        max-recv-level parameter].
----

Then the paragraph on top of page 45, after the bullet points should be:
----
   The encoder should recompute this value for each picture. The
   value of max-smbps MUST be greater than [or equal to] the value
   of MaxMBPS given [explicitly as the value of the max-mbps
   parameter or implicitly] in Table A-1 of [1] for the [signaled]
   highest level. Senders MAY use this knowledge to send pictures
   of a given size at a higher picture rate than is indicated in the
   signaled highest level.
----

This way the "highest level" is explained initially in the scope of the 
parameter and referenced as "signaled highest level" later on as done 
for the other parameters.

-- Tom

On 10/03/2010 02:46 PM, Roni Even wrote:
> Tom,
> Can you suggest a final text so that we can ask the AD what to do
> Roni
>
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Tom Kristensen
>> Sent: Sunday, October 03, 2010 11:49 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: Randell Jesup; keith.drage@alcatel-lucent.com; IETF AVT WG; Roni
>> Even
>> Subject: Re: [AVT] rfc3984bis - max-smbps limit
>>
>> Indeed, Mo, we have to take a potential max-mbps into account in the
>> description and the clarification of "highest level" also in the
>> max-smbps text part is just fine.
>>
>> Even though "highest level" is described under a common heading for
>> these parameters, it is repeated for the other parameters.
>> Cf.:
>>          max-mbps, max-smbps, max-fs, max-cpb, max-dpb, and max-br:
>>             These parameters MAY be used to signal the capabilities of a
>>             receiver implementation. These parameters MUST NOT be used
>>             for any other purpose.  The highest level conveyed in the
>>             value of the profile-level-id parameter or the max-recv-
>> level
>>             parameter MUST be such that the receiver is fully capable of
>>             supporting.
>>
>> -- Tom
>>
>> On 10/02/2010 03:08 PM, Mo Zanaty (mzanaty) wrote:
>>> The draft is missing more than just "or equal to".
>>>
>>>> The value of max-smbps MUST be greater than the value of MaxMBPS
>> given in
>>>> Table A-1 of [1] for the highest level.
>>>
>>> It should also take max-mbps into account. And the reference to
>> "highest level" has no antecedent in this section; it is in the prior
>> section on max-mbps.
>>>
>>> I suggest this [new] text:
>>>
>>> "The value of max-smbps MUST be greater than [or equal to] the value
>> of MaxMBPS given [explicitly in max-mbps or implicitly] in Table A-1 of
>> [1] for the highest level [given in profile-level-id or max-recv-
>> level]."
>>>
>>> Regards,
>>> Mo
>>>
>>> -----Original Message-----
>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Roni Even
>>> Sent: Friday, October 01, 2010 4:49 PM
>>> To: 'IETF AVT WG'
>>> Cc: 'Randell Jesup'; 'Tom Kristensen'; keith.drage@alcatel-lucent.com
>>> Subject: Re: [AVT] rfc3984bis - max-smbps limit
>>>
>>> Hi,
>>>
>>> We got the following comment as a private email (bellow) . The draft
>> has not
>>> be published yet and we can do this change before publication.
>>> Please review and send comments to the list since it changes the
>> current
>>> semantics. If there are no objections we will do this change.
>>> Since this parameter has been added in the bis draft and was not part
>> of RFC
>>> 3984 we do not see any interoperability issues.
>>>
>>> Thanks
>>> Roni Even
>>> As a co-author of the draft
>>>
>>>
>>> Email from Tom Kristensen
>>>
>>> I discovered missing alignment between H.241 MaxStaticMBPS and max-
>> smbps as
>>> defined in draft-ietf-avt-rtp-rfc3984bis.
>>>
>>>    From Table 8-9, H.241:
>>> ---
>>> The encoder should recompute this value for each picture.
>>> The value of (MaxStaticMBPS ª 500) shall not be less than the value
>> MaxMBPS
>>> for the Level given in Table A-1/H.264, and if CustomMaxMBPS is
>> signalled,
>>> shall not be less than the value (CustomMaxMBPS ª 500).
>>> ---
>>>
>>> On the other hand in the SIP world, rfc3984bis says:
>>> ---
>>> The value of max-smbps MUST be greater than the value of MaxMBPS
>> given in
>>> Table A-1 of [1] for the highest level.
>>> ---
>>> http://tools.ietf.org/html/draft-ietf-avt-rtp-rfc3984bis-11#page-44
>>>
>>>
>>> H.241 says      max-smbps>= max-mbps.
>>> rfc3984bis says max-smbps>    max-mbps.
>>> We are missing "or equal".
>>>
>>> Of course it doesn't really make much meaning to include a max-smbps
>>> equal to max-mbps, but we have a mismatch between H.241 and the SDP
>>> counterpart.
>>>
>>> Is it possible and feasible to alter the text given the current state
>> of
>>> soon-to-be-RFC draft-ietf-avt-rtp-rfc3984bis?
>>>
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>


From magnus.westerlund@ericsson.com  Tue Oct  5 01:27:51 2010
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Ali C. Begen (abegen) skrev 2010-10-04 18:04:
> I am glad that we are converging on something.
> 
>>>>>> 3. I worried that what is intended to be an RAMS-R update from the
>>>>>> client will be interpreted as new RAMS-R request. The reason is the only
>>>>>> that separates these two cases are timing, does it arrive before the
>>>>>> burst ends or not. Relying on this heuristics is quite weak and error
>>>>>> prone. I still wished that an sequence number had been added to RAMS-R.
>>>>>
>>>>> I gave enough arguments why we should not use a seqnum for the RAMS-R messages earlier. It is not justified IMO.
>> Based
>>>> on the SSRC of the primary and the requesting client's CNAME, things are pretty straightforward.
>>>>
>>>> Yes, I know I am in the rough here. From my perspective, it makes the
>>>> use of update RAMS-R uncertain to use. Maybe not a big issue if most
>>>> doesn't implement updates. But if one finds need for it then there will
>>>> be issues. Consider the AD and IESG notified about the potential issue.
>>>
>>> Is the updated text above good for you?
>>
>> Not, really, because it has only clarified the issue. The issue is
>> really the uncertainty for the client how its request will be
>> interpreted, either as updated request or a new request, all depending
>> on timing. The biggest issue is the updated request which might be
>> interpreted as new request, when the client is likely satisfied and
> 
> Why is this a problem? For this to happen, the initial request must have been lost. And if it is lost and the second makes it, the client still gets what it wants. I don't see a problem here.

No, if you intended to send an update and that gets handled as new
request, then you get the unintended consequence of getting a second burst.

> 
>> might even have gone one to join the multicast group when the
>> RAMS-R(update) is processed as a new RAMS-R. Then causing congestion and
> 
> Are you saying this:
> 1- The client makes a request but it gets lost. In the mean time, the client sends an update and the server thinks it is a new request and starts sending the burst

No,
1. the client sends a request intended to update an ongoing burst.

2. The RAMS-R (update) arrive to late. Thus triggering a second burst.

3. In the mean time, client gets an RAMS-I and joins the multicast group

4. Burst + mcast creates a congestion.

> 
>> unwanted traffic. The client that detects this can surely terminate the
>> burst, but that will be after some delay, and traffic has arrived.
> 
> OK, so you are saying what I wrote above. Well, the client has to send a RAMS-T and upon receiving it, the server stops the burst. So that is not a big deal. If you are concerned about RAMS-T being lost (which is repeated if the bust is not stopped), then I will just remind you that this is a protocol where control messages are not fully reliable (they are just repeated for redundancy as frequently as 4585 allows). If you are really this unlucky, you will have a problem but it will be only temporary.
>

I will agree that you can stop the burst, but you have biased your
protocol by design against sending any RAMS-R with the purpose of
updating the parameter because other things happens than what you
intended to. Thus it will in most cases be better to not send an RAMS-R
update message and avoid the risk of tripping a second burst.


> I don't think there is anything here that requires for us to burn our energy. FWIW, your proposal would not solve this problem entirely, either.

I agree that a RAMS-R sequence number would not resolve the issue you
described. However, it resolves the issue I have tried to describe, I
hope the above makes it clear.

> 
>>>
>>>>>
>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globally
>>>>>>         unique CNAME identifier."
[snip]

>>>
>>>>>> If I understand the impact of a CNAME collision is that the collision
>>>>>> clients request will be mixed up, for example terminating each others
>>>>>> request, or update the values in the RAMS-R.
>>>>>
>>>>> When they are unique, this won't happen.
>>>>
>>>> Just checking, but if that is the case then I am missing a discussion of
>>>> hijacking attacks in the security consideration section by guessing your
>>>> targets CNAME.
>>>
>>> We should probably mention this but I am not sure how the server can deal with this. Hijacking is not easy since the attack
>> must take place at the same instant (more or less) with the request from the authentic client. One of your family members
>> can probably do this :)
>>
>> The real solution is where you have an more permanent id system in place
>> that you can connect through source authentication of the requests.
>>
>> In an SSM session that uses simple feedback model the RTP_Rx cname may
>> leak as they are redistributed.
>>
>> Based on that you could bombard a BRS with RAMS-T for example for all
>> known CNAMES and do that in a round-robin fashion across channels and
>> time. Depending on source address spoofing you will more or less easy to
>> find. But I do agree that it becomes a little bit more a brute force
>> attack, but an attacker could gain knowledge about an important piece of
>> information to mount the attack at all.
> 
> SRTCP?

SRTCP keyed with unique keys for each client will prevent anyone else to
send RAMS-T to terminate a burst you have initiated.

> 
>>>>>
>>>>>> 7. Section 7.3:
>>>>>> "The MSN value SHALL
>>>>>>       be set to zero only when a new RAMS request is received."
>>>>>>
>>>>>> How is that actually known? And why reset it at all? Why not increase if
>>>>>> for each new RAMS-I message with new content, independently if it is an
>>>>>> update or a new request.
>>>>>
>>>>> If this is relating to a new burst request, then it is reset. Ow, what is the point of having a seqnum? If something has
>>>> changed compared to the previous RAMS-I, then MSN is incremented. If it is just a re-xmit, MSN stays the same.
>>>>
>>>> I fully agree with the need for separating retransmissions from updates.
>>>> However, I wonder over the reset of the field for each new RAMS-R. It
>>>> appears to me to be more robust to simply increment it rather than
>>>> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN = 0.
>>>
>>> I think we discussed this before. The RAMS-R numbers are no way correlated with the RAMS-I numbers. You are still trying
>> to correlate them.
>>
>> Nope, the number here are still only to indicate that they are different
>> to get the sequence right. My point is that the client can determine
>> based on MSN if it is an repeat or a new RAMS-I based on a new request.
> 
> When the client receives a RAMS-I with MSN=0, it knows that RAMS-I was sent for a RAMS-R message that was received the first time by the server or an identical repeat of the initial RAMS-I message.
> 
> But even if the client sends an updated request, the server may ignore it, may ignore the changes and subsequently repeat the earlier RAMS-I with no changes in it, which will have the previous MSN. The server may not send anything at all or the message may get lost. The client cannot assume the changes it requested were honored by the server UNLESS there was an updated RAMS-I from the server. Even in that case, the RAMS-I changes may be due to other things the server has observed - not the changes the client asked.
>
> So, the client should not really read too much in to the MSN values received. That is what I have been trying to explain in this discussion.

Also in this case I don't think we have been considering the same case.
My case was the following.

1. C->S RAMS-R
2a. S->C RAMS-I (MSN=0)
2b. S->C Burst starts
3. C->S RAMS-R(Intended to update first RAMS-R)
4. S: Burst ends
5. S: RAMS-R from step 3 arrives in server and trigger new burst
6. S->C RAMS-I (MSN=0)
7. S->C Second burst transmitted

When the RAMS-I message from step 6 arrives the client may think this is
the same as the one in 2a.

Are you assuming that there is a 4b RAMS-I message which indicates that
the first burst will be terminated that has MSN=1? What if that is lost
or not sent?

> 
>>>
>>>> Then a RAMS-R(2) that is intended to be an update but becomes an new
>>>> request results in an RAMS-I with MSN = 0. The client will not know if
>>>> this is an retransmission of RAMS-R(1) info. The updated should result
>>>> in MSN=1. So without comparing the RAMS-I you can't determine if there
>>>
>>> The client checks the RAMS-I seqnum to see whether it is a repeat or new info. If RAMS-I MSN is zero, that is the first
>> RAMS-I anyway so it must be fully parsed. Does not matter which RAMS-R actually generated it since that is the info from the
>> server until an updated RAMS-I is received. This is how the protocol works.
>>
>> As I try to explain there is a case where you can get two RAMS-I with
>> MSN=0 in a row with different information. Thus not providing any
>> relieve for the client in the need to compare the whole request with the
>> previous one.
> 
> So what? If you made a single request and received two RAMS-I messages with MSN=0, that means they are identical. No need to compare them. If you made two requests and received two RAMS-I messages with MSN=0, they are different messages and you need to fully read them anyway.

Okay, so your point is that as soon you have sent more than one RAMS-R
message to a BRS you will need to look at all RAMS-I and the MSN becomes
completely useless. But, then I think the document needs to point out
that MSN is only reliable to detect repeat transmissions as long as you
have sent no additional RAMS-R messages during a minute or so.

>>>>>> 14. Section 10:
>>>>>>
>>>>>> Shouldn't the security consideration make it clear that RAMS-R are
>>>>>> especially suspectible to Replay attacks as there is no information in
>>>>>> the packet that one can use to detect that it is out of sequence.
>>>>>
>>>>> There is a wording about this in that section (which simply refers the reader to 5760). Are you considering a RAMS-
>> specific
>>>> replay attack here?
>>>>
>>>>
>>>> Yes, I am considering that it is easy to target RAMS-R specifically for
>>>> an replay attack. Especially when sent in a reduced size RTCP packet
>>>> only containing RAMS-R and SDES CNAME. That has no time specific
>>>> information and all replay detection must happen in the security protocol.
>>>
>>> The Token stuff (or the cookie stuff) will avoid request messages from being sent by others (it will do at least a reverse path
>> check). Beyond that is not SRTCP a good solution to avoid this?
>>
>> SRTCP is if keyed correctly a good solution yes.
>>
>>  My point is that one normally document the vulnerability and then
>> ensure that one has a remedy for it. When it comes to the token stuff I
>> am not certain that it helps against a replay attack. Unless the token
>> has expired on the server you still will get traffic to the target. That
>> is why I think a reverse path check when you start delivering should
>> happen. Because first in that case if there is no listener at the
>> indicated address and port will you find out that you are attempting a
>> delivery not intended.
> 
> Token provides RPF check. But, let's work on this Sec. section together. I could use some help.
> 

Ok

-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From gonzalo.camarillo@ericsson.com  Tue Oct  5 02:00:29 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi Colin,

> Unless their policy has changed recently, RTP is one of the few well- 
> known acronyms the RFC editor allows to be used without expansion in  
> document titles.

yes, it is:

https://www.rfc-editor.org/rfc-style-guide/abbrev.expansion.txt

Anyway, for all other acronyms in the draft, the following rule (also
taken from the link above) applies:

"Editorial guidelines for the RFC series generally require expansion of
these abbreviations on first occurrence in a title, in an abstract, or
in the body of the document."

Thanks,

Gonzalo

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Dropping the ietf list off.

> > Are you saying this:
> > 1- The client makes a request but it gets lost. In the mean time, =
the client sends an update and the server thinks it is a new
> request and starts sending the burst
>=20
> No,
> 1. the client sends a request intended to update an ongoing burst.
>=20
> 2. The RAMS-R (update) arrive to late. Thus triggering a second burst.

Oh, Ok. Now I see your scenario (update arriving late and causing a new =
burst to start). Well, this is an ill scenario with mid-session updates. =
While the chances are pretty slim for this happen, it can happen. For =
this to happen, the client must really send an update close to the join =
time and then the RAMS-R gets delayed for some reason.

The only solution for this is that the client will send a RAMS-T for the =
second burst as soon as it detects it. Furthermore, if the primary =
stream packets have a higher priority than the burst packets (and they =
should if the network supports this), mcast stream won't be hurt.

FWIW, mid-sessions updates are not problem-free. So if someone wants to =
implement it, they will have to live with the complications.
=20
> 3. In the mean time, client gets an RAMS-I and joins the multicast =
group
>=20
> 4. Burst + mcast creates a congestion.
>=20
> >
> >> unwanted traffic. The client that detects this can surely terminate =
the
> >> burst, but that will be after some delay, and traffic has arrived.
> >
> > OK, so you are saying what I wrote above. Well, the client has to =
send a RAMS-T and upon receiving it, the server stops the
> burst. So that is not a big deal. If you are concerned about RAMS-T =
being lost (which is repeated if the bust is not stopped),
> then I will just remind you that this is a protocol where control =
messages are not fully reliable (they are just repeated for
> redundancy as frequently as 4585 allows). If you are really this =
unlucky, you will have a problem but it will be only
> temporary.
> >
>=20
> I will agree that you can stop the burst, but you have biased your
> protocol by design against sending any RAMS-R with the purpose of
> updating the parameter because other things happens than what you
> intended to. Thus it will in most cases be better to not send an =
RAMS-R
> update message and avoid the risk of tripping a second burst.

Personally, I don't see the much benefit in using the RAMS-R updates. =
That's why they are optional for those who want to implement them.
=20
>=20
> > I don't think there is anything here that requires for us to burn =
our energy. FWIW, your proposal would not solve this
> problem entirely, either.
>=20
> I agree that a RAMS-R sequence number would not resolve the issue you
> described. However, it resolves the issue I have tried to describe, I
> hope the above makes it clear.
>=20
> >
> >>>
> >>>>>
> >>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a =
globally
> >>>>>>         unique CNAME identifier."
> [snip]
>=20
> >>>
> >>>>>> If I understand the impact of a CNAME collision is that the =
collision
> >>>>>> clients request will be mixed up, for example terminating each =
others
> >>>>>> request, or update the values in the RAMS-R.
> >>>>>
> >>>>> When they are unique, this won't happen.
> >>>>
> >>>> Just checking, but if that is the case then I am missing a =
discussion of
> >>>> hijacking attacks in the security consideration section by =
guessing your
> >>>> targets CNAME.
> >>>
> >>> We should probably mention this but I am not sure how the server =
can deal with this. Hijacking is not easy since the
> attack
> >> must take place at the same instant (more or less) with the request =
from the authentic client. One of your family
> members
> >> can probably do this :)
> >>
> >> The real solution is where you have an more permanent id system in =
place
> >> that you can connect through source authentication of the requests.
> >>
> >> In an SSM session that uses simple feedback model the RTP_Rx cname =
may
> >> leak as they are redistributed.
> >>
> >> Based on that you could bombard a BRS with RAMS-T for example for =
all
> >> known CNAMES and do that in a round-robin fashion across channels =
and
> >> time. Depending on source address spoofing you will more or less =
easy to
> >> find. But I do agree that it becomes a little bit more a brute =
force
> >> attack, but an attacker could gain knowledge about an important =
piece of
> >> information to mount the attack at all.
> >
> > SRTCP?
>=20
> SRTCP keyed with unique keys for each client will prevent anyone else =
to
> send RAMS-T to terminate a burst you have initiated.

OK.
=20
> >
> >>>>>
> >>>>>> 7. Section 7.3:
> >>>>>> "The MSN value SHALL
> >>>>>>       be set to zero only when a new RAMS request is received."
> >>>>>>
> >>>>>> How is that actually known? And why reset it at all? Why not =
increase if
> >>>>>> for each new RAMS-I message with new content, independently if =
it is an
> >>>>>> update or a new request.
> >>>>>
> >>>>> If this is relating to a new burst request, then it is reset. =
Ow, what is the point of having a seqnum? If something has
> >>>> changed compared to the previous RAMS-I, then MSN is incremented. =
If it is just a re-xmit, MSN stays the same.
> >>>>
> >>>> I fully agree with the need for separating retransmissions from =
updates.
> >>>> However, I wonder over the reset of the field for each new =
RAMS-R. It
> >>>> appears to me to be more robust to simply increment it rather =
than
> >>>> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN =
=3D 0.
> >>>
> >>> I think we discussed this before. The RAMS-R numbers are no way =
correlated with the RAMS-I numbers. You are still
> trying
> >> to correlate them.
> >>
> >> Nope, the number here are still only to indicate that they are =
different
> >> to get the sequence right. My point is that the client can =
determine
> >> based on MSN if it is an repeat or a new RAMS-I based on a new =
request.
> >
> > When the client receives a RAMS-I with MSN=3D0, it knows that RAMS-I =
was sent for a RAMS-R message that was received
> the first time by the server or an identical repeat of the initial =
RAMS-I message.
> >
> > But even if the client sends an updated request, the server may =
ignore it, may ignore the changes and subsequently repeat
> the earlier RAMS-I with no changes in it, which will have the previous =
MSN. The server may not send anything at all or the
> message may get lost. The client cannot assume the changes it =
requested were honored by the server UNLESS there was an
> updated RAMS-I from the server. Even in that case, the RAMS-I changes =
may be due to other things the server has observed -
> not the changes the client asked.
> >
> > So, the client should not really read too much in to the MSN values =
received. That is what I have been trying to explain in
> this discussion.
>=20
> Also in this case I don't think we have been considering the same =
case.
> My case was the following.
>=20
> 1. C->S RAMS-R
> 2a. S->C RAMS-I (MSN=3D0)
> 2b. S->C Burst starts
> 3. C->S RAMS-R(Intended to update first RAMS-R)
> 4. S: Burst ends
> 5. S: RAMS-R from step 3 arrives in server and trigger new burst
> 6. S->C RAMS-I (MSN=3D0)
> 7. S->C Second burst transmitted
>=20
> When the RAMS-I message from step 6 arrives the client may think this =
is
> the same as the one in 2a.
>=20
> Are you assuming that there is a 4b RAMS-I message which indicates =
that
> the first burst will be terminated that has MSN=3D1? What if that is =
lost
> or not sent?

Well that RAMS-I is not necessarily sent. And if sent, it may get lost. =
Again, this is an ill scenario with RAMS-R updates. But here the problem =
is the second burst not failing to identify the second RAMS-I. When the =
2nd burst starts, the client will figure out, things were screwed up.
=20
> >
> >>>
> >>>> Then a RAMS-R(2) that is intended to be an update but becomes an =
new
> >>>> request results in an RAMS-I with MSN =3D 0. The client will not =
know if
> >>>> this is an retransmission of RAMS-R(1) info. The updated should =
result
> >>>> in MSN=3D1. So without comparing the RAMS-I you can't determine =
if there
> >>>
> >>> The client checks the RAMS-I seqnum to see whether it is a repeat =
or new info. If RAMS-I MSN is zero, that is the first
> >> RAMS-I anyway so it must be fully parsed. Does not matter which =
RAMS-R actually generated it since that is the info from
> the
> >> server until an updated RAMS-I is received. This is how the =
protocol works.
> >>
> >> As I try to explain there is a case where you can get two RAMS-I =
with
> >> MSN=3D0 in a row with different information. Thus not providing any
> >> relieve for the client in the need to compare the whole request =
with the
> >> previous one.
> >
> > So what? If you made a single request and received two RAMS-I =
messages with MSN=3D0, that means they are identical. No
> need to compare them. If you made two requests and received two RAMS-I =
messages with MSN=3D0, they are different
> messages and you need to fully read them anyway.
>=20
> Okay, so your point is that as soon you have sent more than one RAMS-R
> message to a BRS you will need to look at all RAMS-I and the MSN =
becomes
> completely useless. But, then I think the document needs to point out
> that MSN is only reliable to detect repeat transmissions as long as =
you
> have sent no additional RAMS-R messages during a minute or so.

We should put some text about this. IF the client sent RAMS-R update(s), =
it should probably check every RAMS-I regardless of MSN values.

-acbegen


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Hi,

I would say that it is a failure of the design, to not make RAMS-R
updates work well. I see one fix that would seem easy to avoid these
problems. Create a new message type RAMS-U that is identical to RAMS-R
except the message type and restrict it to work on a ongoing burst. The
other way is to rip out update for now to avoid having this obviously
broken mechanism stay in there. And if someone needs it in the future
then define it properly.

I do want this finished up and published, but not with serious issue
part of the spec.

Magnus




Ali C. Begen (abegen) skrev 2010-10-06 10:09:
> Dropping the ietf list off.
> 
>>> Are you saying this:
>>> 1- The client makes a request but it gets lost. In the mean time, the client sends an update and the server thinks it is a new
>> request and starts sending the burst
>>
>> No,
>> 1. the client sends a request intended to update an ongoing burst.
>>
>> 2. The RAMS-R (update) arrive to late. Thus triggering a second burst.
> 
> Oh, Ok. Now I see your scenario (update arriving late and causing a new burst to start). Well, this is an ill scenario with mid-session updates. While the chances are pretty slim for this happen, it can happen. For this to happen, the client must really send an update close to the join time and then the RAMS-R gets delayed for some reason.
> 
> The only solution for this is that the client will send a RAMS-T for the second burst as soon as it detects it. Furthermore, if the primary stream packets have a higher priority than the burst packets (and they should if the network supports this), mcast stream won't be hurt.
> 
> FWIW, mid-sessions updates are not problem-free. So if someone wants to implement it, they will have to live with the complications.
>  
>> 3. In the mean time, client gets an RAMS-I and joins the multicast group
>>
>> 4. Burst + mcast creates a congestion.
>>
>>>
>>>> unwanted traffic. The client that detects this can surely terminate the
>>>> burst, but that will be after some delay, and traffic has arrived.
>>>
>>> OK, so you are saying what I wrote above. Well, the client has to send a RAMS-T and upon receiving it, the server stops the
>> burst. So that is not a big deal. If you are concerned about RAMS-T being lost (which is repeated if the bust is not stopped),
>> then I will just remind you that this is a protocol where control messages are not fully reliable (they are just repeated for
>> redundancy as frequently as 4585 allows). If you are really this unlucky, you will have a problem but it will be only
>> temporary.
>>>
>>
>> I will agree that you can stop the burst, but you have biased your
>> protocol by design against sending any RAMS-R with the purpose of
>> updating the parameter because other things happens than what you
>> intended to. Thus it will in most cases be better to not send an RAMS-R
>> update message and avoid the risk of tripping a second burst.
> 
> Personally, I don't see the much benefit in using the RAMS-R updates. That's why they are optional for those who want to implement them.
>  
>>
>>> I don't think there is anything here that requires for us to burn our energy. FWIW, your proposal would not solve this
>> problem entirely, either.
>>
>> I agree that a RAMS-R sequence number would not resolve the issue you
>> described. However, it resolves the issue I have tried to describe, I
>> hope the above makes it clear.
>>
>>>
>>>>>
>>>>>>>
>>>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globally
>>>>>>>>         unique CNAME identifier."
>> [snip]
>>
>>>>>
>>>>>>>> If I understand the impact of a CNAME collision is that the collision
>>>>>>>> clients request will be mixed up, for example terminating each others
>>>>>>>> request, or update the values in the RAMS-R.
>>>>>>>
>>>>>>> When they are unique, this won't happen.
>>>>>>
>>>>>> Just checking, but if that is the case then I am missing a discussion of
>>>>>> hijacking attacks in the security consideration section by guessing your
>>>>>> targets CNAME.
>>>>>
>>>>> We should probably mention this but I am not sure how the server can deal with this. Hijacking is not easy since the
>> attack
>>>> must take place at the same instant (more or less) with the request from the authentic client. One of your family
>> members
>>>> can probably do this :)
>>>>
>>>> The real solution is where you have an more permanent id system in place
>>>> that you can connect through source authentication of the requests.
>>>>
>>>> In an SSM session that uses simple feedback model the RTP_Rx cname may
>>>> leak as they are redistributed.
>>>>
>>>> Based on that you could bombard a BRS with RAMS-T for example for all
>>>> known CNAMES and do that in a round-robin fashion across channels and
>>>> time. Depending on source address spoofing you will more or less easy to
>>>> find. But I do agree that it becomes a little bit more a brute force
>>>> attack, but an attacker could gain knowledge about an important piece of
>>>> information to mount the attack at all.
>>>
>>> SRTCP?
>>
>> SRTCP keyed with unique keys for each client will prevent anyone else to
>> send RAMS-T to terminate a burst you have initiated.
> 
> OK.
>  
>>>
>>>>>>>
>>>>>>>> 7. Section 7.3:
>>>>>>>> "The MSN value SHALL
>>>>>>>>       be set to zero only when a new RAMS request is received."
>>>>>>>>
>>>>>>>> How is that actually known? And why reset it at all? Why not increase if
>>>>>>>> for each new RAMS-I message with new content, independently if it is an
>>>>>>>> update or a new request.
>>>>>>>
>>>>>>> If this is relating to a new burst request, then it is reset. Ow, what is the point of having a seqnum? If something has
>>>>>> changed compared to the previous RAMS-I, then MSN is incremented. If it is just a re-xmit, MSN stays the same.
>>>>>>
>>>>>> I fully agree with the need for separating retransmissions from updates.
>>>>>> However, I wonder over the reset of the field for each new RAMS-R. It
>>>>>> appears to me to be more robust to simply increment it rather than
>>>>>> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN = 0.
>>>>>
>>>>> I think we discussed this before. The RAMS-R numbers are no way correlated with the RAMS-I numbers. You are still
>> trying
>>>> to correlate them.
>>>>
>>>> Nope, the number here are still only to indicate that they are different
>>>> to get the sequence right. My point is that the client can determine
>>>> based on MSN if it is an repeat or a new RAMS-I based on a new request.
>>>
>>> When the client receives a RAMS-I with MSN=0, it knows that RAMS-I was sent for a RAMS-R message that was received
>> the first time by the server or an identical repeat of the initial RAMS-I message.
>>>
>>> But even if the client sends an updated request, the server may ignore it, may ignore the changes and subsequently repeat
>> the earlier RAMS-I with no changes in it, which will have the previous MSN. The server may not send anything at all or the
>> message may get lost. The client cannot assume the changes it requested were honored by the server UNLESS there was an
>> updated RAMS-I from the server. Even in that case, the RAMS-I changes may be due to other things the server has observed -
>> not the changes the client asked.
>>>
>>> So, the client should not really read too much in to the MSN values received. That is what I have been trying to explain in
>> this discussion.
>>
>> Also in this case I don't think we have been considering the same case.
>> My case was the following.
>>
>> 1. C->S RAMS-R
>> 2a. S->C RAMS-I (MSN=0)
>> 2b. S->C Burst starts
>> 3. C->S RAMS-R(Intended to update first RAMS-R)
>> 4. S: Burst ends
>> 5. S: RAMS-R from step 3 arrives in server and trigger new burst
>> 6. S->C RAMS-I (MSN=0)
>> 7. S->C Second burst transmitted
>>
>> When the RAMS-I message from step 6 arrives the client may think this is
>> the same as the one in 2a.
>>
>> Are you assuming that there is a 4b RAMS-I message which indicates that
>> the first burst will be terminated that has MSN=1? What if that is lost
>> or not sent?
> 
> Well that RAMS-I is not necessarily sent. And if sent, it may get lost. Again, this is an ill scenario with RAMS-R updates. But here the problem is the second burst not failing to identify the second RAMS-I. When the 2nd burst starts, the client will figure out, things were screwed up.
>  
>>>
>>>>>
>>>>>> Then a RAMS-R(2) that is intended to be an update but becomes an new
>>>>>> request results in an RAMS-I with MSN = 0. The client will not know if
>>>>>> this is an retransmission of RAMS-R(1) info. The updated should result
>>>>>> in MSN=1. So without comparing the RAMS-I you can't determine if there
>>>>>
>>>>> The client checks the RAMS-I seqnum to see whether it is a repeat or new info. If RAMS-I MSN is zero, that is the first
>>>> RAMS-I anyway so it must be fully parsed. Does not matter which RAMS-R actually generated it since that is the info from
>> the
>>>> server until an updated RAMS-I is received. This is how the protocol works.
>>>>
>>>> As I try to explain there is a case where you can get two RAMS-I with
>>>> MSN=0 in a row with different information. Thus not providing any
>>>> relieve for the client in the need to compare the whole request with the
>>>> previous one.
>>>
>>> So what? If you made a single request and received two RAMS-I messages with MSN=0, that means they are identical. No
>> need to compare them. If you made two requests and received two RAMS-I messages with MSN=0, they are different
>> messages and you need to fully read them anyway.
>>
>> Okay, so your point is that as soon you have sent more than one RAMS-R
>> message to a BRS you will need to look at all RAMS-I and the MSN becomes
>> completely useless. But, then I think the document needs to point out
>> that MSN is only reliable to detect repeat transmissions as long as you
>> have sent no additional RAMS-R messages during a minute or so.
> 
> We should put some text about this. IF the client sent RAMS-R update(s), it should probably check every RAMS-I regardless of MSN values.
> 
> -acbegen
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for H.264 Reduced-Complexity Decoding Operation (RCDO) Video
	Author(s)       : T. Kristensen, P. Luthi
	Filename        : draft-ietf-avt-rtp-h264-rcdo-07.txt
	Pages           : 23
	Date            : 2010-10-06

This document describes an RTP payload format for the Reduced-
Complexity Decoding Operation (RCDO) for H.264 Baseline profile
bitstreams, as specified in ITU-T Recommendation H.241.  RCDO reduces
the decoding cost and resource consumption of the video processing.
The RCDO RTP payload format is based on the H.264 RTP payload format.

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Subject: Re: [AVT] AD review: draft-ietf-avt-rtp-h264-rcdo-06
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On 4 October 2010 12:19, Gonzalo Camarillo
<Gonzalo.Camarillo@ericsson.com> wrote:
> Hi,
>
> as part of a load balancing exercise between Robert and I, I will be
> acting as the responsible AD for this draft.
>
> https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-h264-rcdo/
>
> Below you can find my AD review. As soon as the authors address these
> comments (it should be trivial to address them), I will IETF Last Call
> this draft.

Great, thanks! I just submitted a new version addressing these
comments as detailed below.

[...]
> The acronyms RTP and RCDO need to be expanded in the title of the
> draft. In general, all acronyms need to be expanded on their first
> use.

RTP kept as is, cf. answer to SVC draft on this list. Other "not
well-known" acronyms are now expanded.

> The document refers to [3] as RFC3984bis. Given that [3] is a
> normative reference, this document will not be published until [3] is
> published as well. So, referring to it as, say, RFCYYYY and then get
> the RFC Editor to update the text seems like a better option than
> calling it RFC3984bis.

Done. We also have to remember syncing the identical parts of the
media subtype definition with draft-ietf-avt-rtp-rfc3984bis-11 when
it's 100% sure this will not change.

> Page 19: section X should be section 9.

Fixed.

> Section 9: "although if suitable the usage of SRTP [11] is
> recommended". Should that be a normative RECOMMENDED instead?

Not so sure, so in spirit of draft-ietf-avt-srtp-not-mandatory the
following sentence is used instead:

"Usage of data origin authentication and data integrity protection of
at least the RTP packet is RECOMMENDED, for example by use of the
Secure Real-time Transport Protocol (SRTP) <xref target=3D"RFC3711"/>."

> The draft should have a reference to the SDP spec: RFC 4566.

Yes, now referenced in the Introduction section.

-- Tom

--=20
# Cisco
## tomkrist@cisco.com =A0| =A0http://www.tandberg.com
### http://folk.uio.no/tomkri/

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Thread-Topic: Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to	Proposed Standard
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to	Proposed Standard
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, October 06, 2010 10:21 AM
> To: Ali C. Begen (abegen)
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; =
avt@ietf.org
> Subject: Re: Last Call: =
<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid =
Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>=20
> Hi,
>=20
> I would say that it is a failure of the design, to not make RAMS-R
> updates work well. I see one fix that would seem easy to avoid these
> problems. Create a new message type RAMS-U that is identical to RAMS-R
> except the message type and restrict it to work on a ongoing burst. =
The

I don't think this is a good design choice. When the updates feature was =
desired by some folks, we only agreed to have it if it the messages =
still stayed "self-defining". Now, when a server receives a RAMS-U =
message but it does not have an ongoing burst since the first message =
(RAMS-R) was lost, it won't process it based on your definition.

> other way is to rip out update for now to avoid having this obviously
> broken mechanism stay in there. And if someone needs it in the future
> then define it properly.

I think we can still keep the current text (about the update process) =
for those who are still planning to use it by putting a caution against =
such problems. Would that work for you?

-acbegen
=20
> I do want this finished up and published, but not with serious issue
> part of the spec.
>=20
> Magnus
>=20
>=20
>=20
>=20
> Ali C. Begen (abegen) skrev 2010-10-06 10:09:
> > Dropping the ietf list off.
> >
> >>> Are you saying this:
> >>> 1- The client makes a request but it gets lost. In the mean time, =
the client sends an update and the server thinks it is a
> new
> >> request and starts sending the burst
> >>
> >> No,
> >> 1. the client sends a request intended to update an ongoing burst.
> >>
> >> 2. The RAMS-R (update) arrive to late. Thus triggering a second =
burst.
> >
> > Oh, Ok. Now I see your scenario (update arriving late and causing a =
new burst to start). Well, this is an ill scenario with
> mid-session updates. While the chances are pretty slim for this =
happen, it can happen. For this to happen, the client must
> really send an update close to the join time and then the RAMS-R gets =
delayed for some reason.
> >
> > The only solution for this is that the client will send a RAMS-T for =
the second burst as soon as it detects it. Furthermore, if
> the primary stream packets have a higher priority than the burst =
packets (and they should if the network supports this),
> mcast stream won't be hurt.
> >
> > FWIW, mid-sessions updates are not problem-free. So if someone wants =
to implement it, they will have to live with the
> complications.
> >
> >> 3. In the mean time, client gets an RAMS-I and joins the multicast =
group
> >>
> >> 4. Burst + mcast creates a congestion.
> >>
> >>>
> >>>> unwanted traffic. The client that detects this can surely =
terminate the
> >>>> burst, but that will be after some delay, and traffic has =
arrived.
> >>>
> >>> OK, so you are saying what I wrote above. Well, the client has to =
send a RAMS-T and upon receiving it, the server stops
> the
> >> burst. So that is not a big deal. If you are concerned about RAMS-T =
being lost (which is repeated if the bust is not stopped),
> >> then I will just remind you that this is a protocol where control =
messages are not fully reliable (they are just repeated for
> >> redundancy as frequently as 4585 allows). If you are really this =
unlucky, you will have a problem but it will be only
> >> temporary.
> >>>
> >>
> >> I will agree that you can stop the burst, but you have biased your
> >> protocol by design against sending any RAMS-R with the purpose of
> >> updating the parameter because other things happens than what you
> >> intended to. Thus it will in most cases be better to not send an =
RAMS-R
> >> update message and avoid the risk of tripping a second burst.
> >
> > Personally, I don't see the much benefit in using the RAMS-R =
updates. That's why they are optional for those who want to
> implement them.
> >
> >>
> >>> I don't think there is anything here that requires for us to burn =
our energy. FWIW, your proposal would not solve this
> >> problem entirely, either.
> >>
> >> I agree that a RAMS-R sequence number would not resolve the issue =
you
> >> described. However, it resolves the issue I have tried to describe, =
I
> >> hope the above makes it clear.
> >>
> >>>
> >>>>>
> >>>>>>>
> >>>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a =
globally
> >>>>>>>>         unique CNAME identifier."
> >> [snip]
> >>
> >>>>>
> >>>>>>>> If I understand the impact of a CNAME collision is that the =
collision
> >>>>>>>> clients request will be mixed up, for example terminating =
each others
> >>>>>>>> request, or update the values in the RAMS-R.
> >>>>>>>
> >>>>>>> When they are unique, this won't happen.
> >>>>>>
> >>>>>> Just checking, but if that is the case then I am missing a =
discussion of
> >>>>>> hijacking attacks in the security consideration section by =
guessing your
> >>>>>> targets CNAME.
> >>>>>
> >>>>> We should probably mention this but I am not sure how the server =
can deal with this. Hijacking is not easy since the
> >> attack
> >>>> must take place at the same instant (more or less) with the =
request from the authentic client. One of your family
> >> members
> >>>> can probably do this :)
> >>>>
> >>>> The real solution is where you have an more permanent id system =
in place
> >>>> that you can connect through source authentication of the =
requests.
> >>>>
> >>>> In an SSM session that uses simple feedback model the RTP_Rx =
cname may
> >>>> leak as they are redistributed.
> >>>>
> >>>> Based on that you could bombard a BRS with RAMS-T for example for =
all
> >>>> known CNAMES and do that in a round-robin fashion across channels =
and
> >>>> time. Depending on source address spoofing you will more or less =
easy to
> >>>> find. But I do agree that it becomes a little bit more a brute =
force
> >>>> attack, but an attacker could gain knowledge about an important =
piece of
> >>>> information to mount the attack at all.
> >>>
> >>> SRTCP?
> >>
> >> SRTCP keyed with unique keys for each client will prevent anyone =
else to
> >> send RAMS-T to terminate a burst you have initiated.
> >
> > OK.
> >
> >>>
> >>>>>>>
> >>>>>>>> 7. Section 7.3:
> >>>>>>>> "The MSN value SHALL
> >>>>>>>>       be set to zero only when a new RAMS request is =
received."
> >>>>>>>>
> >>>>>>>> How is that actually known? And why reset it at all? Why not =
increase if
> >>>>>>>> for each new RAMS-I message with new content, independently =
if it is an
> >>>>>>>> update or a new request.
> >>>>>>>
> >>>>>>> If this is relating to a new burst request, then it is reset. =
Ow, what is the point of having a seqnum? If something has
> >>>>>> changed compared to the previous RAMS-I, then MSN is =
incremented. If it is just a re-xmit, MSN stays the same.
> >>>>>>
> >>>>>> I fully agree with the need for separating retransmissions from =
updates.
> >>>>>> However, I wonder over the reset of the field for each new =
RAMS-R. It
> >>>>>> appears to me to be more robust to simply increment it rather =
than
> >>>>>> reset. Otherwise you can send RAMS-R(1) resulting in RAMS-I MSN =
=3D 0.
> >>>>>
> >>>>> I think we discussed this before. The RAMS-R numbers are no way =
correlated with the RAMS-I numbers. You are still
> >> trying
> >>>> to correlate them.
> >>>>
> >>>> Nope, the number here are still only to indicate that they are =
different
> >>>> to get the sequence right. My point is that the client can =
determine
> >>>> based on MSN if it is an repeat or a new RAMS-I based on a new =
request.
> >>>
> >>> When the client receives a RAMS-I with MSN=3D0, it knows that =
RAMS-I was sent for a RAMS-R message that was received
> >> the first time by the server or an identical repeat of the initial =
RAMS-I message.
> >>>
> >>> But even if the client sends an updated request, the server may =
ignore it, may ignore the changes and subsequently
> repeat
> >> the earlier RAMS-I with no changes in it, which will have the =
previous MSN. The server may not send anything at all or the
> >> message may get lost. The client cannot assume the changes it =
requested were honored by the server UNLESS there was
> an
> >> updated RAMS-I from the server. Even in that case, the RAMS-I =
changes may be due to other things the server has
> observed -
> >> not the changes the client asked.
> >>>
> >>> So, the client should not really read too much in to the MSN =
values received. That is what I have been trying to explain in
> >> this discussion.
> >>
> >> Also in this case I don't think we have been considering the same =
case.
> >> My case was the following.
> >>
> >> 1. C->S RAMS-R
> >> 2a. S->C RAMS-I (MSN=3D0)
> >> 2b. S->C Burst starts
> >> 3. C->S RAMS-R(Intended to update first RAMS-R)
> >> 4. S: Burst ends
> >> 5. S: RAMS-R from step 3 arrives in server and trigger new burst
> >> 6. S->C RAMS-I (MSN=3D0)
> >> 7. S->C Second burst transmitted
> >>
> >> When the RAMS-I message from step 6 arrives the client may think =
this is
> >> the same as the one in 2a.
> >>
> >> Are you assuming that there is a 4b RAMS-I message which indicates =
that
> >> the first burst will be terminated that has MSN=3D1? What if that =
is lost
> >> or not sent?
> >
> > Well that RAMS-I is not necessarily sent. And if sent, it may get =
lost. Again, this is an ill scenario with RAMS-R updates. But
> here the problem is the second burst not failing to identify the =
second RAMS-I. When the 2nd burst starts, the client will
> figure out, things were screwed up.
> >
> >>>
> >>>>>
> >>>>>> Then a RAMS-R(2) that is intended to be an update but becomes =
an new
> >>>>>> request results in an RAMS-I with MSN =3D 0. The client will =
not know if
> >>>>>> this is an retransmission of RAMS-R(1) info. The updated should =
result
> >>>>>> in MSN=3D1. So without comparing the RAMS-I you can't determine =
if there
> >>>>>
> >>>>> The client checks the RAMS-I seqnum to see whether it is a =
repeat or new info. If RAMS-I MSN is zero, that is the first
> >>>> RAMS-I anyway so it must be fully parsed. Does not matter which =
RAMS-R actually generated it since that is the info
> from
> >> the
> >>>> server until an updated RAMS-I is received. This is how the =
protocol works.
> >>>>
> >>>> As I try to explain there is a case where you can get two RAMS-I =
with
> >>>> MSN=3D0 in a row with different information. Thus not providing =
any
> >>>> relieve for the client in the need to compare the whole request =
with the
> >>>> previous one.
> >>>
> >>> So what? If you made a single request and received two RAMS-I =
messages with MSN=3D0, that means they are identical.
> No
> >> need to compare them. If you made two requests and received two =
RAMS-I messages with MSN=3D0, they are different
> >> messages and you need to fully read them anyway.
> >>
> >> Okay, so your point is that as soon you have sent more than one =
RAMS-R
> >> message to a BRS you will need to look at all RAMS-I and the MSN =
becomes
> >> completely useless. But, then I think the document needs to point =
out
> >> that MSN is only reliable to detect repeat transmissions as long as =
you
> >> have sent no additional RAMS-R messages during a minute or so.
> >
> > We should put some text about this. IF the client sent RAMS-R =
update(s), it should probably check every RAMS-I regardless
> of MSN values.
> >
> > -acbegen
> >
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------

From wwwrun@core3.amsl.com  Tue Oct  5 09:39:45 2010
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To: tom.kristensen@tandberg.com, yekuiwang@huawei.com, ron.even.tlv@gmail.com, rjesup@wgate.com
From: IETF Secretariat <ietf-ipr@ietf.org>
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Cc: keith.drage@alcatel-lucent.com, housley@vigilsec.com, gonzalo.camarillo@ericsson.com, avt@ietf.org, ipr-announce@ietf.org
Subject: [AVT] IPR Disclosure: Nokia Corporation's Statement about IPR related to draft-ietf-avt-rtp-rfc3984bis-11
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Dear Tom Kristensen, Ye-Kui Wang, Roni Even, Randell Jesup:

An IPR disclosure that pertains to your Internet-Draft entitled "RTP Payload
Format for H.264 Video" (draft-ietf-avt-rtp-rfc3984bis) was submitted to the
IETF Secretariat on 2010-10-05 and has been posted on the "IETF Page of
Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/1415/). The title of the IPR disclosure is
"Nokia Corporation's Statement about IPR related to
draft-ietf-avt-rtp-rfc3984bis-11."

The IETF Secretariat



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Co-chairs,=20

We are still waiting for your response about this issue. Will you let us =
know on how to proceed? It will be a shame if we cannot make a single =
bit progress between the meetings.

-acbegen

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Ali C. Begen (abegen)
> Sent: Wednesday, September 29, 2010 12:08 AM
> To: Even.roni@huawei.com; keith.drage@alcatel-lucent.com; avt@ietf.org
> Subject: [AVT] Token or Cookie?
>=20
> Co-chairs,
>=20
> The submission deadline is quickly approaching for the next meeting =
and as the authors we are still waiting for a green light
> to move forward on this issue. We have not made progress since the =
last meeting.
>=20
> Based on the discussion in the mailing list, I believe we will go with =
the Token approach but could you make it formal in the
> mailing list?
>=20
> Thanks.
> -acbegen
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

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Subject: Re: [AVT] AD review: draft-ietf-avt-rtp-h264-rcdo-06
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Hi Tom,

thanks for putting this new revision rogether. The next step is for me
to start its IETF LC. I will do that as soon as I have a minute.

Cheers,

Gonzalo

On 06/10/2010 5:50 PM, Tom Kristensen wrote:
> On 4 October 2010 12:19, Gonzalo Camarillo
> <Gonzalo.Camarillo@ericsson.com> wrote:
>> Hi,
>>
>> as part of a load balancing exercise between Robert and I, I will be
>> acting as the responsible AD for this draft.
>>
>> https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-h264-rcdo/
>>
>> Below you can find my AD review. As soon as the authors address these
>> comments (it should be trivial to address them), I will IETF Last Call
>> this draft.
> 
> Great, thanks! I just submitted a new version addressing these
> comments as detailed below.
> 
> [...]
>> The acronyms RTP and RCDO need to be expanded in the title of the
>> draft. In general, all acronyms need to be expanded on their first
>> use.
> 
> RTP kept as is, cf. answer to SVC draft on this list. Other "not
> well-known" acronyms are now expanded.
> 
>> The document refers to [3] as RFC3984bis. Given that [3] is a
>> normative reference, this document will not be published until [3] is
>> published as well. So, referring to it as, say, RFCYYYY and then get
>> the RFC Editor to update the text seems like a better option than
>> calling it RFC3984bis.
> 
> Done. We also have to remember syncing the identical parts of the
> media subtype definition with draft-ietf-avt-rtp-rfc3984bis-11 when
> it's 100% sure this will not change.
> 
>> Page 19: section X should be section 9.
> 
> Fixed.
> 
>> Section 9: "although if suitable the usage of SRTP [11] is
>> recommended". Should that be a normative RECOMMENDED instead?
> 
> Not so sure, so in spirit of draft-ietf-avt-srtp-not-mandatory the
> following sentence is used instead:
> 
> "Usage of data origin authentication and data integrity protection of
> at least the RTP packet is RECOMMENDED, for example by use of the
> Secure Real-time Transport Protocol (SRTP) <xref target="RFC3711"/>."
> 
>> The draft should have a reference to the SDP spec: RFC 4566.
> 
> Yes, now referenced in the Introduction section.
> 
> -- Tom
> 


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Subject: Re: [AVT] Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to	Proposed Standard
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Ali C. Begen (abegen) skrev 2010-10-06 17:53:
> 
> 
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Wednesday, October 06, 2010 10:21 AM
>> To: Ali C. Begen (abegen)
>> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; avt@ietf.org
>> Subject: Re: Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid Acquisition of Multicast RTP
>> Sessions) to Proposed Standard
>>
>> Hi,
>>
>> I would say that it is a failure of the design, to not make RAMS-R
>> updates work well. I see one fix that would seem easy to avoid these
>> problems. Create a new message type RAMS-U that is identical to RAMS-R
>> except the message type and restrict it to work on a ongoing burst. The
> 
> I don't think this is a good design choice. When the updates feature was desired by some folks, we only agreed to have it if it the messages still stayed "self-defining". Now, when a server receives a RAMS-U message but it does not have an ongoing burst since the first message (RAMS-R) was lost, it won't process it based on your definition.

Yes, you are right that design would have such a impact. It was clearly
not thought through in that dimension.

> 
>> other way is to rip out update for now to avoid having this obviously
>> broken mechanism stay in there. And if someone needs it in the future
>> then define it properly.
> 
> I think we can still keep the current text (about the update process) for those who are still planning to use it by putting a caution against such problems. Would that work for you?

I think such a warning is the minimal change. As it appears that I am
the only one that seems concerned despite the returning discussion of
this issue I will hesitantly accept that.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From tom.van_caenegem@alcatel-lucent.com  Thu Oct  7 03:30:04 2010
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Subject: Re: [AVT] Last Call:	<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt>	(Unicast-Based Rapid Acquisition of Multicast RTP Sessions)	to	Proposed Standard
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Hi,

I have never been in favour of the "UPDATE" usage of RAMS-R, and I guess we=
 did not think sufficiently through the corner cases that were created with=
 introducing/allowing it. Possibly a sequence number field for RAMS-R (and =
proper usage definition) could have avoided the corner case Magnus mentions=
. The least we should do is indeed mention the risks when a client uses RAM=
S-R which could be also covered in the section "failure cases". Maybe it is=
 also worthwhile to restrict the usage of sending  RAMS-R update message(s)=
 only after the RTP RX has received a RAMS-I message for the 1st RAMS-R mes=
sage sent.

I would be also in favour of having the possibility for a BRS to tell a RTP=
-Rx NOT to make use of RAMS-R update messages  (where for the same RAMS/bur=
st transaction, at least one TLV field is provided with a different value o=
r at least one TLV field is new, compared to the original RAMS-R). Such ind=
ication could be by means of defining e.g. a 202 response code (RAMS reques=
t has been accepted BUT no RAMS-R update messages allowed) in the 1st RAMS-=
I message.

Tom




-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ali C=
. Begen (abegen)
Sent: woensdag 6 oktober 2010 17:53
To: Magnus Westerlund
Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; avt@ietf.org
Subject: Re: [AVT] Last Call: <draft-ietf-avt-rapid-acquisition-for-rtp-15.=
txt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to Propose=
d Standard



> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, October 06, 2010 10:21 AM
> To: Ali C. Begen (abegen)
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org;
> avt@ietf.org
> Subject: Re: Last Call:
> <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid
> Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>
> Hi,
>
> I would say that it is a failure of the design, to not make RAMS-R
> updates work well. I see one fix that would seem easy to avoid these
> problems. Create a new message type RAMS-U that is identical to RAMS-R
> except the message type and restrict it to work on a ongoing burst.
> The

I don't think this is a good design choice. When the updates feature was de=
sired by some folks, we only agreed to have it if it the messages still sta=
yed "self-defining". Now, when a server receives a RAMS-U message but it do=
es not have an ongoing burst since the first message (RAMS-R) was lost, it =
won't process it based on your definition.

> other way is to rip out update for now to avoid having this obviously
> broken mechanism stay in there. And if someone needs it in the future
> then define it properly.

I think we can still keep the current text (about the update process) for t=
hose who are still planning to use it by putting a caution against such pro=
blems. Would that work for you?

-acbegen

> I do want this finished up and published, but not with serious issue
> part of the spec.
>
> Magnus
>
>
>
>
> Ali C. Begen (abegen) skrev 2010-10-06 10:09:
> > Dropping the ietf list off.
> >
> >>> Are you saying this:
> >>> 1- The client makes a request but it gets lost. In the mean time,
> >>> the client sends an update and the server thinks it is a
> new
> >> request and starts sending the burst
> >>
> >> No,
> >> 1. the client sends a request intended to update an ongoing burst.
> >>
> >> 2. The RAMS-R (update) arrive to late. Thus triggering a second burst.
> >
> > Oh, Ok. Now I see your scenario (update arriving late and causing a
> > new burst to start). Well, this is an ill scenario with
> mid-session updates. While the chances are pretty slim for this
> happen, it can happen. For this to happen, the client must really send an=
 update close to the join time and then the RAMS-R gets delayed for some re=
ason.
> >
> > The only solution for this is that the client will send a RAMS-T for
> > the second burst as soon as it detects it. Furthermore, if
> the primary stream packets have a higher priority than the burst
> packets (and they should if the network supports this), mcast stream won'=
t be hurt.
> >
> > FWIW, mid-sessions updates are not problem-free. So if someone wants
> > to implement it, they will have to live with the
> complications.
> >
> >> 3. In the mean time, client gets an RAMS-I and joins the multicast
> >> group
> >>
> >> 4. Burst + mcast creates a congestion.
> >>
> >>>
> >>>> unwanted traffic. The client that detects this can surely
> >>>> terminate the burst, but that will be after some delay, and traffic =
has arrived.
> >>>
> >>> OK, so you are saying what I wrote above. Well, the client has to
> >>> send a RAMS-T and upon receiving it, the server stops
> the
> >> burst. So that is not a big deal. If you are concerned about RAMS-T
> >> being lost (which is repeated if the bust is not stopped), then I
> >> will just remind you that this is a protocol where control messages
> >> are not fully reliable (they are just repeated for redundancy as frequ=
ently as 4585 allows). If you are really this unlucky, you will have a prob=
lem but it will be only temporary.
> >>>
> >>
> >> I will agree that you can stop the burst, but you have biased your
> >> protocol by design against sending any RAMS-R with the purpose of
> >> updating the parameter because other things happens than what you
> >> intended to. Thus it will in most cases be better to not send an
> >> RAMS-R update message and avoid the risk of tripping a second burst.
> >
> > Personally, I don't see the much benefit in using the RAMS-R
> > updates. That's why they are optional for those who want to
> implement them.
> >
> >>
> >>> I don't think there is anything here that requires for us to burn
> >>> our energy. FWIW, your proposal would not solve this
> >> problem entirely, either.
> >>
> >> I agree that a RAMS-R sequence number would not resolve the issue
> >> you described. However, it resolves the issue I have tried to
> >> describe, I hope the above makes it clear.
> >>
> >>>
> >>>>>
> >>>>>>>
> >>>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a globa=
lly
> >>>>>>>>         unique CNAME identifier."
> >> [snip]
> >>
> >>>>>
> >>>>>>>> If I understand the impact of a CNAME collision is that the
> >>>>>>>> collision clients request will be mixed up, for example
> >>>>>>>> terminating each others request, or update the values in the RAM=
S-R.
> >>>>>>>
> >>>>>>> When they are unique, this won't happen.
> >>>>>>
> >>>>>> Just checking, but if that is the case then I am missing a
> >>>>>> discussion of hijacking attacks in the security consideration
> >>>>>> section by guessing your targets CNAME.
> >>>>>
> >>>>> We should probably mention this but I am not sure how the server
> >>>>> can deal with this. Hijacking is not easy since the
> >> attack
> >>>> must take place at the same instant (more or less) with the
> >>>> request from the authentic client. One of your family
> >> members
> >>>> can probably do this :)
> >>>>
> >>>> The real solution is where you have an more permanent id system
> >>>> in place that you can connect through source authentication of the r=
equests.
> >>>>
> >>>> In an SSM session that uses simple feedback model the RTP_Rx
> >>>> cname may leak as they are redistributed.
> >>>>
> >>>> Based on that you could bombard a BRS with RAMS-T for example for
> >>>> all known CNAMES and do that in a round-robin fashion across
> >>>> channels and time. Depending on source address spoofing you will
> >>>> more or less easy to find. But I do agree that it becomes a
> >>>> little bit more a brute force attack, but an attacker could gain
> >>>> knowledge about an important piece of information to mount the attac=
k at all.
> >>>
> >>> SRTCP?
> >>
> >> SRTCP keyed with unique keys for each client will prevent anyone
> >> else to send RAMS-T to terminate a burst you have initiated.
> >
> > OK.
> >
> >>>
> >>>>>>>
> >>>>>>>> 7. Section 7.3:
> >>>>>>>> "The MSN value SHALL
> >>>>>>>>       be set to zero only when a new RAMS request is received."
> >>>>>>>>
> >>>>>>>> How is that actually known? And why reset it at all? Why not
> >>>>>>>> increase if for each new RAMS-I message with new content,
> >>>>>>>> independently if it is an update or a new request.
> >>>>>>>
> >>>>>>> If this is relating to a new burst request, then it is reset.
> >>>>>>> Ow, what is the point of having a seqnum? If something has
> >>>>>> changed compared to the previous RAMS-I, then MSN is incremented. =
If it is just a re-xmit, MSN stays the same.
> >>>>>>
> >>>>>> I fully agree with the need for separating retransmissions from up=
dates.
> >>>>>> However, I wonder over the reset of the field for each new
> >>>>>> RAMS-R. It appears to me to be more robust to simply increment
> >>>>>> it rather than reset. Otherwise you can send RAMS-R(1) resulting i=
n RAMS-I MSN =3D 0.
> >>>>>
> >>>>> I think we discussed this before. The RAMS-R numbers are no way
> >>>>> correlated with the RAMS-I numbers. You are still
> >> trying
> >>>> to correlate them.
> >>>>
> >>>> Nope, the number here are still only to indicate that they are
> >>>> different to get the sequence right. My point is that the client
> >>>> can determine based on MSN if it is an repeat or a new RAMS-I based =
on a new request.
> >>>
> >>> When the client receives a RAMS-I with MSN=3D0, it knows that RAMS-I
> >>> was sent for a RAMS-R message that was received
> >> the first time by the server or an identical repeat of the initial RAM=
S-I message.
> >>>
> >>> But even if the client sends an updated request, the server may
> >>> ignore it, may ignore the changes and subsequently
> repeat
> >> the earlier RAMS-I with no changes in it, which will have the
> >> previous MSN. The server may not send anything at all or the
> >> message may get lost. The client cannot assume the changes it
> >> requested were honored by the server UNLESS there was
> an
> >> updated RAMS-I from the server. Even in that case, the RAMS-I
> >> changes may be due to other things the server has
> observed -
> >> not the changes the client asked.
> >>>
> >>> So, the client should not really read too much in to the MSN
> >>> values received. That is what I have been trying to explain in
> >> this discussion.
> >>
> >> Also in this case I don't think we have been considering the same case=
.
> >> My case was the following.
> >>
> >> 1. C->S RAMS-R
> >> 2a. S->C RAMS-I (MSN=3D0)
> >> 2b. S->C Burst starts
> >> 3. C->S RAMS-R(Intended to update first RAMS-R) 4. S: Burst ends 5.
> >> S: RAMS-R from step 3 arrives in server and trigger new burst 6.
> >> S->C RAMS-I (MSN=3D0) 7. S->C Second burst transmitted
> >>
> >> When the RAMS-I message from step 6 arrives the client may think
> >> this is the same as the one in 2a.
> >>
> >> Are you assuming that there is a 4b RAMS-I message which indicates
> >> that the first burst will be terminated that has MSN=3D1? What if
> >> that is lost or not sent?
> >
> > Well that RAMS-I is not necessarily sent. And if sent, it may get
> > lost. Again, this is an ill scenario with RAMS-R updates. But
> here the problem is the second burst not failing to identify the
> second RAMS-I. When the 2nd burst starts, the client will figure out, thi=
ngs were screwed up.
> >
> >>>
> >>>>>
> >>>>>> Then a RAMS-R(2) that is intended to be an update but becomes
> >>>>>> an new request results in an RAMS-I with MSN =3D 0. The client
> >>>>>> will not know if this is an retransmission of RAMS-R(1) info.
> >>>>>> The updated should result in MSN=3D1. So without comparing the
> >>>>>> RAMS-I you can't determine if there
> >>>>>
> >>>>> The client checks the RAMS-I seqnum to see whether it is a
> >>>>> repeat or new info. If RAMS-I MSN is zero, that is the first
> >>>> RAMS-I anyway so it must be fully parsed. Does not matter which
> >>>> RAMS-R actually generated it since that is the info
> from
> >> the
> >>>> server until an updated RAMS-I is received. This is how the protocol=
 works.
> >>>>
> >>>> As I try to explain there is a case where you can get two RAMS-I
> >>>> with MSN=3D0 in a row with different information. Thus not
> >>>> providing any relieve for the client in the need to compare the
> >>>> whole request with the previous one.
> >>>
> >>> So what? If you made a single request and received two RAMS-I message=
s with MSN=3D0, that means they are identical.
> No
> >> need to compare them. If you made two requests and received two
> >> RAMS-I messages with MSN=3D0, they are different messages and you need=
 to fully read them anyway.
> >>
> >> Okay, so your point is that as soon you have sent more than one
> >> RAMS-R message to a BRS you will need to look at all RAMS-I and the
> >> MSN becomes completely useless. But, then I think the document
> >> needs to point out that MSN is only reliable to detect repeat
> >> transmissions as long as you have sent no additional RAMS-R messages d=
uring a minute or so.
> >
> > We should put some text about this. IF the client sent RAMS-R
> > update(s), it should probably check every RAMS-I regardless
> of MSN values.
> >
> > -acbegen
> >
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

From csp@csperkins.org  Thu Oct  7 04:55:50 2010
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From: Colin Perkins <csp@csperkins.org>
To: Qin Wu <sunseawq@huawei.com>
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Cc: Roland.Schott@telekom.de, avt@ietf.org
Subject: Re: [AVT] Fw: I-D Action:draft-wu-avt-rtcp-xr-quality-monitoring-03.txt
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Qin,

These changes generally look good, but I do still have some comments:

- Section 4.2 has been rewritten as a General Synchronisation Offset =20
Metrics Block. This is a good direction, but the resulting text still =20=

assumes that only two streams exist. In an RTP multimedia session =20
there can be an arbitrary number of streams, with the same RTCP CNAME, =20=

not just two. I would suggest that this block be written such that one =20=

stream reports a General Synchronisation Offset of zero, and other =20
streams with the same RTCP CNAME report their offset relative to that. =20=

Note that the streams being synchronised are not necessarily audio and =20=

video streams.

- Section 5.1 updates the definition of the Frame Type Indicator to be =20=

a Picture Type Indicator. The intent of the new definition is okay, =20
but the text is very unclear. Can you clarify? (same issue later)

- Unlike the other metrics, the TR 101 290 decodability metrics report =20=

block is entirely MPEG-2 specific. I suggest splitting this into a =20
separate draft, since it doesn=92t really relate to the other metrics =20=

defined in this draft.

Colin



On 28 Sep 2010, at 07:53, Qin Wu wrote:
> Hi,
> The new version of draft-wu-avt-rtcp-xr-quality-monitoring-03 is =20
> available at:
> =
http://www.ietf.org/internet-drafts/draft-wu-avt-rtcp-xr-quality-monitorin=
g-03.txt
> which probably address the comments early received on the list.
> Your further comments and suggestions are welcome!
>
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Tuesday, September 28, 2010 2:15 PM
> Subject: I-D Action:draft-wu-avt-rtcp-xr-quality-monitoring-03.txt
>
>
>> A New Internet-Draft is available from the on-line Internet-Drafts =20=

>> directories.
>>
>> Title           : RTP Control Protocol Extended Reports (RTCP XR) =20
>> Report Blocks for Real- time Video Quality Monitoring
>> Author(s)       : W. Wu, et al.
>> Filename        : draft-wu-avt-rtcp-xr-quality-monitoring-03.txt
>> Pages           : 26
>> Date            : 2010-09-27
>>
>> This document defines a set of RTP Control Protocol Extended Reports
>> (RTCP XR) Report Blocks and associated SDP parameters allowing the
>> report of video quality metrics, primarily for video applications of
>> RTP.
>>
>> A URL for this Internet-Draft is:
>> =
http://www.ietf.org/internet-drafts/draft-wu-avt-rtcp-xr-quality-monitorin=
g-03.txt


--=20
Colin Perkins
http://csperkins.org/




From Even.roni@huawei.com  Thu Oct  7 09:23:41 2010
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Date: Thu, 07 Oct 2010 18:21:45 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'Glen Zorn' <gwz@net-zen.net>, "'Ali C. Begen (abegen)'" <abegen@cisco.com>, "'DRAGE, Keith (Keith)'" <keith.drage@alcatel-lucent.com>
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Cc: 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] Consensus call - Token approach	withindraft-ietf-avt-ports-for-ucast-mcast-rtp
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Hi Glen,
I will try to clarify the issue.

The Port Mapping Between Unicast and Multicast RTP Sessions described in
http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast-rtp-02 was
discussed for a couple of IETF meeting and on the list. The initial approach
that was agreed after we had a breakout session in previous IETF meeting was
to use a cookie based solution even though it required more messages and you
needed a cookie for every RTP session. 


Before the last IETF meeting there were proposal to update the solution to a
token approach.
In the token approach:
- Tokens are port-independent (cookies are port-dependent) and only depend
on the client's IP. That is, once a client gets its token, it can use it
with any of its ports and with any Feedback Target.
- Tokens address the concern raised by Magnus
(http://www.ietf.org/mail-archive/web/avt/current/msg12617.html)
- No need for keep-alives

We had a discussion in Maastricht and the outcome is in
http://www.ietf.org/proceedings/78/minutes/avt.txt 

There is a concern about the token solution as mentioned in the minutes in
section 9 of
http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01 

My understanding is that people wanted a token based solution but there are
two approaches

1. Use the token only solution

2. Use a combination when if the request has only an IP address the response
will be a token and if it will have both IP address and port the response
will be a  cookie.

According to the minutes the support is for the second approach.

We were trying to ask the list if this approach is OK. 

It is true that the begen individual draft as currently written is not a
full replacement of the WG draft.

Roni Even
AVT co-chair






> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Glen Zorn
> Sent: Friday, September 17, 2010 8:18 AM
> To: 'Ali C. Begen (abegen)'; 'DRAGE, Keith (Keith)'
> Cc: 'IETF AVT WG'
> Subject: Re: [AVT] Consensus call - Token approach withindraft-ietf-
> avt-ports-for-ucast-mcast-rtp
> 
> Ali C. Begen (abegen) [mailto://abegen@cisco.com] writes:
> 
> ...
> 
> > > 1) Incorporate material from draft-begen-avt-token-for-portmapping-
> 01
> > into
> > > draft-ietf-avt-ports-for-ucast-mcast-rtp
> >
> > The token draft, IMHO, is good enough and addresses our issues
> > adequately. If we agree with this, we can simply continue working on
> it
> > and finish the work.
> >
> > If we disagree with this and think Cookie is what we need, then we
> > finish the Cookie draft.
> 
> Maybe I'm wrong but it sounds like you are proposing the wholesale
> replacement of the text of a WG document with that your own individual
> submission.  It seems to me that in order for this to be a reasonable
> course
> the replacement text would need to be far superior to the existing
> draft,
> not just good enough/adequate for the purpose, unless, of course, the
> existing WG draft is not good enough or inadequate; but if that is the
> case,
> it begs the question how it became a WG draft in the first place.
> Furthermore, I would expect that such an outcome would necessarily be
> the
> result of strong consensus in the WG, not merely the absence of
> complaint...
> 
> ...
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From iesg-secretary@ietf.org  Thu Oct  7 11:07:43 2010
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Subject: [AVT] Last Call: <draft-ietf-avt-rtp-h264-rcdo-07.txt> (RTP Payload Format	for H.264 Reduced-Complexity Decoding Operation (RCDO) Video)	to Proposed Standard
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The IESG has received a request from the Audio/Video Transport WG (avt)
to consider the following document:
- 'RTP Payload Format for H.264 Reduced-Complexity Decoding Operation
   (RCDO) Video'
  <draft-ietf-avt-rtp-h264-rcdo-07.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2010-10-21. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-h264-rcdo/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-h264-rcdo/


The following IPR Declarations may be related to this I-D:

/ipr/105/
/ipr/485/

From abegen@cisco.com  Thu Oct  7 15:05:45 2010
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Thread-Topic: [AVT] Last Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt>	(Unicast-Based Rapid Acquisition of Multicast RTP Sessions)	to	Proposed Standard
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Last Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt>	(Unicast-Based Rapid Acquisition of Multicast RTP Sessions)	to	Proposed Standard
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> -----Original Message-----
> From: Van Caenegem, Tom (Tom) =
[mailto:tom.van_caenegem@alcatel-lucent.com]
> Sent: Thursday, October 07, 2010 6:31 PM
> To: Ali C. Begen (abegen); Magnus Westerlund
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; =
avt@ietf.org
> Subject: RE: [AVT] Last =
Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based =
Rapid Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>=20
> Hi,
>=20
> I have never been in favour of the "UPDATE" usage of RAMS-R, and I =
guess we did not think sufficiently through the corner
> cases that were created with introducing/allowing it. Possibly a =
sequence number field for RAMS-R (and proper usage
> definition) could have avoided the corner case Magnus mentions. The =
least we should do is indeed mention the risks when a
> client uses RAMS-R which could be also covered in the section "failure =
cases". Maybe it is also worthwhile to restrict the
> usage of sending  RAMS-R update message(s) only after the RTP RX has =
received a RAMS-I message for the 1st RAMS-R
> message sent.
>=20
> I would be also in favour of having the possibility for a BRS to tell =
a RTP-Rx NOT to make use of RAMS-R update messages
> (where for the same RAMS/burst transaction, at least one TLV field is =
provided with a different value or at least one TLV field
> is new, compared to the original RAMS-R). Such indication could be by =
means of defining e.g. a 202 response code (RAMS
> request has been accepted BUT no RAMS-R update messages allowed) in =
the 1st RAMS-I message.

If the rams-updates is not signaled in SDP, the client is not supposed =
to send update messages. So, just don't put it in the SDP.

-acbegen
=20
> Tom
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Ali C. Begen (abegen)
> Sent: woensdag 6 oktober 2010 17:53
> To: Magnus Westerlund
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; =
avt@ietf.org
> Subject: Re: [AVT] Last Call: =
<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid =
Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>=20
>=20
>=20
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Wednesday, October 06, 2010 10:21 AM
> > To: Ali C. Begen (abegen)
> > Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org;
> > avt@ietf.org
> > Subject: Re: Last Call:
> > <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based =
Rapid
> > Acquisition of Multicast RTP
> > Sessions) to Proposed Standard
> >
> > Hi,
> >
> > I would say that it is a failure of the design, to not make RAMS-R
> > updates work well. I see one fix that would seem easy to avoid these
> > problems. Create a new message type RAMS-U that is identical to =
RAMS-R
> > except the message type and restrict it to work on a ongoing burst.
> > The
>=20
> I don't think this is a good design choice. When the updates feature =
was desired by some folks, we only agreed to have it if it
> the messages still stayed "self-defining". Now, when a server receives =
a RAMS-U message but it does not have an ongoing
> burst since the first message (RAMS-R) was lost, it won't process it =
based on your definition.
>=20
> > other way is to rip out update for now to avoid having this =
obviously
> > broken mechanism stay in there. And if someone needs it in the =
future
> > then define it properly.
>=20
> I think we can still keep the current text (about the update process) =
for those who are still planning to use it by putting a
> caution against such problems. Would that work for you?
>=20
> -acbegen
>=20
> > I do want this finished up and published, but not with serious issue
> > part of the spec.
> >
> > Magnus
> >
> >
> >
> >
> > Ali C. Begen (abegen) skrev 2010-10-06 10:09:
> > > Dropping the ietf list off.
> > >
> > >>> Are you saying this:
> > >>> 1- The client makes a request but it gets lost. In the mean =
time,
> > >>> the client sends an update and the server thinks it is a
> > new
> > >> request and starts sending the burst
> > >>
> > >> No,
> > >> 1. the client sends a request intended to update an ongoing =
burst.
> > >>
> > >> 2. The RAMS-R (update) arrive to late. Thus triggering a second =
burst.
> > >
> > > Oh, Ok. Now I see your scenario (update arriving late and causing =
a
> > > new burst to start). Well, this is an ill scenario with
> > mid-session updates. While the chances are pretty slim for this
> > happen, it can happen. For this to happen, the client must really =
send an update close to the join time and then the RAMS-R
> gets delayed for some reason.
> > >
> > > The only solution for this is that the client will send a RAMS-T =
for
> > > the second burst as soon as it detects it. Furthermore, if
> > the primary stream packets have a higher priority than the burst
> > packets (and they should if the network supports this), mcast stream =
won't be hurt.
> > >
> > > FWIW, mid-sessions updates are not problem-free. So if someone =
wants
> > > to implement it, they will have to live with the
> > complications.
> > >
> > >> 3. In the mean time, client gets an RAMS-I and joins the =
multicast
> > >> group
> > >>
> > >> 4. Burst + mcast creates a congestion.
> > >>
> > >>>
> > >>>> unwanted traffic. The client that detects this can surely
> > >>>> terminate the burst, but that will be after some delay, and =
traffic has arrived.
> > >>>
> > >>> OK, so you are saying what I wrote above. Well, the client has =
to
> > >>> send a RAMS-T and upon receiving it, the server stops
> > the
> > >> burst. So that is not a big deal. If you are concerned about =
RAMS-T
> > >> being lost (which is repeated if the bust is not stopped), then I
> > >> will just remind you that this is a protocol where control =
messages
> > >> are not fully reliable (they are just repeated for redundancy as =
frequently as 4585 allows). If you are really this unlucky,
> you will have a problem but it will be only temporary.
> > >>>
> > >>
> > >> I will agree that you can stop the burst, but you have biased =
your
> > >> protocol by design against sending any RAMS-R with the purpose of
> > >> updating the parameter because other things happens than what you
> > >> intended to. Thus it will in most cases be better to not send an
> > >> RAMS-R update message and avoid the risk of tripping a second =
burst.
> > >
> > > Personally, I don't see the much benefit in using the RAMS-R
> > > updates. That's why they are optional for those who want to
> > implement them.
> > >
> > >>
> > >>> I don't think there is anything here that requires for us to =
burn
> > >>> our energy. FWIW, your proposal would not solve this
> > >> problem entirely, either.
> > >>
> > >> I agree that a RAMS-R sequence number would not resolve the issue
> > >> you described. However, it resolves the issue I have tried to
> > >> describe, I hope the above makes it clear.
> > >>
> > >>>
> > >>>>>
> > >>>>>>>
> > >>>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a =
globally
> > >>>>>>>>         unique CNAME identifier."
> > >> [snip]
> > >>
> > >>>>>
> > >>>>>>>> If I understand the impact of a CNAME collision is that the
> > >>>>>>>> collision clients request will be mixed up, for example
> > >>>>>>>> terminating each others request, or update the values in =
the RAMS-R.
> > >>>>>>>
> > >>>>>>> When they are unique, this won't happen.
> > >>>>>>
> > >>>>>> Just checking, but if that is the case then I am missing a
> > >>>>>> discussion of hijacking attacks in the security consideration
> > >>>>>> section by guessing your targets CNAME.
> > >>>>>
> > >>>>> We should probably mention this but I am not sure how the =
server
> > >>>>> can deal with this. Hijacking is not easy since the
> > >> attack
> > >>>> must take place at the same instant (more or less) with the
> > >>>> request from the authentic client. One of your family
> > >> members
> > >>>> can probably do this :)
> > >>>>
> > >>>> The real solution is where you have an more permanent id system
> > >>>> in place that you can connect through source authentication of =
the requests.
> > >>>>
> > >>>> In an SSM session that uses simple feedback model the RTP_Rx
> > >>>> cname may leak as they are redistributed.
> > >>>>
> > >>>> Based on that you could bombard a BRS with RAMS-T for example =
for
> > >>>> all known CNAMES and do that in a round-robin fashion across
> > >>>> channels and time. Depending on source address spoofing you =
will
> > >>>> more or less easy to find. But I do agree that it becomes a
> > >>>> little bit more a brute force attack, but an attacker could =
gain
> > >>>> knowledge about an important piece of information to mount the =
attack at all.
> > >>>
> > >>> SRTCP?
> > >>
> > >> SRTCP keyed with unique keys for each client will prevent anyone
> > >> else to send RAMS-T to terminate a burst you have initiated.
> > >
> > > OK.
> > >
> > >>>
> > >>>>>>>
> > >>>>>>>> 7. Section 7.3:
> > >>>>>>>> "The MSN value SHALL
> > >>>>>>>>       be set to zero only when a new RAMS request is =
received."
> > >>>>>>>>
> > >>>>>>>> How is that actually known? And why reset it at all? Why =
not
> > >>>>>>>> increase if for each new RAMS-I message with new content,
> > >>>>>>>> independently if it is an update or a new request.
> > >>>>>>>
> > >>>>>>> If this is relating to a new burst request, then it is =
reset.
> > >>>>>>> Ow, what is the point of having a seqnum? If something has
> > >>>>>> changed compared to the previous RAMS-I, then MSN is =
incremented. If it is just a re-xmit, MSN stays the same.
> > >>>>>>
> > >>>>>> I fully agree with the need for separating retransmissions =
from updates.
> > >>>>>> However, I wonder over the reset of the field for each new
> > >>>>>> RAMS-R. It appears to me to be more robust to simply =
increment
> > >>>>>> it rather than reset. Otherwise you can send RAMS-R(1) =
resulting in RAMS-I MSN =3D 0.
> > >>>>>
> > >>>>> I think we discussed this before. The RAMS-R numbers are no =
way
> > >>>>> correlated with the RAMS-I numbers. You are still
> > >> trying
> > >>>> to correlate them.
> > >>>>
> > >>>> Nope, the number here are still only to indicate that they are
> > >>>> different to get the sequence right. My point is that the =
client
> > >>>> can determine based on MSN if it is an repeat or a new RAMS-I =
based on a new request.
> > >>>
> > >>> When the client receives a RAMS-I with MSN=3D0, it knows that =
RAMS-I
> > >>> was sent for a RAMS-R message that was received
> > >> the first time by the server or an identical repeat of the =
initial RAMS-I message.
> > >>>
> > >>> But even if the client sends an updated request, the server may
> > >>> ignore it, may ignore the changes and subsequently
> > repeat
> > >> the earlier RAMS-I with no changes in it, which will have the
> > >> previous MSN. The server may not send anything at all or the
> > >> message may get lost. The client cannot assume the changes it
> > >> requested were honored by the server UNLESS there was
> > an
> > >> updated RAMS-I from the server. Even in that case, the RAMS-I
> > >> changes may be due to other things the server has
> > observed -
> > >> not the changes the client asked.
> > >>>
> > >>> So, the client should not really read too much in to the MSN
> > >>> values received. That is what I have been trying to explain in
> > >> this discussion.
> > >>
> > >> Also in this case I don't think we have been considering the same =
case.
> > >> My case was the following.
> > >>
> > >> 1. C->S RAMS-R
> > >> 2a. S->C RAMS-I (MSN=3D0)
> > >> 2b. S->C Burst starts
> > >> 3. C->S RAMS-R(Intended to update first RAMS-R) 4. S: Burst ends =
5.
> > >> S: RAMS-R from step 3 arrives in server and trigger new burst 6.
> > >> S->C RAMS-I (MSN=3D0) 7. S->C Second burst transmitted
> > >>
> > >> When the RAMS-I message from step 6 arrives the client may think
> > >> this is the same as the one in 2a.
> > >>
> > >> Are you assuming that there is a 4b RAMS-I message which =
indicates
> > >> that the first burst will be terminated that has MSN=3D1? What if
> > >> that is lost or not sent?
> > >
> > > Well that RAMS-I is not necessarily sent. And if sent, it may get
> > > lost. Again, this is an ill scenario with RAMS-R updates. But
> > here the problem is the second burst not failing to identify the
> > second RAMS-I. When the 2nd burst starts, the client will figure =
out, things were screwed up.
> > >
> > >>>
> > >>>>>
> > >>>>>> Then a RAMS-R(2) that is intended to be an update but becomes
> > >>>>>> an new request results in an RAMS-I with MSN =3D 0. The =
client
> > >>>>>> will not know if this is an retransmission of RAMS-R(1) info.
> > >>>>>> The updated should result in MSN=3D1. So without comparing =
the
> > >>>>>> RAMS-I you can't determine if there
> > >>>>>
> > >>>>> The client checks the RAMS-I seqnum to see whether it is a
> > >>>>> repeat or new info. If RAMS-I MSN is zero, that is the first
> > >>>> RAMS-I anyway so it must be fully parsed. Does not matter which
> > >>>> RAMS-R actually generated it since that is the info
> > from
> > >> the
> > >>>> server until an updated RAMS-I is received. This is how the =
protocol works.
> > >>>>
> > >>>> As I try to explain there is a case where you can get two =
RAMS-I
> > >>>> with MSN=3D0 in a row with different information. Thus not
> > >>>> providing any relieve for the client in the need to compare the
> > >>>> whole request with the previous one.
> > >>>
> > >>> So what? If you made a single request and received two RAMS-I =
messages with MSN=3D0, that means they are identical.
> > No
> > >> need to compare them. If you made two requests and received two
> > >> RAMS-I messages with MSN=3D0, they are different messages and you =
need to fully read them anyway.
> > >>
> > >> Okay, so your point is that as soon you have sent more than one
> > >> RAMS-R message to a BRS you will need to look at all RAMS-I and =
the
> > >> MSN becomes completely useless. But, then I think the document
> > >> needs to point out that MSN is only reliable to detect repeat
> > >> transmissions as long as you have sent no additional RAMS-R =
messages during a minute or so.
> > >
> > > We should put some text about this. IF the client sent RAMS-R
> > > update(s), it should probably check every RAMS-I regardless
> > of MSN values.
> > >
> > > -acbegen
> > >
> > >
> >
> >
> > --
> >
> > Magnus Westerlund
> >
> > =
----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > =
----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > =
----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From tom.van_caenegem@alcatel-lucent.com  Fri Oct  8 01:20:40 2010
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Fri, 8 Oct 2010 10:19:30 +0200
Thread-Topic: [AVT] Last Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt>	(Unicast-Based Rapid Acquisition of Multicast RTP Sessions)	to	Proposed Standard
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Subject: Re: [AVT] Last Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt>	(Unicast-Based Rapid Acquisition of Multicast RTP Sessions)	to	Proposed Standard
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Ali,

My mistake, I forgot about the SDP att. This draft development and completi=
on process has taken (too) much time...
Tom

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]
Sent: vrijdag 8 oktober 2010 0:06
To: Van Caenegem, Tom (Tom); Magnus Westerlund
Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org; avt@ietf.org
Subject: RE: [AVT] Last Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.t=
xt> (Unicast-Based Rapid Acquisition of Multicast RTP Sessions) to Proposed=
 Standard



> -----Original Message-----
> From: Van Caenegem, Tom (Tom)
> [mailto:tom.van_caenegem@alcatel-lucent.com]
> Sent: Thursday, October 07, 2010 6:31 PM
> To: Ali C. Begen (abegen); Magnus Westerlund
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org;
> avt@ietf.org
> Subject: RE: [AVT] Last
> Call:<draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based
> Rapid Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>
> Hi,
>
> I have never been in favour of the "UPDATE" usage of RAMS-R, and I
> guess we did not think sufficiently through the corner cases that were
> created with introducing/allowing it. Possibly a sequence number field
> for RAMS-R (and proper usage
> definition) could have avoided the corner case Magnus mentions. The
> least we should do is indeed mention the risks when a client uses
> RAMS-R which could be also covered in the section "failure cases".
> Maybe it is also worthwhile to restrict the usage of sending  RAMS-R upda=
te message(s) only after the RTP RX has received a RAMS-I message for the 1=
st RAMS-R message sent.
>
> I would be also in favour of having the possibility for a BRS to tell
> a RTP-Rx NOT to make use of RAMS-R update messages (where for the same
> RAMS/burst transaction, at least one TLV field is provided with a
> different value or at least one TLV field is new, compared to the origina=
l RAMS-R). Such indication could be by means of defining e.g. a 202 respons=
e code (RAMS request has been accepted BUT no RAMS-R update messages allowe=
d) in the 1st RAMS-I message.

If the rams-updates is not signaled in SDP, the client is not supposed to s=
end update messages. So, just don't put it in the SDP.

-acbegen

> Tom
>
>
>
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Ali C. Begen (abegen)
> Sent: woensdag 6 oktober 2010 17:53
> To: Magnus Westerlund
> Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org;
> avt@ietf.org
> Subject: Re: [AVT] Last Call:
> <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based Rapid
> Acquisition of Multicast RTP
> Sessions) to Proposed Standard
>
>
>
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Wednesday, October 06, 2010 10:21 AM
> > To: Ali C. Begen (abegen)
> > Cc: draft-ietf-avt-rapid-acquisition-for-rtp@tools.ietf.org;
> > avt@ietf.org
> > Subject: Re: Last Call:
> > <draft-ietf-avt-rapid-acquisition-for-rtp-15.txt> (Unicast-Based
> > Rapid Acquisition of Multicast RTP
> > Sessions) to Proposed Standard
> >
> > Hi,
> >
> > I would say that it is a failure of the design, to not make RAMS-R
> > updates work well. I see one fix that would seem easy to avoid these
> > problems. Create a new message type RAMS-U that is identical to
> > RAMS-R except the message type and restrict it to work on a ongoing bur=
st.
> > The
>
> I don't think this is a good design choice. When the updates feature
> was desired by some folks, we only agreed to have it if it the
> messages still stayed "self-defining". Now, when a server receives a RAMS=
-U message but it does not have an ongoing burst since the first message (R=
AMS-R) was lost, it won't process it based on your definition.
>
> > other way is to rip out update for now to avoid having this
> > obviously broken mechanism stay in there. And if someone needs it in
> > the future then define it properly.
>
> I think we can still keep the current text (about the update process)
> for those who are still planning to use it by putting a caution against s=
uch problems. Would that work for you?
>
> -acbegen
>
> > I do want this finished up and published, but not with serious issue
> > part of the spec.
> >
> > Magnus
> >
> >
> >
> >
> > Ali C. Begen (abegen) skrev 2010-10-06 10:09:
> > > Dropping the ietf list off.
> > >
> > >>> Are you saying this:
> > >>> 1- The client makes a request but it gets lost. In the mean
> > >>> time, the client sends an update and the server thinks it is a
> > new
> > >> request and starts sending the burst
> > >>
> > >> No,
> > >> 1. the client sends a request intended to update an ongoing burst.
> > >>
> > >> 2. The RAMS-R (update) arrive to late. Thus triggering a second burs=
t.
> > >
> > > Oh, Ok. Now I see your scenario (update arriving late and causing
> > > a new burst to start). Well, this is an ill scenario with
> > mid-session updates. While the chances are pretty slim for this
> > happen, it can happen. For this to happen, the client must really
> > send an update close to the join time and then the RAMS-R
> gets delayed for some reason.
> > >
> > > The only solution for this is that the client will send a RAMS-T
> > > for the second burst as soon as it detects it. Furthermore, if
> > the primary stream packets have a higher priority than the burst
> > packets (and they should if the network supports this), mcast stream wo=
n't be hurt.
> > >
> > > FWIW, mid-sessions updates are not problem-free. So if someone
> > > wants to implement it, they will have to live with the
> > complications.
> > >
> > >> 3. In the mean time, client gets an RAMS-I and joins the
> > >> multicast group
> > >>
> > >> 4. Burst + mcast creates a congestion.
> > >>
> > >>>
> > >>>> unwanted traffic. The client that detects this can surely
> > >>>> terminate the burst, but that will be after some delay, and traffi=
c has arrived.
> > >>>
> > >>> OK, so you are saying what I wrote above. Well, the client has
> > >>> to send a RAMS-T and upon receiving it, the server stops
> > the
> > >> burst. So that is not a big deal. If you are concerned about
> > >> RAMS-T being lost (which is repeated if the bust is not stopped),
> > >> then I will just remind you that this is a protocol where control
> > >> messages are not fully reliable (they are just repeated for
> > >> redundancy as frequently as 4585 allows). If you are really this
> > >> unlucky,
> you will have a problem but it will be only temporary.
> > >>>
> > >>
> > >> I will agree that you can stop the burst, but you have biased
> > >> your protocol by design against sending any RAMS-R with the
> > >> purpose of updating the parameter because other things happens
> > >> than what you intended to. Thus it will in most cases be better
> > >> to not send an RAMS-R update message and avoid the risk of tripping =
a second burst.
> > >
> > > Personally, I don't see the much benefit in using the RAMS-R
> > > updates. That's why they are optional for those who want to
> > implement them.
> > >
> > >>
> > >>> I don't think there is anything here that requires for us to
> > >>> burn our energy. FWIW, your proposal would not solve this
> > >> problem entirely, either.
> > >>
> > >> I agree that a RAMS-R sequence number would not resolve the issue
> > >> you described. However, it resolves the issue I have tried to
> > >> describe, I hope the above makes it clear.
> > >>
> > >>>
> > >>>>>
> > >>>>>>>
> > >>>>>>>> 4. Section 6.2, bullet 5: " Thus, the RTP_Rx MUST choose a glo=
bally
> > >>>>>>>>         unique CNAME identifier."
> > >> [snip]
> > >>
> > >>>>>
> > >>>>>>>> If I understand the impact of a CNAME collision is that the
> > >>>>>>>> collision clients request will be mixed up, for example
> > >>>>>>>> terminating each others request, or update the values in the R=
AMS-R.
> > >>>>>>>
> > >>>>>>> When they are unique, this won't happen.
> > >>>>>>
> > >>>>>> Just checking, but if that is the case then I am missing a
> > >>>>>> discussion of hijacking attacks in the security consideration
> > >>>>>> section by guessing your targets CNAME.
> > >>>>>
> > >>>>> We should probably mention this but I am not sure how the
> > >>>>> server can deal with this. Hijacking is not easy since the
> > >> attack
> > >>>> must take place at the same instant (more or less) with the
> > >>>> request from the authentic client. One of your family
> > >> members
> > >>>> can probably do this :)
> > >>>>
> > >>>> The real solution is where you have an more permanent id system
> > >>>> in place that you can connect through source authentication of the=
 requests.
> > >>>>
> > >>>> In an SSM session that uses simple feedback model the RTP_Rx
> > >>>> cname may leak as they are redistributed.
> > >>>>
> > >>>> Based on that you could bombard a BRS with RAMS-T for example
> > >>>> for all known CNAMES and do that in a round-robin fashion
> > >>>> across channels and time. Depending on source address spoofing
> > >>>> you will more or less easy to find. But I do agree that it
> > >>>> becomes a little bit more a brute force attack, but an attacker
> > >>>> could gain knowledge about an important piece of information to mo=
unt the attack at all.
> > >>>
> > >>> SRTCP?
> > >>
> > >> SRTCP keyed with unique keys for each client will prevent anyone
> > >> else to send RAMS-T to terminate a burst you have initiated.
> > >
> > > OK.
> > >
> > >>>
> > >>>>>>>
> > >>>>>>>> 7. Section 7.3:
> > >>>>>>>> "The MSN value SHALL
> > >>>>>>>>       be set to zero only when a new RAMS request is received.=
"
> > >>>>>>>>
> > >>>>>>>> How is that actually known? And why reset it at all? Why
> > >>>>>>>> not increase if for each new RAMS-I message with new
> > >>>>>>>> content, independently if it is an update or a new request.
> > >>>>>>>
> > >>>>>>> If this is relating to a new burst request, then it is reset.
> > >>>>>>> Ow, what is the point of having a seqnum? If something has
> > >>>>>> changed compared to the previous RAMS-I, then MSN is incremented=
. If it is just a re-xmit, MSN stays the same.
> > >>>>>>
> > >>>>>> I fully agree with the need for separating retransmissions from =
updates.
> > >>>>>> However, I wonder over the reset of the field for each new
> > >>>>>> RAMS-R. It appears to me to be more robust to simply
> > >>>>>> increment it rather than reset. Otherwise you can send RAMS-R(1)=
 resulting in RAMS-I MSN =3D 0.
> > >>>>>
> > >>>>> I think we discussed this before. The RAMS-R numbers are no
> > >>>>> way correlated with the RAMS-I numbers. You are still
> > >> trying
> > >>>> to correlate them.
> > >>>>
> > >>>> Nope, the number here are still only to indicate that they are
> > >>>> different to get the sequence right. My point is that the
> > >>>> client can determine based on MSN if it is an repeat or a new RAMS=
-I based on a new request.
> > >>>
> > >>> When the client receives a RAMS-I with MSN=3D0, it knows that
> > >>> RAMS-I was sent for a RAMS-R message that was received
> > >> the first time by the server or an identical repeat of the initial R=
AMS-I message.
> > >>>
> > >>> But even if the client sends an updated request, the server may
> > >>> ignore it, may ignore the changes and subsequently
> > repeat
> > >> the earlier RAMS-I with no changes in it, which will have the
> > >> previous MSN. The server may not send anything at all or the
> > >> message may get lost. The client cannot assume the changes it
> > >> requested were honored by the server UNLESS there was
> > an
> > >> updated RAMS-I from the server. Even in that case, the RAMS-I
> > >> changes may be due to other things the server has
> > observed -
> > >> not the changes the client asked.
> > >>>
> > >>> So, the client should not really read too much in to the MSN
> > >>> values received. That is what I have been trying to explain in
> > >> this discussion.
> > >>
> > >> Also in this case I don't think we have been considering the same ca=
se.
> > >> My case was the following.
> > >>
> > >> 1. C->S RAMS-R
> > >> 2a. S->C RAMS-I (MSN=3D0)
> > >> 2b. S->C Burst starts
> > >> 3. C->S RAMS-R(Intended to update first RAMS-R) 4. S: Burst ends 5.
> > >> S: RAMS-R from step 3 arrives in server and trigger new burst 6.
> > >> S->C RAMS-I (MSN=3D0) 7. S->C Second burst transmitted
> > >>
> > >> When the RAMS-I message from step 6 arrives the client may think
> > >> this is the same as the one in 2a.
> > >>
> > >> Are you assuming that there is a 4b RAMS-I message which
> > >> indicates that the first burst will be terminated that has MSN=3D1?
> > >> What if that is lost or not sent?
> > >
> > > Well that RAMS-I is not necessarily sent. And if sent, it may get
> > > lost. Again, this is an ill scenario with RAMS-R updates. But
> > here the problem is the second burst not failing to identify the
> > second RAMS-I. When the 2nd burst starts, the client will figure out, t=
hings were screwed up.
> > >
> > >>>
> > >>>>>
> > >>>>>> Then a RAMS-R(2) that is intended to be an update but becomes
> > >>>>>> an new request results in an RAMS-I with MSN =3D 0. The client
> > >>>>>> will not know if this is an retransmission of RAMS-R(1) info.
> > >>>>>> The updated should result in MSN=3D1. So without comparing the
> > >>>>>> RAMS-I you can't determine if there
> > >>>>>
> > >>>>> The client checks the RAMS-I seqnum to see whether it is a
> > >>>>> repeat or new info. If RAMS-I MSN is zero, that is the first
> > >>>> RAMS-I anyway so it must be fully parsed. Does not matter which
> > >>>> RAMS-R actually generated it since that is the info
> > from
> > >> the
> > >>>> server until an updated RAMS-I is received. This is how the protoc=
ol works.
> > >>>>
> > >>>> As I try to explain there is a case where you can get two
> > >>>> RAMS-I with MSN=3D0 in a row with different information. Thus not
> > >>>> providing any relieve for the client in the need to compare the
> > >>>> whole request with the previous one.
> > >>>
> > >>> So what? If you made a single request and received two RAMS-I messa=
ges with MSN=3D0, that means they are identical.
> > No
> > >> need to compare them. If you made two requests and received two
> > >> RAMS-I messages with MSN=3D0, they are different messages and you ne=
ed to fully read them anyway.
> > >>
> > >> Okay, so your point is that as soon you have sent more than one
> > >> RAMS-R message to a BRS you will need to look at all RAMS-I and
> > >> the MSN becomes completely useless. But, then I think the
> > >> document needs to point out that MSN is only reliable to detect
> > >> repeat transmissions as long as you have sent no additional RAMS-R m=
essages during a minute or so.
> > >
> > > We should put some text about this. IF the client sent RAMS-R
> > > update(s), it should probably check every RAMS-I regardless
> > of MSN values.
> > >
> > > -acbegen
> > >
> > >
> >
> >
> > --
> >
> > Magnus Westerlund
> >
> > --------------------------------------------------------------------
> > -- Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > --------------------------------------------------------------------
> > --
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

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Subject: [AVT] I-D Action:draft-ietf-avt-rtp-svc-23.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for Scalable Video Coding
	Author(s)       : S. Wenger, et al.
	Filename        : draft-ietf-avt-rtp-svc-23.txt
	Pages           : 104
	Date            : 2010-10-08

This memo describes an RTP payload format for Scalable Video Coding
(SVC) as defined in_Annex G of ITU-T Recommendation H.264, which is
technically identical to Amendment 3 of ISO/IEC International
Standard 14496-10.  The RTP payload format allows for packetization
of one or more Network Abstraction Layer (NAL) units in each RTP
packet payload, as well as fragmentation of a NAL unit in multiple
RTP packets.  Furthermore, it supports transmission of an SVC stream
over a single as well as multiple RTP sessions.  The payload format
defines a new media subtype name "H264-SVC", but is still backwards
compatible to [I-D.ietf-avt-rtp-rfc3984bis] since the base layer,
when encapsulated in its own RTP stream, must use the H.264 media
subtype name ("H264") and the packetization method specified in [I-
D.ietf-avt-rtp-rfc3984bis].  The payload format has wide
applicability in videoconferencing, Internet video streaming, and
high bit-rate entertainment-quality video, among others.Table of Contents

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-svc-23.txt

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From yekuiwang@huawei.com  Fri Oct  8 01:35:23 2010
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Date: Fri, 08 Oct 2010 04:35:34 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
In-reply-to: <4CA9CE38.1040802@ericsson.com>
To: 'Gonzalo Camarillo' <Gonzalo.Camarillo@ericsson.com>, avt@ietf.org
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Subject: Re: [AVT] AD review: draft-ietf-avt-rtp-svc-22
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Hi Gonzalo,

Thanks for your AD reivew. I have just submitted -23 wherein all your
comments were addressed.

> The acronyms RTP and SVC need to be expanded in the title. 
> All acronyms in the draft need to be expanded on their first use.
> 

'SVC' expended in the title. 'RTP' was not, per your discussion with Colin.
I did not find any other acronyms that are not expanded on their first use.

> Section 1.2.2 has a few non-normative statements such as "SST 
> should be used...". This normatively specified at a later 
> point in the document (Section 4.4). If Section 1 is meant to 
> be a non-normative overview of the specification, the draft 
> should say so somewhere.
> 

Done, the last sentence before section 1.1 has been changed to: 
   A non-normative overview of the SVC codec and the payload is given
   in the remainder of this section.

> Page 80. When specifying behavior beyond RFC 3264, the draft 
> should use normative language. However, when specifying 
> behavior that is already specified in RFC 3264, the draft 
> should not use normative language. For example: "the offer 
> SHOULD also be used in the answer, as specified in 
> [RFC3264]". This issue also appears in other sections in the 
> document (e.g., in page 85). Simply s/SHOULD/should/ would 
> resolve the issue, for instance.

Done. I systematically checked all 'SHOULD', and did not find any other
places with this issue.

BR, YK  



From Even.roni@huawei.com  Fri Oct  8 01:49:07 2010
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Date: Fri, 08 Oct 2010 10:47:16 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'Roni Even' <Even.roni@huawei.com>
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Cc: "'DRAGE, Keith \(Keith\)'" <keith.drage@alcatel-lucent.com>, 'IETF AVT WG' <avt@ietf.org>, 'Dan Wing' <dwing@cisco.com>
Subject: [AVT] Revised Consensus call - On Token approach for draft-ietf-avt-ports-for-ucast-mcast-rtp
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Hi guys,
There were some concerns about how we drafted the consensus call so I would
like to try and give it another try.  Sorry for the long message which I
hope will clarify what are the proposed options. We would like to get
feedback by October 18th.


The Port Mapping Between Unicast and Multicast RTP Sessions described in
http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast-rtp-02 was
discussed for a couple of IETF meeting and on the list. The initial approach
that was agreed after we had a breakout session in previous IETF meetings
was to use a cookie based solution even though it required more messages and
you needed a cookie for every RTP session. 

Before the last IETF meeting there were two proposals to update the solution
to a token based approach.

In the token approach:
- Tokens are port-independent (cookies are port-dependent) and only depend
on the client's IP. That is, once a client gets its token, it can use it
with any of its ports and with any Feedback Target.
- Tokens address the concern raised by Magnus
(http://www.ietf.org/mail-archive/web/avt/current/msg12617.html)
- No need for keep-alives

We had a discussion in Maastricht and the outcome is in the meeting minutes
http://www.ietf.org/proceedings/78/minutes/avt.txt (see the section on
draft-begen-avt-token-for-portmapping). The token draft has Dan Wing (who is
also BEHAVE co-chair) as a co-author and he looked at the solution verifying
that it will work well with NATs


At the meeting the preference was for a token based solution but there are
were two approaches discussed.

1. Use the token only solution

2. Use a combination when if the request has only an IP address the response
will be a token and if it will have both IP address and port the response
will be a  cookie.

According to the minutes the support was for the second approach. The major
reason was a concern that carrier grade NATs can allocate more than one IP
address as the public address.

The authors of
http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01 suggest
that they addressed the concerns about the token only solution (see section
9 of the draft) and prefer to have a token only solution. This is a change
from the meeting minutes and this is the reason for the second question in
this consensus call.

So here are two questions for the group !!!

1. Do the people in the mailing list support the proposal to use a token
based solution instead on the cookie one. (this was the preference in the
consensus call in Maastricht) and we would like to confirm it on the list.

2. We have two options for the token approach, please give your preferences.

2.1 Token only solution as proposed in
http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
2.2 Token + cookie, in this case if the request will include only IP address
the response is a token, if it also includes a port it will be a cookie.
This approach is for cases were the NAT may use more than one IP address for
mapping a local address.


Please respond even if you did it on the previous consensus call.

Roni Even
AVT co-chair


From gonzalo.camarillo@ericsson.com  Fri Oct  8 04:06:12 2010
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Subject: Re: [AVT] AD review: draft-ietf-avt-rtp-svc-22
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Hi,

thanks for revising the draft. I have just requested its IETF LC.

Cheers,

Gonzalo

On 08/10/2010 11:35 AM, Ye-Kui Wang wrote:
> 
> Hi Gonzalo,
> 
> Thanks for your AD reivew. I have just submitted -23 wherein all your
> comments were addressed.
> 
>> The acronyms RTP and SVC need to be expanded in the title. 
>> All acronyms in the draft need to be expanded on their first use.
>>
> 
> 'SVC' expended in the title. 'RTP' was not, per your discussion with Colin.
> I did not find any other acronyms that are not expanded on their first use.
> 
>> Section 1.2.2 has a few non-normative statements such as "SST 
>> should be used...". This normatively specified at a later 
>> point in the document (Section 4.4). If Section 1 is meant to 
>> be a non-normative overview of the specification, the draft 
>> should say so somewhere.
>>
> 
> Done, the last sentence before section 1.1 has been changed to: 
>    A non-normative overview of the SVC codec and the payload is given
>    in the remainder of this section.
> 
>> Page 80. When specifying behavior beyond RFC 3264, the draft 
>> should use normative language. However, when specifying 
>> behavior that is already specified in RFC 3264, the draft 
>> should not use normative language. For example: "the offer 
>> SHOULD also be used in the answer, as specified in 
>> [RFC3264]". This issue also appears in other sections in the 
>> document (e.g., in page 85). Simply s/SHOULD/should/ would 
>> resolve the issue, for instance.
> 
> Done. I systematically checked all 'SHOULD', and did not find any other
> places with this issue.
> 
> BR, YK  
> 
> 


From iesg-secretary@ietf.org  Fri Oct  8 08:09:39 2010
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Subject: [AVT] Last Call: <draft-ietf-avt-rtp-svc-23.txt> (RTP Payload Format for	Scalable Video Coding) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport WG (avt)
to consider the following document:
- 'RTP Payload Format for Scalable Video Coding'
  <draft-ietf-avt-rtp-svc-23.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2010-10-22. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-svc/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-avt-rtp-svc/


The following IPR Declarations may be related to this I-D:

/ipr/24/
/ipr/105/
/ipr/286/
/ipr/451/
/ipr/485/
/ipr/496/
/ipr/502/
/ipr/675/
/ipr/678/
/ipr/697/
/ipr/832/
/ipr/852/
/ipr/854/
/ipr/910/
/ipr/939/
/ipr/996/
/ipr/1365/
/ipr/1415/
/ipr/1416/
/ipr/1417/
/ipr/1418/

From sunseawq@huawei.com  Fri Oct  8 21:48:23 2010
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Date: Sat, 09 Oct 2010 12:49:11 +0800
From: Qin Wu <sunseawq@huawei.com>
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Folks,

We have updated draft-wu-avt-retransmission-suppression-rtp which should now reflect the discussion we had since IETF78. 

The feedback suppression solution has also been greatly generalized comparing to the previous version.

Comments are solicited.

Regards!
-Qin
----- Original Message ----- 
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To: <i-d-announce@ietf.org>
Sent: Saturday, October 09, 2010 12:45 PM
Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt


>A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> Title           : RTCP Report Extension for Feedback Suppression
> Author(s)       : W. Wu, et al.
> Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> Pages           : 24
> Date            : 2010-10-08
> 
> This document specifies an extension to the RTCP feedback messages
> defined in the Audio-Visual Profile with Feedback (AVPF).  This
> extension allows an intermediate node in the network (such as an RTP
> translator) inform downstream receivers that sending a feedback
> message concerning a specified set of RTP messages may be
> unnecessary, or even harmful.  For example, in a source-specific
> multicast session with large fan-out, packet loss close to the media
> or distribution source of the session is detected as a loss by a
> large number of receivers.  The RTCP NACK messages used to request
> retransmission of the missing packets are all addressed to the media
> sender, or a designated feedback target.  This may result in a
> phenomenon known variously as a "NACK implosion" or "feedback storm".
> The Feedback Suppression message defined herein is used to notify
> receivers that packet loss was detected and that the sender of the
> report will either proactively recover, or no recovery is possible.
> Receivers respond to receipt of a feedback suppression message by not
> sending a feedback message (e.g. a NACK) that they otherwise would,
> This in turn reduces load on both the feedback target and on the
> network.  This document registers two new RTCP messages for Feedback
> Suppression.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-03.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for MVC Video
	Author(s)       : Y. Wang, T. Schierl
	Filename        : draft-ietf-avt-rtp-mvc-01.txt
	Pages           : 25
	Date            : 2010-10-09

This memo describes an RTP payload format for the multiview
extension of the ITU-T Recommendation H.264 video codec that is
technically identical to ISO/IEC International Standard 14496-10.
The RTP payload format allows for packetization of one or more
Network Abstraction Layer (NAL) units, produced by the video encoder,
in each RTP payload.  The payload format can be applied in RTP based
3D video transmissions such as such as 3D video streaming, free-
viewpoint video, and 3DTV.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mvc-01.txt

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ftp://ftp.ietf.org/internet-drafts/

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From yekuiwang@huawei.com  Sat Oct  9 02:31:23 2010
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Date: Sat, 09 Oct 2010 05:30:55 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
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Subject: Re: [AVT] I-D Action:draft-ietf-avt-rtp-mvc-01.txt
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Changes in relative to -00: 

Updated the NAL unit header syntax and semantics in section 3.3 per the
latest MVC specification, and made some other minor edits.

BR, YK 

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Internet-Drafts@ietf.org
> Sent: Saturday, October 09, 2010 5:15 AM
> To: i-d-announce@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] I-D Action:draft-ietf-avt-rtp-mvc-01.txt
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> This draft is a work item of the Audio/Video Transport 
> Working Group of the IETF.
> 
> 
> 	Title           : RTP Payload Format for MVC Video
> 	Author(s)       : Y. Wang, T. Schierl
> 	Filename        : draft-ietf-avt-rtp-mvc-01.txt
> 	Pages           : 25
> 	Date            : 2010-10-09
> 
> This memo describes an RTP payload format for the multiview 
> extension of the ITU-T Recommendation H.264 video codec that 
> is technically identical to ISO/IEC International Standard 14496-10.
> The RTP payload format allows for packetization of one or 
> more Network Abstraction Layer (NAL) units, produced by the 
> video encoder, in each RTP payload.  The payload format can 
> be applied in RTP based 3D video transmissions such as such 
> as 3D video streaming, free- viewpoint video, and 3DTV.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mvc-01.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail 
> reader implementation to automatically retrieve the ASCII 
> version of the Internet-Draft.
> 



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Subject: [AVT] I-D Action:draft-ietf-avt-rtp-rfc3984bis-12.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for H.264 Video
	Author(s)       : Y. Wang, et al.
	Filename        : draft-ietf-avt-rtp-rfc3984bis-12.txt
	Pages           : 104
	Date            : 2010-10-09

This memo describes an RTP Payload format for the ITU-T
Recommendation H.264 video codec and the technically identical
ISO/IEC International Standard 14496-10 video codec, excluding the
Scalable Video Coding (SVC) extension and the Multivew Video Coding
extension, for which the RTP payload formats are defined elsewhere.
The RTP payload format allows for packetization of one or more
Network Abstraction Layer Units (NALUs), produced by an H.264 video
encoder, in each RTP payload.  The payload format has wide
applicability, as it supports applications from simple low bit-rate
conversational usage, to Internet video streaming with interleaved
transmission, to high bit-rate video-on-demand.

This memo obsoletes RFC 3984.  Changes from RFC 3984 are summarized
in section 15.  Issues on backward compatibility to RFC 3984 are
discussed in section 14.



Table of Contents

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-rfc3984bis-12.txt

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From yekuiwang@huawei.com  Sat Oct  9 03:27:10 2010
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Date: Sat, 09 Oct 2010 06:27:09 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
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Subject: Re: [AVT] I-D Action:draft-ietf-avt-rtp-rfc3984bis-12.txt
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This revision addressed the max-smbps issue discussed in the list, with a
change from "The value of max-smbps MUST be greater than the value of ..."
to "The value of max-smbps MUST be equal to or greater than the value of
...", which was supported by a few persons and was not objected by anybody.
In addition, some minor editorial changes were made. 

BR, YK 

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Internet-Drafts@ietf.org
> Sent: Saturday, October 09, 2010 6:15 AM
> To: i-d-announce@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] I-D Action:draft-ietf-avt-rtp-rfc3984bis-12.txt
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> This draft is a work item of the Audio/Video Transport 
> Working Group of the IETF.
> 
> 
> 	Title           : RTP Payload Format for H.264 Video
> 	Author(s)       : Y. Wang, et al.
> 	Filename        : draft-ietf-avt-rtp-rfc3984bis-12.txt
> 	Pages           : 104
> 	Date            : 2010-10-09
> 
> This memo describes an RTP Payload format for the ITU-T 
> Recommendation H.264 video codec and the technically 
> identical ISO/IEC International Standard 14496-10 video 
> codec, excluding the Scalable Video Coding (SVC) extension 
> and the Multivew Video Coding extension, for which the RTP 
> payload formats are defined elsewhere.
> The RTP payload format allows for packetization of one or 
> more Network Abstraction Layer Units (NALUs), produced by an 
> H.264 video encoder, in each RTP payload.  The payload format 
> has wide applicability, as it supports applications from 
> simple low bit-rate conversational usage, to Internet video 
> streaming with interleaved transmission, to high bit-rate 
> video-on-demand.
> 
> This memo obsoletes RFC 3984.  Changes from RFC 3984 are 
> summarized in section 15.  Issues on backward compatibility 
> to RFC 3984 are discussed in section 14.
> 
> 
> 
> Table of Contents
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-rfc3984
> bis-12.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail 
> reader implementation to automatically retrieve the ASCII 
> version of the Internet-Draft.
> 



From abegen@cisco.com  Sat Oct  9 09:37:07 2010
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Thread-Topic: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Qin Wu" <sunseawq@huawei.com>, <avt@ietf.org>
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Comments:
1) Abstract is too long, could you shorten it?

Typos in abstract:
a) .. that *are* sending ...
b) RTP messages -> RTP packets
c) "the sender of the report will either proactively recover": Well, it =
does not have to be the sender of the report who will attempt the =
recovery. I would rephrase this.

2) Throughout the document, avoiding "sender" might be a good idea. For =
the source, use media or distribution source. Sender may mean different =
things in different contexts.

3) Intro 1st paragraph: The retransmission payload format is 4588 not =
4585.

4) Most of the discussion in the introduction section can be shortened =
in a concrete way by referring to other existing RFC in this area. I am =
actually quite surprised that RFC 5740 is not mentioned at all here. I =
would rather refer to RFC 5740 (rather than DVB-IPTV).

5) Do not use (upper-case) 2119 keywords in the introduction. IESG does =
not like it :)

6) Section 2: The definitions seem to evolve around SSM. Why would not =
they apply to ASM?

7) Section 3 and further on: There are several lower-case 2119 keywords =
in the text which you might wanna replace.

8) Section 6.1.1 says "If the loss reporter is part of group, the =
Distribution source MUST filter this packet out and not forward it back =
to this loss reporter." How does it filter that loss reporter out?

9) What about the cases where the receiver will actually not send a NACK =
despite the packet loss? Consider where there is also a FEC stream that =
can recover the lost packets. Should the loss reporter listen to the FEC =
channel as well and act accordingly?

10) You seem to assume that different loss reporters will always see the =
same losses. That is not necessarily true. What should be reported back =
to the receivers if the reports coming to the FT are conflicting? Do you =
report the union or intersection of the reports back to the receivers?

11) You also seem to be relating this NACK implosion to retransmission. =
I would actually prefer to keep the retransmission stuff out of the text =
here (just mention it in the introduction and then forget about it). =
Whether the distribution source actually retransmits the missing packets =
or not has nothing to do with the messages defined in this draft.

-acbegen=20



> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Qin Wu
> Sent: Saturday, October 09, 2010 12:49 PM
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-03
>=20
> Folks,
>=20
> We have updated draft-wu-avt-retransmission-suppression-rtp which =
should now reflect the discussion we had since IETF78.
>=20
> The feedback suppression solution has also been greatly generalized =
comparing to the previous version.
>=20
> Comments are solicited.
>=20
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
>=20
>=20
> >A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback
> > message concerning a specified set of RTP messages may be
> > unnecessary, or even harmful.  For example, in a source-specific
> > multicast session with large fan-out, packet loss close to the media
> > or distribution source of the session is detected as a loss by a
> > large number of receivers.  The RTCP NACK messages used to request
> > retransmission of the missing packets are all addressed to the media
> > sender, or a designated feedback target.  This may result in a
> > phenomenon known variously as a "NACK implosion" or "feedback =
storm".
> > The Feedback Suppression message defined herein is used to notify
> > receivers that packet loss was detected and that the sender of the
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by =
not
> > sending a feedback message (e.g. a NACK) that they otherwise would,
> > This in turn reduces load on both the feedback target and on the
> > network.  This document registers two new RTCP messages for Feedback
> > Suppression.
> >
> > A URL for this Internet-Draft is:
> > =
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supressio=
n-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >
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> =
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> > _______________________________________________
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> > I-D-Announce@ietf.org
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> > Internet-Draft directories: http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >

From Even.roni@huawei.com  Sun Oct 10 14:03:23 2010
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Date: Sun, 10 Oct 2010 23:01:28 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: [AVT] Requests for agenda time in Beijing - IETF 79
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Hi,

 

If you wish to have agenda time at the upcoming meeting in Beijing , please
send an agenda request to the chairs. We are trying to prepare the agenda
early and not wait for the last minute.

 

We will allocate time for port mapping to finalize the token/cookie issue.

 

We will use the rest of the time for presentations giving priority based on
the AVT milestones and open issues. 

 

 

We have two time slot in the preliminary agenda:

 

Monday, Afternoon Session I 13:00-15:00 

 

Wednesday , Afternoon Session I 13:00-15:00

 

 

Please note that this is not the final agenda yet.

 
Thanks
Roni Even and Keith Drage
AVT Chairs
 

 


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<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>If you wish to have agenda time at the
upcoming meeting in Beijing , please send an agenda request to the =
chairs. We
are trying to prepare the agenda early and not wait for the last =
minute.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>We will allocate time for port mapping =
to
finalize the token/cookie issue.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>We will use the rest of the time for =
presentations
giving priority based on the AVT milestones and open issues. =
<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>We have two time slot in the =
preliminary
agenda:<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Monday,
Afternoon Session I 13:00-15:00 <o:p></o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Wednesday
, Afternoon Session I 13:00-15:00<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><b><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif"'>Please note that this is not the final =
agenda
yet.<o:p></o:p></span></p>

<pre><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></pre>=
<pre><span
style=3D'font-family:"Arial","sans-serif"'>Thanks<o:p></o:p></span></pre>=
<pre><span
style=3D'font-family:"Arial","sans-serif"'>Roni Even and Keith =
Drage<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Arial","sans-serif"'>AVT =
Chairs<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></pre>=


<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></p>

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From sunseawq@huawei.com  Mon Oct 11 01:37:26 2010
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Date: Mon, 11 Oct 2010 16:38:22 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, avt@ietf.org
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Hi, Ali
Thank for your comments, please see my reply inline.
----- Original Message ----- 
From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Qin Wu" <sunseawq@huawei.com>; <avt@ietf.org>
Sent: Sunday, October 10, 2010 12:38 AM
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03


Comments:
1) Abstract is too long, could you shorten it?

[Qin] Okay.

Typos in abstract:
a) .. that *are* sending ...

[Qin]: You may misunderstand, the subject is actually "sending a feedback message"

b) RTP messages -> RTP packets

[Qin]: yes.

c) "the sender of the report will either proactively recover": Well, it does not have to be the sender of the report who will attempt the recovery. I would rephrase this.

[Qin]: Okay.

2) Throughout the document, avoiding "sender" might be a good idea. For the source, use media or distribution source. Sender may mean different things in different contexts.

[Qin]: Okay.

3) Intro 1st paragraph: The retransmission payload format is 4588 not 4585.

[Qin]: Good catch, thanks.

4) Most of the discussion in the introduction section can be shortened in a concrete way by referring to other existing RFC in this area. I am actually quite surprised that RFC 5740 is not mentioned at all here. I would rather refer to RFC 5740 (rather than DVB-IPTV).

[Qin]; Okay.

5) Do not use (upper-case) 2119 keywords in the introduction. IESG does not like it :)

[Qin]: Okay.

6) Section 2: The definitions seem to evolve around SSM. Why would not they apply to ASM?

[Qin]: Right, they should apply to ASM.

7) Section 3 and further on: There are several lower-case 2119 keywords in the text which you might wanna replace.

[Qin]: Okay, thanks.

8) Section 6.1.1 says "If the loss reporter is part of group, the Distribution source MUST filter this packet out and not forward it back to this loss reporter." How does it filter that loss reporter out?

[Qin]: It is one optimized behavior.
One possible way to do this is that the Distribution Source knows the address of loss reporter or know in which port it can receive the RTCP message from the loss reporter.
Is that reasonable?

9) What about the cases where the receiver will actually not send a NACK despite the packet loss? Consider where there is also a FEC stream that can recover the lost packets. Should the loss reporter listen to the FEC channel as well and act accordingly?

[Qin]: Good question. I think if the reciever know it will not send a NACK, then the distribution source should know as well. In such case, the distribution source may choose not to send feedback suppression message to the receiver.

10) You seem to assume that different loss reporters will always see the same losses. That is not necessarily true. What should be reported back to the receivers if the reports coming to the FT are conflicting? Do you report the union or intersection of the reports back to the receivers?

[Qin]: If there are duplication reports, the Distribution source should filter the duplicated report out and only send one copy of report to the reciever. What the distribution source does is to create one summary report which only append different loss in this summary report.

11) You also seem to be relating this NACK implosion to retransmission. I would actually prefer to keep the retransmission stuff out of the text here (just mention it in the introduction and then forget about it). Whether the distribution source actually retransmits the missing packets or not has nothing to do with the messages defined in this draft.

[Qin] Okay.

-acbegen 



> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Qin Wu
> Sent: Saturday, October 09, 2010 12:49 PM
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
> 
> Folks,
> 
> We have updated draft-wu-avt-retransmission-suppression-rtp which should now reflect the discussion we had since IETF78.
> 
> The feedback suppression solution has also been greatly generalized comparing to the previous version.
> 
> Comments are solicited.
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
> 
> 
> >A New Internet-Draft is available from the on-line Internet-Drafts directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback
> > message concerning a specified set of RTP messages may be
> > unnecessary, or even harmful.  For example, in a source-specific
> > multicast session with large fan-out, packet loss close to the media
> > or distribution source of the session is detected as a loss by a
> > large number of receivers.  The RTCP NACK messages used to request
> > retransmission of the missing packets are all addressed to the media
> > sender, or a designated feedback target.  This may result in a
> > phenomenon known variously as a "NACK implosion" or "feedback storm".
> > The Feedback Suppression message defined herein is used to notify
> > receivers that packet loss was detected and that the sender of the
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by not
> > sending a feedback message (e.g. a NACK) that they otherwise would,
> > This in turn reduces load on both the feedback target and on the
> > network.  This document registers two new RTCP messages for Feedback
> > Suppression.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >
> 
> 
> --------------------------------------------------------------------------------
> 
> 
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >

From csp@csperkins.org  Mon Oct 11 04:02:05 2010
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From: Colin Perkins <csp@csperkins.org>
To: Ali C. Begen (abegen) <abegen@cisco.com>
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This draft is generally along the right lines, and =96 as discussed in =20=

Maastricht =96 we should consider adopting it as a working group draft. =20=

That said, I have a number of comments, which generally echo Ali's =20
comments:

- The abstract needs to be rewritten as an abstract, rather than an =20
introduction.

- Should note, in the introduction, that the NACK storm respects the =20
RTCP bandwidth constraint on average, but that na=EFve implementations =20=

that don=92t choose to dither their response can cause short term =20
overload. There is no risk of congestion collapse, if the receivers =20
are correctly implemented. While there is a real problem to be solved, =20=

I believe the discussion in the introduction overplays the risks.

- The introduction includes RFC 2119 language specifying requirements =20=

=96 this should be moved into Section 3. Specifically, I suggest the =20
last six paragraphs of the Introduction be moved into Section 3.

- Section 3 contains a lot of discussion of the loss reporter and the =20=

SSM use case. Much of this seems unnecessary in this part of the =20
draft: discussion of SSM and the loss reporter concept should be in =20
section 6.1, leaving section 3 to specify the general protocol =20
behaviour, independent of particular use cases.

- Section 4 has unnecessary subheadings. Suggest removing the 4.1 =20
header then renaming section 4.1.1 to =934.1 Transport Layer Feedback: =20=

NACK Suppression Report=94, and also removing the 4.2 header then =20
renaming section 4.2.1 to =934.2 Payload Specific Feedback: FIR =20
suppression report=94.

- Section 7 should note that middleboxes that are not visible at the =20
RTP layer that wish to send NACK/FIR suppression reports on behalf of =20=

the sender can only do so if they spoof the SSRC of the sender. This =20
is difficult is SRTP is in use. If the middlebox is visible at the RTP =20=

layer, this is not an issue, provided the middlebox is part of the =20
security context for the session.

- Section 7 should note that endpoints that receive a NACK/FIR =20
suppression request would be well-advised to ignore it, unless it is =20
authenticated via SRTCP or similar. Accepting un-authenticated NACK/=20
FIR suppression requests can lead to a denial of service attack, where =20=

the endpoint accepts poor quality media that could be repaired.

None of these are fundamental, and I don't think there's anything that =20=

should delay adoption as a working group draft.

Colin



On 9 Oct 2010, at 17:38, Ali C. Begen (abegen) wrote:
> Comments:
> 1) Abstract is too long, could you shorten it?
>
> Typos in abstract:
> a) .. that *are* sending ...
> b) RTP messages -> RTP packets
> c) "the sender of the report will either proactively recover": Well, =20=

> it does not have to be the sender of the report who will attempt the =20=

> recovery. I would rephrase this.
>
> 2) Throughout the document, avoiding "sender" might be a good idea. =20=

> For the source, use media or distribution source. Sender may mean =20
> different things in different contexts.
>
> 3) Intro 1st paragraph: The retransmission payload format is 4588 =20
> not 4585.
>
> 4) Most of the discussion in the introduction section can be =20
> shortened in a concrete way by referring to other existing RFC in =20
> this area. I am actually quite surprised that RFC 5740 is not =20
> mentioned at all here. I would rather refer to RFC 5740 (rather than =20=

> DVB-IPTV).
>
> 5) Do not use (upper-case) 2119 keywords in the introduction. IESG =20
> does not like it :)
>
> 6) Section 2: The definitions seem to evolve around SSM. Why would =20
> not they apply to ASM?
>
> 7) Section 3 and further on: There are several lower-case 2119 =20
> keywords in the text which you might wanna replace.
>
> 8) Section 6.1.1 says "If the loss reporter is part of group, the =20
> Distribution source MUST filter this packet out and not forward it =20
> back to this loss reporter." How does it filter that loss reporter =20
> out?
>
> 9) What about the cases where the receiver will actually not send a =20=

> NACK despite the packet loss? Consider where there is also a FEC =20
> stream that can recover the lost packets. Should the loss reporter =20
> listen to the FEC channel as well and act accordingly?
>
> 10) You seem to assume that different loss reporters will always see =20=

> the same losses. That is not necessarily true. What should be =20
> reported back to the receivers if the reports coming to the FT are =20
> conflicting? Do you report the union or intersection of the reports =20=

> back to the receivers?
>
> 11) You also seem to be relating this NACK implosion to =20
> retransmission. I would actually prefer to keep the retransmission =20
> stuff out of the text here (just mention it in the introduction and =20=

> then forget about it). Whether the distribution source actually =20
> retransmits the missing packets or not has nothing to do with the =20
> messages defined in this draft.
>
> -acbegen
>
>
>
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =20=

>> Of Qin Wu
>> Sent: Saturday, October 09, 2010 12:49 PM
>> To: avt@ietf.org
>> Subject: [AVT] New Version Notification for draft-wu-avt-=20
>> retransmission-supression-rtp-03
>>
>> Folks,
>>
>> We have updated draft-wu-avt-retransmission-suppression-rtp which =20
>> should now reflect the discussion we had since IETF78.
>>
>> The feedback suppression solution has also been greatly generalized =20=

>> comparing to the previous version.
>>
>> Comments are solicited.
>>
>> Regards!
>> -Qin
>> ----- Original Message -----
>> From: <Internet-Drafts@ietf.org>
>> To: <i-d-announce@ietf.org>
>> Sent: Saturday, October 09, 2010 12:45 PM
>> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
>>
>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts =20=

>>> directories.
>>>
>>> Title           : RTCP Report Extension for Feedback Suppression
>>> Author(s)       : W. Wu, et al.
>>> Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
>>> Pages           : 24
>>> Date            : 2010-10-08
>>>
>>> This document specifies an extension to the RTCP feedback messages
>>> defined in the Audio-Visual Profile with Feedback (AVPF).  This
>>> extension allows an intermediate node in the network (such as an RTP
>>> translator) inform downstream receivers that sending a feedback
>>> message concerning a specified set of RTP messages may be
>>> unnecessary, or even harmful.  For example, in a source-specific
>>> multicast session with large fan-out, packet loss close to the media
>>> or distribution source of the session is detected as a loss by a
>>> large number of receivers.  The RTCP NACK messages used to request
>>> retransmission of the missing packets are all addressed to the media
>>> sender, or a designated feedback target.  This may result in a
>>> phenomenon known variously as a "NACK implosion" or "feedback =20
>>> storm".
>>> The Feedback Suppression message defined herein is used to notify
>>> receivers that packet loss was detected and that the sender of the
>>> report will either proactively recover, or no recovery is possible.
>>> Receivers respond to receipt of a feedback suppression message by =20=

>>> not
>>> sending a feedback message (e.g. a NACK) that they otherwise would,
>>> This in turn reduces load on both the feedback target and on the
>>> network.  This document registers two new RTCP messages for Feedback
>>> Suppression.
>>>
>>> A URL for this Internet-Draft is:
>>> =
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression=
-rtp-03.txt
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> Below is the data which will enable a MIME compliant mail reader
>>> implementation to automatically retrieve the ASCII version of the
>>> Internet-Draft.
>>>
>>
>>
>> =
--------------------------------------------------------------------------=
------
>>
>>
>>> _______________________________________________
>>> I-D-Announce mailing list
>>> I-D-Announce@ietf.org
>>> https://www.ietf.org/mailman/listinfo/i-d-announce
>>> Internet-Draft directories: http://www.ietf.org/shadow.html
>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From csp@csperkins.org  Mon Oct 11 04:27:28 2010
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From: Colin Perkins <csp@csperkins.org>
To: Qin Wu <sunseawq@huawei.com>
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Subject: Re: [AVT] Fw: I-D Action:draft-wu-avt-rtcp-xr-quality-monitoring-02.txt
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On 20 Sep 2010, at 08:37, Qin Wu wrote:
> Hi, Colin:
> Thank for your valuable comments. Please see my reply inline.
>
> Regards!
> -Qin
> ----- Original Message -----
> From: "Colin Perkins" <csp@csperkins.org>
> To: "Qin Wu" <sunseawq@huawei.com>
> Cc: <avt@ietf.org>; <Roland.Schott@telekom.de>
> Sent: Friday, September 17, 2010 5:50 PM
> Subject: Re: [AVT] Fw: I-D Action:draft-wu-avt-rtcp-xr-quality-=20
> monitoring-02.txt
>
>
> I've read this draft, and have some comments. In general, I think it's
> on the right lines, but there are a number of details to be resolved.
>
> [Qin]: Yes, you are right. The main changes we made for this draft =20
> is try to align with Monitoring Architecture draft,
> which more focus on how to extend metric report using RTCP XR for =20
> monitoring purpose in the latest version.
>
> Initial synchronisation delay: okay
>
> Audio-video playout offset: makes sense in principle, but is defined
> in a way that=92s specific to a simple scenario with one audio and one
> video stream. This may be better defined as a general synchronisation
> offset metric, which reports the offset of this stream relative to
> other streams sent with the same RTCP CNAME (where the reference
> stream reports offset of zero, and the others are signed offsets
> relative to it). That would allow it to be used to report on more
> complex synchronisation scenarios, rather than define other similar
> metrics for those scenarios later
>
> [Qin]: Good suggestion. One generic metric will benefit for a large =20=

> broad appliations. it is reasonable to replace the audio-video =20
> playout offset report with one more genric reusable metric. I will =20
> do this  in the next version.
>
> In the layered stream statistics report block, the layer type flag is
> phrased in terms of H.264 SVC video, but there is nothing particularly
> SVC-specific here. Can it be written in a more general way, just
> specifying base and enhancement layers, so the same metric can be used
> for other layered codecs?
>
> [Qin]: I am not sure whether we need more finely granularity of =20
> enchancement layers to distinguish one enhancment layer from in =20
> other layered codecs Is it reasonable to only specify base and =20
> enhancment layers?

Probably - depends on your use case.

> It=92s not clear that the loss and duplicate reports need to be =
optional
> in the layered streams statistics metric block. They are straight
> forward to calculate, and making them optional seems to only introduce
> unnecessary complexity into the protocol. Similar for the video
> statistics summary report block.
>
> [Qin] Good point. It seems these two indication fields for loss and =20=

> duplication are
> over-designed, I will remove them in the new version.
>
> The definition of the lost_layered component packets and dup_layered
> component packets is under-specified.
>
> [Qin]: Okay, thank for catching this. we will specify this in the =20
> new version.
>
> Is the frame type indicator field used in the video statistics report
> block and the video stream loss and discard metrics block well
> defined? The characterisation of video frames into I, P, and B frames
> makes sense for MPEG-2, but is it appropriate for other codecs?
>
> [Qin]: I think classifying video media into  I, P and frames also =20
> works for MPEG-4, but they may take the different name like
> I-VOP, B-VOP and P-VOP, for the other video Codec like V6, it also =20
> characterize the video into different picture types like I and P.
>
> Therefore may I propose to use  key frame and reference frame to =20
> replace I,P, and B frames, which is more generic and can be used to =20=

> distinguish different picture types.

I like the idea, but I'm not sure that terminology is clearer.

> Unlike the other metrics, the TR 101 290 decodability metrics report
> block is entirely MPEG-2 specific. I suggest splitting this into a
> separate draft, since it doesn=92t really relate to the other metrics
> defined in this draft.
>
> [Qin]: I am not sure whether we should leave TR101 290 decodability =20=

> metrics out from this draft. But I think this metrics could be taken =20=

> as one of input to calculate the MOS-V which reflects the MPEG-2 =20
> media quality in some way.

Yes, but that doesn't mean they need to be specified in this draft. As =20=

I said before, this is entirely MPEG-2 specific, and so would be =20
better if submitted as a separate draft, unrelated to this RTP-level =20
metrics draft.

> The RTP Terminal Configuration Metrics Block seems ill-thought out.
> The PLC type field doesn=92t seem appropriate to video - the =
definition
> looks to be copied from an audio metrics draft. The JBA field is ill-
> defined, and doesn=92t give meaningful information. The FEC field is
> very limited, and too tied to particular FEC schemes in a way that is
> not desirable: the media types namespace should be used here, rather
> than defining a new, very limited, two-bit field, if you need to do
> this at all. It's not clear that this information should be reported
> in RTCP, since the same information is available in the SDP signalling
> channel.
>
> [Qin]: I agree the parameters defined in this metrics can be =20
> obtained in the SDP signaling. It seems this Terminal related =20
> metrics looks like a evil. :-) But what I like to argue about is =20
> different type of terminal with differnt capability may demonstrate =20=

> different reception quality to the users. Are we aware of that? =20
> Isn't valuable to take this terminal associated metrics into  accout =20=

> when we evaluate the media quality in the comprehensive way?


Sure, but the information is already available, and doesn't need to be =20=

duplicated here.

--=20
Colin Perkins
http://csperkins.org/




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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Hi All,

We have recently updated our internet-draft for inter-destination-media
synchronization using RTCP XR blocks. The updated version includes more
references and a shortened abstract.

We are looking forward to your comments and are especially interested in
how you think our ID relates to recent work within this WG to specify
guidelines for new XR blocks.=20

Regards,

Ray van Brandenburg



-----Original Message-----
From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]=20
Sent: maandag 11 oktober 2010 14:44
To: Brandenburg, R. (Ray) van
Subject: New Version Notification for
draft-brandenburg-avt-rtcp-for-idms-02=20


A new version of I-D, draft-brandenburg-avt-rtcp-for-idms-02.txt has
been successfully submitted by Ray Brandenburg and posted to the IETF
repository.

Filename:	 draft-brandenburg-avt-rtcp-for-idms
Revision:	 02
Title:		 RTCP XR Block Type for inter-destination media
synchronization
Creation_date:	 2010-10-11
WG ID:		 Independent Submission
Number_of_pages: 11

Abstract:
This document specifies an RTCP XR Block Type and associated SDP
parameter for inter-destination media synchronization (IDMS). The RTCP
Block Type is used to collect media play-out information from
participants in a group watching a specific RTP media stream and to
distribute a summary of the collected information so that the
participants can synchronize play-out.

Typical applications for inter-destination media synchronization are
social TV, watching apart together and shared service control (i.e.
applications where two or more geographically separated users are
watching a media stream together).
=20



The IETF Secretariat.


This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


From peter.musgrave@magorcorp.com  Mon Oct 11 07:15:33 2010
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HI Ray,

I think this is a useful XR.

I lack background in the IDMS application.  Is the Boronat2009
reference available online anywhere? (I am interested in whether there
are separate XRs for audio and video and how the lipsync alignment
issues are handled within the overall synchronization strategy).

Should the relationship of this doc to ETSI TISPAN be mentioned in the Abst=
ract?


Nits:
3. First use of MSAS is not expanded.
3.3 Typo s/SSCR/SSRC/

Regards,

Peter Musgrave

On Mon, Oct 11, 2010 at 9:14 AM, Brandenburg, R. (Ray) van
<ray.vanbrandenburg@tno.nl> wrote:
> Hi All,
>
> We have recently updated our internet-draft for inter-destination-media
> synchronization using RTCP XR blocks. The updated version includes more
> references and a shortened abstract.
>
> We are looking forward to your comments and are especially interested in
> how you think our ID relates to recent work within this WG to specify
> guidelines for new XR blocks.
>
> Regards,
>
> Ray van Brandenburg
>
>
>
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> Sent: maandag 11 oktober 2010 14:44
> To: Brandenburg, R. (Ray) van
> Subject: New Version Notification for
> draft-brandenburg-avt-rtcp-for-idms-02
>
>
> A new version of I-D, draft-brandenburg-avt-rtcp-for-idms-02.txt has
> been successfully submitted by Ray Brandenburg and posted to the IETF
> repository.
>
> Filename: =A0 =A0 =A0 =A0draft-brandenburg-avt-rtcp-for-idms
> Revision: =A0 =A0 =A0 =A002
> Title: =A0 =A0 =A0 =A0 =A0 RTCP XR Block Type for inter-destination media
> synchronization
> Creation_date: =A0 2010-10-11
> WG ID: =A0 =A0 =A0 =A0 =A0 Independent Submission
> Number_of_pages: 11
>
> Abstract:
> This document specifies an RTCP XR Block Type and associated SDP
> parameter for inter-destination media synchronization (IDMS). The RTCP
> Block Type is used to collect media play-out information from
> participants in a group watching a specific RTP media stream and to
> distribute a summary of the collected information so that the
> participants can synchronize play-out.
>
> Typical applications for inter-destination media synchronization are
> social TV, watching apart together and shared service control (i.e.
> applications where two or more geographically separated users are
> watching a media stream together).
>
>
>
>
> The IETF Secretariat.
>
>
> This e-mail and its contents are subject to the DISCLAIMER at http://www.=
tno.nl/disclaimer/email.html
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

From ishan.vaishnavi@gmail.com  Mon Oct 11 08:39:15 2010
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--0016e686cd3aa2e2990492592d8c
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Hi Peter,

Thanks for taking time to review this document. Find my comments inline.

Cheers,
Ishan




> I lack background in the IDMS application.  Is the Boronat2009
> reference available online anywhere?


You need either an Elsevier account or a Science Direct one.
Here<http://www.sciencedirect.com/science?_ob=3DArticleURL&_udi=3DB6V0G-4SK=
B3J2-1&_user=3D10&_coverDate=3D03%2F31%2F2009&_rdoc=3D1&_fmt=3Dhigh&_orig=
=3Dsearch&_origin=3Dsearch&_sort=3Dd&_docanchor=3D&view=3Dc&_rerunOrigin=3D=
scholar.google&_acct=3DC000050221&_version=3D1&_urlVersion=3D0&_userid=3D10=
&md5=3Ded63673a77c38dc50c5fbc2199129d08&searchtype=3Da>is
the link. Hope that helps.



> (I am interested in whether there
> are separate XRs for audio and video and how the lipsync alignment
> issues are handled within the overall synchronization strategy).
>
>
In my opinion lip synchronization is somewhat out of the scope of this
document. The idea is that each distributed client reports its play-out
position of its "primary" stream. This primary stream can be a virtual
stream or a real audio, video, subtitles or images for that matter.  All
other streams locally on each client synchronize to that stream, this must
be done by the respective application. The current XR specification relates
to how to get these primary streams are synchronized across different
geographically separated clients. Maybe it can be extended to include this
local synchronization as well, but that's another discussion.



> Should the relationship of this doc to ETSI TISPAN be mentioned in the
> Abstract?
>

I will leave this one for Ray.

--0016e686cd3aa2e2990492592d8c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Peter,=A0<div><br></div><div>Thanks for taking time to review this docum=
ent. Find my comments inline.</div><div><br></div><div>Cheers,</div><div>Is=
han<br><br><div class=3D"gmail_quote"><br><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">

I lack background in the IDMS application. =A0Is the Boronat2009<br>
reference available online anywhere?</blockquote><div><br></div><div>You ne=
ed either an Elsevier account or a Science Direct one. <a href=3D"http://ww=
w.sciencedirect.com/science?_ob=3DArticleURL&amp;_udi=3DB6V0G-4SKB3J2-1&amp=
;_user=3D10&amp;_coverDate=3D03%2F31%2F2009&amp;_rdoc=3D1&amp;_fmt=3Dhigh&a=
mp;_orig=3Dsearch&amp;_origin=3Dsearch&amp;_sort=3Dd&amp;_docanchor=3D&amp;=
view=3Dc&amp;_rerunOrigin=3Dscholar.google&amp;_acct=3DC000050221&amp;_vers=
ion=3D1&amp;_urlVersion=3D0&amp;_userid=3D10&amp;md5=3Ded63673a77c38dc50c5f=
bc2199129d08&amp;searchtype=3Da">Here</a> is the link. Hope that helps.</di=
v>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"> (I am interes=
ted in whether there<br>
are separate XRs for audio and video and how the lipsync alignment<br>
issues are handled within the overall synchronization strategy).<br>
<br></blockquote><div><br></div><div>In my opinion lip synchronization is s=
omewhat out of the scope of this document. The idea is that each distribute=
d client reports its play-out position of its &quot;primary&quot; stream. T=
his primary stream can be a virtual stream or a real audio, video, subtitle=
s or images for that matter. =A0All other streams locally on each client sy=
nchronize to that stream, this must be done by the respective application. =
The current XR specification relates to how to get these primary streams ar=
e synchronized across different geographically separated clients. Maybe it =
can be extended to include this local synchronization as well, but that&#39=
;s another discussion.=A0</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Should the relationship of this doc to ETSI TISPAN be mentioned in the Abst=
ract?<br></blockquote><div><br></div><div>I will leave this one for Ray.=A0=
</div><div><br></div><div><br></div><div><br></div></div>
</div>

--0016e686cd3aa2e2990492592d8c--

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...
> Typos in abstract:
> a) .. that *are* sending ...
>=20
> [Qin]: You may misunderstand, the subject is actually "sending a =
feedback message"

Right. Shortening the sentence would definitely help here.
=20
...
> 8) Section 6.1.1 says "If the loss reporter is part of group, the =
Distribution source MUST filter this packet out and not forward
> it back to this loss reporter." How does it filter that loss reporter =
out?
>=20
> [Qin]: It is one optimized behavior.
> One possible way to do this is that the Distribution Source knows the =
address of loss reporter or know in which port it can
> receive the RTCP message from the loss reporter.
> Is that reasonable?

Are your reports unicast or multicast? In the latter, how do you deal =
with it?
=20
> 9) What about the cases where the receiver will actually not send a =
NACK despite the packet loss? Consider where there is
> also a FEC stream that can recover the lost packets. Should the loss =
reporter listen to the FEC channel as well and act
> accordingly?
>=20
> [Qin]: Good question. I think if the reciever know it will not send a =
NACK, then the distribution source should know as well. In

Not necessarily. If the FEC is on another distribution channel (which is =
usually the case), the DS does not know what the receivers are getting =
in terms of FEC.

> such case, the distribution source may choose not to send feedback =
suppression message to the receiver.

I think we need to think about this a bit further. If FEC recovers most =
of the errors for majority of the receivers, maybe doing this (feedback =
suppression) will not be needed as much.
=20
> 10) You seem to assume that different loss reporters will always see =
the same losses. That is not necessarily true. What
> should be reported back to the receivers if the reports coming to the =
FT are conflicting? Do you report the union or
> intersection of the reports back to the receivers?
>=20
> [Qin]: If there are duplication reports, the Distribution source =
should filter the duplicated report out and only send one copy
> of report to the reciever. What the distribution source does is to =
create one summary report which only append different loss
> in this summary report.

This does not answer my question. If the reports are identical, yes you =
can remove the duplicate. But if the reports are not identical. Now you =
have a decision to make? And whatever you do may actually change the =
performance.
=20
-acbegen

From csp@csperkins.org  Mon Oct 11 18:14:30 2010
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From: Colin Perkins <csp@csperkins.org>
To: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] Revised Consensus call - On Token approach for draft-ietf-avt-ports-for-ucast-mcast-rtp
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On 8 Oct 2010, at 09:47, Roni Even wrote:
> There were some concerns about how we drafted the consensus call so  
> I would like to try and give it another try.  Sorry for the long  
> message which I hope will clarify what are the proposed options. We  
> would like to get feedback by October 18th.
>
>
> The Port Mapping Between Unicast and Multicast RTP Sessions  
> described in http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast-rtp-02 
>  was discussed for a couple of IETF meeting and on the list. The  
> initial approach that was agreed after we had a breakout session in  
> previous IETF meetings was to use a cookie based solution even  
> though it required more messages and you needed a cookie for every  
> RTP session.
>
> Before the last IETF meeting there were two proposals to update the  
> solution to a token based approach.
>
> In the token approach:
> - Tokens are port-independent (cookies are port-dependent) and only  
> depend on the client's IP. That is, once a client gets its token, it  
> can use it with any of its ports and with any Feedback Target.
> - Tokens address the concern raised by Magnus (http://www.ietf.org/mail-archive/web/avt/current/msg12617.html 
> )
> - No need for keep-alives
>
> We had a discussion in Maastricht and the outcome is in the meeting  
> minutes http://www.ietf.org/proceedings/78/minutes/avt.txt (see the  
> section on draft-begen-avt-token-for-portmapping). The token draft  
> has Dan Wing (who is also BEHAVE co-chair) as a co-author and he  
> looked at the solution verifying that it will work well with NATs
>
>
> At the meeting the preference was for a token based solution but  
> there are were two approaches discussed.
>
> 1. Use the token only solution
>
> 2. Use a combination when if the request has only an IP address the  
> response will be a token and if it will have both IP address and  
> port the response will be a  cookie.
>
> According to the minutes the support was for the second approach.  
> The major reason was a concern that carrier grade NATs can allocate  
> more than one IP address as the public address.
>
> The authors of
> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01  
> suggest that they addressed the concerns about the token only  
> solution (see section 9 of the draft) and prefer to have a token  
> only solution. This is a change from the meeting minutes and this is  
> the reason for the second question in this consensus call.
>
> So here are two questions for the group !!!
>
> 1. Do the people in the mailing list support the proposal to use a  
> token based solution instead on the cookie one. (this was the  
> preference in the consensus call in Maastricht) and we would like to  
> confirm it on the list.
>
> 2. We have two options for the token approach, please give your  
> preferences.
>
> 2.1 Token only solution as proposed in
> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
> 2.2 Token + cookie, in this case if the request will include only IP  
> address the response is a token, if it also includes a port it will  
> be a cookie. This approach is for cases were the NAT may use more  
> than one IP address for mapping a local address.
>
>
> Please respond even if you did it on the previous consensus call.


The approach in draft-begen-avt-token-for-portmapping-01 looks to be  
sufficient to allow a server to determine that multiple requests come  
from the same client, provided the client isn't multi-homed and isn't  
sharing an IP address with other clients. The problem statement is not  
clear enough for me to be sure if that's sufficient.

If it were important to do so, the server could distinguish multiple  
clients sharing the same IP address if each client supplied a random  
nonce that was taken into account when generating the token in  
addition to the client's IP address. This may be an issue with use of  
fractional IP addresses in future, and so might be something to  
consider adding.

Distinguishing multi-homed clients, or clients where requests come  
from several different IP addresses, is more complex. Is this an  
issue, or is it sufficient to assume that a client requests a new  
token when changing the address it uses?

I'd also like to note that the port mapping operation is independent  
of the RAMS drafts. It would be good if the draft the group ends up  
with is written as a standalone mechanism, with RAMS described as one  
example use case, rather than being written in a RAMS-specific manner  
as are the current drafts.

Cheers,
Colin


-- 
Colin Perkins
http://csperkins.org/




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Date: Tue, 12 Oct 2010 15:05:13 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, avt@ietf.org
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Hi, 
----- Original Message ----- 
From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Qin Wu" <sunseawq@huawei.com>; <avt@ietf.org>
Sent: Tuesday, October 12, 2010 7:38 AM
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03


...
> Typos in abstract:
> a) .. that *are* sending ...
> 
> [Qin]: You may misunderstand, the subject is actually "sending a feedback message"

Right. Shortening the sentence would definitely help here.

[Qin]: Okay.
...
> 8) Section 6.1.1 says "If the loss reporter is part of group, the Distribution source MUST filter this packet out and not forward
> it back to this loss reporter." How does it filter that loss reporter out?
> 
> [Qin]: It is one optimized behavior.
> One possible way to do this is that the Distribution Source knows the address of loss reporter or know in which port it can
> receive the RTCP message from the loss reporter.
> Is that reasonable?

Are your reports unicast or multicast? In the latter, how do you deal with it?

[Qin]: In the current draft, we assume Report sent from loss reporter to the distribution source is unicast, the report sent from or created  at distribution source is multicast.
Suppose the loss report is separated from the distribution source and send multicast report to the distribution source, the distribution souce can identify this multicast report
based on the dedicated RTCP port used for loss report. Is it acceptable?
 
> 9) What about the cases where the receiver will actually not send a NACK despite the packet loss? Consider where there is
> also a FEC stream that can recover the lost packets. Should the loss reporter listen to the FEC channel as well and act
> accordingly?
> 
> [Qin]: Good question. I think if the reciever know it will not send a NACK, then the distribution source should know as well. In

Not necessarily. If the FEC is on another distribution channel (which is usually the case), the DS does not know what the receivers are getting in terms of FEC.

[Qin]: Right.

> such case, the distribution source may choose not to send feedback suppression message to the receiver.

I think we need to think about this a bit further. If FEC recovers most of the errors for majority of the receivers, maybe doing this (feedback suppression) will not be needed as much.
 
[Qin]: It seems we need media source or receiver to tell distribution source not suppress the specified stream sent from the media source. Right?
Or we assume FEC schemes will not used with suppression mechanism at the same time? Do you have any proposal?

> 10) You seem to assume that different loss reporters will always see the same losses. That is not necessarily true. What
> should be reported back to the receivers if the reports coming to the FT are conflicting? Do you report the union or
> intersection of the reports back to the receivers?
> 
> [Qin]: If there are duplication reports, the Distribution source should filter the duplicated report out and only send one copy
> of report to the reciever. What the distribution source does is to create one summary report which only append different loss
> in this summary report.

This does not answer my question. If the reports are identical, yes you can remove the duplicate. But if the reports are not identical. Now you have a decision to make? 

[Qin]: Append the different report to the summary report.

And whatever you do may actually change the performance.
 
[Qin] In my opinion,  If the report is created at the distribution source and sent from the distribution source to all the reciever, the performance will not be influenced. If the loss report is not part of distribution source, i.e., separated from distribution source, we may ask the loss reporter to send unicast report to the distribution source and then the distribution source create the summary report  in multicast and send it to the recievers. that is to say, the unicast report will be terminated at the distribution source, so  the performance will be affected very little. If you really concern about this, we could only consider the case that the loss report is part of distribution source and distribution source create the multicast report.


-acbegen

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Date: Tue, 12 Oct 2010 15:19:18 +0800
From: Qin Wu <sunseawq@huawei.com>
To: Colin Perkins <csp@csperkins.org>, "Ali C. Begen (abegen)" <abegen@cisco.com>
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Date: Tue, 12 Oct 2010 09:33:40 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <04CAD96D4C5A3D48B1919248A8FE0D540D5BF6E6@xmb-sjc-215.amer.cisco.com>
To: "'Ali C. Begen (abegen)'" <abegen@cisco.com>, 'Qin Wu' <sunseawq@huawei.com>, avt@ietf.org
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Subject: Re: [AVT] New Version Notification for	draft-wu-avt-retransmission-supression-rtp-03
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Hi,
> 
> ...
> > 8) Section 6.1.1 says "If the loss reporter is part of group, the
> Distribution source MUST filter this packet out and not forward
> > it back to this loss reporter." How does it filter that loss reporter
> out?
> >
> > [Qin]: It is one optimized behavior.
> > One possible way to do this is that the Distribution Source knows the
> address of loss reporter or know in which port it can
> > receive the RTCP message from the loss reporter.
> > Is that reasonable?
> 
> Are your reports unicast or multicast? In the latter, how do you deal
> with it?

I think this has to do with your question about merging reports.

> 
> > 9) What about the cases where the receiver will actually not send a
> NACK despite the packet loss? Consider where there is
> > also a FEC stream that can recover the lost packets. Should the loss
> reporter listen to the FEC channel as well and act
> > accordingly?
> >
> > [Qin]: Good question. I think if the reciever know it will not send a
> NACK, then the distribution source should know as well. In
> 
> Not necessarily. If the FEC is on another distribution channel (which
> is usually the case), the DS does not know what the receivers are
> getting in terms of FEC.

>From the DS perspective there are packet losses and letting the receiver
know even if it uses FEC may help the receiver giving him an earlier warning
allowing him to use FEC without waiting. 

> 
> > such case, the distribution source may choose not to send feedback
> suppression message to the receiver.
> 
> I think we need to think about this a bit further. If FEC recovers most
> of the errors for majority of the receivers, maybe doing this (feedback
> suppression) will not be needed as much.
> 
> > 10) You seem to assume that different loss reporters will always see
> the same losses. That is not necessarily true. What
> > should be reported back to the receivers if the reports coming to the
> FT are conflicting? Do you report the union or
> > intersection of the reports back to the receivers?
> >
> > [Qin]: If there are duplication reports, the Distribution source
> should filter the duplicated report out and only send one copy
> > of report to the reciever. What the distribution source does is to
> create one summary report which only append different loss
> > in this summary report.
> 
> This does not answer my question. If the reports are identical, yes you
> can remove the duplicate. But if the reports are not identical. Now you
> have a decision to make? And whatever you do may actually change the
> performance.

This is a good feedback. I think that this is application dependent, the
typical behavior can be union but if the DS knows the network topology it
may take another policy. Anyhow even if the receiver gets the union and on
his path the packet were not lost, it does not break since it will receive
the packets. The suppression just tell the receiver to not sent NACKs. The
draft does not address the behavior of the DS in the upstream direction and
leave it open to the implementation.

Roni Even

> 
> -acbegen
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From ray.vanbrandenburg@tno.nl  Tue Oct 12 00:46:51 2010
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "Peter Musgrave" <peter.musgrave@magorcorp.com>
Cc: avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-brandenburg-avt-rtcp-for-idms-02
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Hi Peter,
=20
Thanks for your comments.=20
=20

	Should the relationship of this doc to ETSI TISPAN be mentioned
in the Abstract?

=20
I agree and will add this in the next version. While we're on the
subject: the main reason we have brought this work to the IETF is
because we feel the AVT WG should be the place where XR blocks are
specified. If there is enough interest around here, one option could be
to have the AVT WG take the lead on the standardization of this XR
block, either as an informational or standards track document. The
relevant sections of the ETSI TISPAN spec could then reference the
resulting RFC. How do you feel about this?
=20
Regards,
=20
Ray van Brandenburg
=20
=20
=20

________________________________

From: Ishan Vaishnavi [mailto:ishan.vaishnavi@gmail.com]=20
Sent: maandag 11 oktober 2010 17:40
To: Peter Musgrave
Cc: Brandenburg, R. (Ray) van; avt@ietf.org
Subject: Re: [AVT] New Version Notification for
draft-brandenburg-avt-rtcp-for-idms-02


Hi Peter, =20

Thanks for taking time to review this document. Find my comments inline.

Cheers,
Ishan



=20

	I lack background in the IDMS application.  Is the Boronat2009
	reference available online anywhere?


You need either an Elsevier account or a Science Direct one. Here
<http://www.sciencedirect.com/science?_ob=3DArticleURL&_udi=3DB6V0G-4SKB3J2-
1&_user=3D10&_coverDate=3D03%2F31%2F2009&_rdoc=3D1&_fmt=3Dhigh&_orig=3Dsear=
ch&_ori
gin=3Dsearch&_sort=3Dd&_docanchor=3D&view=3Dc&_rerunOrigin=3Dscholar.google=
&_acct=3D
C000050221&_version=3D1&_urlVersion=3D0&_userid=3D10&md5=3Ded63673a77c38dc5=
0c5fb
c2199129d08&searchtype=3Da>  is the link. Hope that helps.

=20

	(I am interested in whether there
	are separate XRs for audio and video and how the lipsync
alignment
	issues are handled within the overall synchronization strategy).
=09
=09


In my opinion lip synchronization is somewhat out of the scope of this
document. The idea is that each distributed client reports its play-out
position of its "primary" stream. This primary stream can be a virtual
stream or a real audio, video, subtitles or images for that matter.  All
other streams locally on each client synchronize to that stream, this
must be done by the respective application. The current XR specification
relates to how to get these primary streams are synchronized across
different geographically separated clients. Maybe it can be extended to
include this local synchronization as well, but that's another
discussion.=20

=20

	Should the relationship of this doc to ETSI TISPAN be mentioned
in the Abstract?
=09


I will leave this one for Ray.=20



This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010>Hi Peter,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010></SPAN></FONT><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010>Thanks for your comments. </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1p=
x solid">Should=20
  the relationship of this doc to ETSI TISPAN be mentioned in the=20
Abstract?</BLOCKQUOTE>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010>I agree and will add this in the next version. W=
hile=20
we're on the subject: the main reason we have brought this work to the IETF=
 is=20
because we feel the AVT WG should be the place where XR blocks are specifie=
d. If=20
there is enough interest around here, one option could be to have the AVT W=
G=20
take the lead on the standardization of this XR block, either as an=20
informational or standards track document. The relevant sections of the ETS=
I=20
TISPAN&nbsp;spec could then reference the resulting RFC.&nbsp;How do you fe=
el=20
about this?</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010></SPAN></FONT><FONT face=3DArial color=3D#0000ff=20
size=3D2><SPAN class=3D339392807-12102010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D339392807-12102010>Ray van Brandenburg</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Ishan Vaishnavi=20
[mailto:ishan.vaishnavi@gmail.com] <BR><B>Sent:</B> maandag 11 oktober 2010=20
17:40<BR><B>To:</B> Peter Musgrave<BR><B>Cc:</B> Brandenburg, R. (Ray) van;=20
avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New Version Notification for=20
draft-brandenburg-avt-rtcp-for-idms-02<BR></FONT><BR></DIV>
<DIV></DIV>Hi Peter,&nbsp;
<DIV><BR></DIV>
<DIV>Thanks for taking time to review this document. Find my comments=20
inline.</DIV>
<DIV><BR></DIV>
<DIV>Cheers,</DIV>
<DIV>Ishan<BR><BR>
<DIV class=3Dgmail_quote><BR>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1p=
x solid">I=20
  lack background in the IDMS application. &nbsp;Is the Boronat2009<BR>refe=
rence=20
  available online anywhere?</BLOCKQUOTE>
<DIV><BR></DIV>
<DIV>You need either an Elsevier account or a Science Direct one. <A=20
href=3D"http://www.sciencedirect.com/science?_ob=3DArticleURL&amp;_udi=3DB6=
V0G-4SKB3J2-1&amp;_user=3D10&amp;_coverDate=3D03%2F31%2F2009&amp;_rdoc=3D1&=
amp;_fmt=3Dhigh&amp;_orig=3Dsearch&amp;_origin=3Dsearch&amp;_sort=3Dd&amp;_=
docanchor=3D&amp;view=3Dc&amp;_rerunOrigin=3Dscholar.google&amp;_acct=3DC00=
0050221&amp;_version=3D1&amp;_urlVersion=3D0&amp;_userid=3D10&amp;md5=3Ded6=
3673a77c38dc50c5fbc2199129d08&amp;searchtype=3Da">Here</A>=20
is the link. Hope that helps.</DIV>
<DIV><BR></DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1p=
x solid">(I=20
  am interested in whether there<BR>are separate XRs for audio and video an=
d how=20
  the lipsync alignment<BR>issues are handled within the overall synchroniz=
ation=20
  strategy).<BR><BR></BLOCKQUOTE>
<DIV><BR></DIV>
<DIV>In my opinion lip synchronization is somewhat out of the scope of this=20
document. The idea is that each distributed client reports its play-out pos=
ition=20
of its "primary" stream. This primary stream can be a virtual stream or a r=
eal=20
audio, video, subtitles or images for that matter. &nbsp;All other streams=20
locally on each client synchronize to that stream, this must be done by the=20
respective application. The current XR specification relates to how to get =
these=20
primary streams are synchronized across different geographically separated=20
clients. Maybe it can be extended to include this local synchronization as =
well,=20
but that's another discussion.&nbsp;</DIV>
<DIV><BR></DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1p=
x solid">Should=20
  the relationship of this doc to ETSI TISPAN be mentioned in the=20
Abstract?<BR></BLOCKQUOTE>
<DIV><BR></DIV>
<DIV>I will leave this one for Ray.&nbsp;</DIV>
<DIV><BR></DIV>
<DIV><BR></DIV>
<DIV><BR></DIV></DIV></DIV><pre>This e-mail and its contents are subject to=
 the DISCLAIMER at http://www.tno.nl/disclaimer/email.html
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From tom.van_caenegem@alcatel-lucent.com  Tue Oct 12 03:22:10 2010
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Colin Perkins <csp@csperkins.org>, Roni Even <Even.roni@huawei.com>
Date: Tue, 12 Oct 2010 12:23:05 +0200
Thread-Topic: [AVT] Revised Consensus call - On Token approach for draft-ietf-avt-ports-for-ucast-mcast-rtp
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Subject: Re: [AVT] Revised Consensus call - On Token approach for	draft-ietf-avt-ports-for-ucast-mcast-rtp
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Hi Colin,

See below,
Tom=20

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Colin=
 Perkins
Sent: maandag 11 oktober 2010 18:51
To: Roni Even
Cc: DRAGE, Keith (Keith); 'IETF AVT WG'; 'Dan Wing'
Subject: Re: [AVT] Revised Consensus call - On Token approach for draft-iet=
f-avt-ports-for-ucast-mcast-rtp

On 8 Oct 2010, at 09:47, Roni Even wrote:
> There were some concerns about how we drafted the consensus call so I=20
> would like to try and give it another try.  Sorry for the long message=20
> which I hope will clarify what are the proposed options. We would like=20
> to get feedback by October 18th.
>
>
> The Port Mapping Between Unicast and Multicast RTP Sessions described=20
> in=20
> http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast-rtp-02
>  was discussed for a couple of IETF meeting and on the list. The=20
> initial approach that was agreed after we had a breakout session in=20
> previous IETF meetings was to use a cookie based solution even though=20
> it required more messages and you needed a cookie for every RTP=20
> session.
>
> Before the last IETF meeting there were two proposals to update the=20
> solution to a token based approach.
>
> In the token approach:
> - Tokens are port-independent (cookies are port-dependent) and only=20
> depend on the client's IP. That is, once a client gets its token, it=20
> can use it with any of its ports and with any Feedback Target.
> - Tokens address the concern raised by Magnus=20
> (http://www.ietf.org/mail-archive/web/avt/current/msg12617.html
> )
> - No need for keep-alives
>
> We had a discussion in Maastricht and the outcome is in the meeting=20
> minutes http://www.ietf.org/proceedings/78/minutes/avt.txt (see the=20
> section on draft-begen-avt-token-for-portmapping). The token draft has=20
> Dan Wing (who is also BEHAVE co-chair) as a co-author and he looked at=20
> the solution verifying that it will work well with NATs
>
>
> At the meeting the preference was for a token based solution but there=20
> are were two approaches discussed.
>
> 1. Use the token only solution
>
> 2. Use a combination when if the request has only an IP address the=20
> response will be a token and if it will have both IP address and port=20
> the response will be a  cookie.
>
> According to the minutes the support was for the second approach. =20
> The major reason was a concern that carrier grade NATs can allocate=20
> more than one IP address as the public address.
>
> The authors of
> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
> suggest that they addressed the concerns about the token only solution=20
> (see section 9 of the draft) and prefer to have a token only solution.=20
> This is a change from the meeting minutes and this is the reason for=20
> the second question in this consensus call.
>
> So here are two questions for the group !!!
>
> 1. Do the people in the mailing list support the proposal to use a=20
> token based solution instead on the cookie one. (this was the=20
> preference in the consensus call in Maastricht) and we would like to=20
> confirm it on the list.
>
> 2. We have two options for the token approach, please give your=20
> preferences.
>
> 2.1 Token only solution as proposed in
> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
> 2.2 Token + cookie, in this case if the request will include only IP=20
> address the response is a token, if it also includes a port it will be=20
> a cookie. This approach is for cases were the NAT may use more than=20
> one IP address for mapping a local address.
>
>
> Please respond even if you did it on the previous consensus call.


The approach in draft-begen-avt-token-for-portmapping-01 looks to be suffic=
ient to allow a server to determine that multiple requests come from the sa=
me client, provided the client isn't multi-homed and isn't sharing an IP ad=
dress with other clients. The problem statement is not clear enough for me =
to be sure if that's sufficient.

If it were important to do so, the server could distinguish multiple client=
s sharing the same IP address if each client supplied a random nonce that w=
as taken into account when generating the token in addition to the client's=
 IP address. This may be an issue with use of fractional IP addresses in fu=
ture, and so might be something to consider adding.


TOM: I think this is a good idea. The SSRC of the client associated with th=
e SSM RTP session that is present in the portmapping request message could =
be used for that purpose (hence token is determined by server on the basis =
of IP address, time stamp  and client SSRC)

Distinguishing multi-homed clients, or clients where requests come from sev=
eral different IP addresses, is more complex. Is this an issue, or is it su=
fficient to assume that a client requests a new token when changing the add=
ress it uses?

TOM: IMO that will do..I think it is bad practice if a network stack on a h=
ost device/ NAT device allocates different IP addresses towards a applicati=
on/end-host=20

I'd also like to note that the port mapping operation is independent of the=
 RAMS drafts. It would be good if the draft the group ends up with is writt=
en as a standalone mechanism, with RAMS described as one example use case, =
rather than being written in a RAMS-specific manner as are the current draf=
ts.

TOM: The text is meant to be (only) applicable to scenarios building on SSM=
 sessions with unicast feedback, no? We wanted a fast -inband- way for a cl=
ient to communicate the client receive port for a unicast connection that i=
s linked to such a SSM. In that case, the text is sufficiently generic (e.g=
. RAMS is not even mentioned in the text, whereas the example is about retr=
ansmissions). I take it that any other parameter signaling (port, any attri=
butes,..) for any other scenario needs to leverage the SDP offer-answer mod=
el. =20

Cheers,
Colin


--=20
Colin Perkins
http://csperkins.org/



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

From csp@csperkins.org  Tue Oct 12 04:32:44 2010
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From: Colin Perkins <csp@csperkins.org>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
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Cc: 'Dan Wing' <dwing@cisco.com>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, 'IETF AVT WG' <avt@ietf.org>, Roni Even <Even.roni@huawei.com>
Subject: Re: [AVT] Revised Consensus call - On Token approach for	draft-ietf-avt-ports-for-ucast-mcast-rtp
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On 12 Oct 2010, at 06:23, Van Caenegem, Tom (Tom) wrote:

> Hi Colin,
>
> See below,
> Tom
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf  
> Of Colin Perkins
> Sent: maandag 11 oktober 2010 18:51
> To: Roni Even
> Cc: DRAGE, Keith (Keith); 'IETF AVT WG'; 'Dan Wing'
> Subject: Re: [AVT] Revised Consensus call - On Token approach for  
> draft-ietf-avt-ports-for-ucast-mcast-rtp
>
> On 8 Oct 2010, at 09:47, Roni Even wrote:
>> There were some concerns about how we drafted the consensus call so I
>> would like to try and give it another try.  Sorry for the long  
>> message
>> which I hope will clarify what are the proposed options. We would  
>> like
>> to get feedback by October 18th.
>>
>>
>> The Port Mapping Between Unicast and Multicast RTP Sessions described
>> in
>> http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast- 
>> rtp-02
>> was discussed for a couple of IETF meeting and on the list. The
>> initial approach that was agreed after we had a breakout session in
>> previous IETF meetings was to use a cookie based solution even though
>> it required more messages and you needed a cookie for every RTP
>> session.
>>
>> Before the last IETF meeting there were two proposals to update the
>> solution to a token based approach.
>>
>> In the token approach:
>> - Tokens are port-independent (cookies are port-dependent) and only
>> depend on the client's IP. That is, once a client gets its token, it
>> can use it with any of its ports and with any Feedback Target.
>> - Tokens address the concern raised by Magnus
>> (http://www.ietf.org/mail-archive/web/avt/current/msg12617.html
>> )
>> - No need for keep-alives
>>
>> We had a discussion in Maastricht and the outcome is in the meeting
>> minutes http://www.ietf.org/proceedings/78/minutes/avt.txt (see the
>> section on draft-begen-avt-token-for-portmapping). The token draft  
>> has
>> Dan Wing (who is also BEHAVE co-chair) as a co-author and he looked  
>> at
>> the solution verifying that it will work well with NATs
>>
>>
>> At the meeting the preference was for a token based solution but  
>> there
>> are were two approaches discussed.
>>
>> 1. Use the token only solution
>>
>> 2. Use a combination when if the request has only an IP address the
>> response will be a token and if it will have both IP address and port
>> the response will be a  cookie.
>>
>> According to the minutes the support was for the second approach.
>> The major reason was a concern that carrier grade NATs can allocate
>> more than one IP address as the public address.
>>
>> The authors of
>> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
>> suggest that they addressed the concerns about the token only  
>> solution
>> (see section 9 of the draft) and prefer to have a token only  
>> solution.
>> This is a change from the meeting minutes and this is the reason for
>> the second question in this consensus call.
>>
>> So here are two questions for the group !!!
>>
>> 1. Do the people in the mailing list support the proposal to use a
>> token based solution instead on the cookie one. (this was the
>> preference in the consensus call in Maastricht) and we would like to
>> confirm it on the list.
>>
>> 2. We have two options for the token approach, please give your
>> preferences.
>>
>> 2.1 Token only solution as proposed in
>> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
>> 2.2 Token + cookie, in this case if the request will include only IP
>> address the response is a token, if it also includes a port it will  
>> be
>> a cookie. This approach is for cases were the NAT may use more than
>> one IP address for mapping a local address.
>>
>>
>> Please respond even if you did it on the previous consensus call.
>
>
> The approach in draft-begen-avt-token-for-portmapping-01 looks to be  
> sufficient to allow a server to determine that multiple requests  
> come from the same client, provided the client isn't multi-homed and  
> isn't sharing an IP address with other clients. The problem  
> statement is not clear enough for me to be sure if that's sufficient.
>
> If it were important to do so, the server could distinguish multiple  
> clients sharing the same IP address if each client supplied a random  
> nonce that was taken into account when generating the token in  
> addition to the client's IP address. This may be an issue with use  
> of fractional IP addresses in future, and so might be something to  
> consider adding.
>
>
> TOM: I think this is a good idea. The SSRC of the client associated  
> with the SSM RTP session that is present in the portmapping request  
> message could be used for that purpose (hence token is determined by  
> server on the basis of IP address, time stamp  and client SSRC)

No, because the SSRC can change, and this should be persistent per  
client.


> Distinguishing multi-homed clients, or clients where requests come  
> from several different IP addresses, is more complex. Is this an  
> issue, or is it sufficient to assume that a client requests a new  
> token when changing the address it uses?
>
> TOM: IMO that will do..I think it is bad practice if a network stack  
> on a host device/ NAT device allocates different IP addresses  
> towards a application/end-host
>
> I'd also like to note that the port mapping operation is independent  
> of the RAMS drafts. It would be good if the draft the group ends up  
> with is written as a standalone mechanism, with RAMS described as  
> one example use case, rather than being written in a RAMS-specific  
> manner as are the current drafts.
>
> TOM: The text is meant to be (only) applicable to scenarios building  
> on SSM sessions with unicast feedback, no? We wanted a fast -inband-  
> way for a client to communicate the client receive port for a  
> unicast connection that is linked to such a SSM. In that case, the  
> text is sufficiently generic (e.g. RAMS is not even mentioned in the  
> text, whereas the example is about retransmissions). I take it that  
> any other parameter signaling (port, any attributes,..) for any  
> other scenario needs to leverage the SDP offer-answer model.
>
> Cheers,
> Colin
>
>
> -- 
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



-- 
Colin Perkins
http://csperkins.org/




From tom.van_caenegem@alcatel-lucent.com  Tue Oct 12 05:07:12 2010
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To: Colin Perkins <csp@csperkins.org>
Date: Tue, 12 Oct 2010 14:06:54 +0200
Thread-Topic: [AVT] Revised Consensus call - On Token approach for draft-ietf-avt-ports-for-ucast-mcast-rtp
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Subject: Re: [AVT] Revised Consensus call - On Token approach for	draft-ietf-avt-ports-for-ucast-mcast-rtp
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Yes, you are right on SSRC. A dedicated random nonce will be required.=20
Are you OK with my view on how wide (or narrow) the cookie method can be us=
ed?
Tom

-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]=20
Sent: dinsdag 12 oktober 2010 13:34
To: Van Caenegem, Tom (Tom)
Cc: Roni Even; DRAGE, Keith (Keith); 'IETF AVT WG'; 'Dan Wing'
Subject: Re: [AVT] Revised Consensus call - On Token approach for draft-iet=
f-avt-ports-for-ucast-mcast-rtp


On 12 Oct 2010, at 06:23, Van Caenegem, Tom (Tom) wrote:

> Hi Colin,
>
> See below,
> Tom
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of=20
> Colin Perkins
> Sent: maandag 11 oktober 2010 18:51
> To: Roni Even
> Cc: DRAGE, Keith (Keith); 'IETF AVT WG'; 'Dan Wing'
> Subject: Re: [AVT] Revised Consensus call - On Token approach for=20
> draft-ietf-avt-ports-for-ucast-mcast-rtp
>
> On 8 Oct 2010, at 09:47, Roni Even wrote:
>> There were some concerns about how we drafted the consensus call so I=20
>> would like to try and give it another try.  Sorry for the long=20
>> message which I hope will clarify what are the proposed options. We=20
>> would like to get feedback by October 18th.
>>
>>
>> The Port Mapping Between Unicast and Multicast RTP Sessions described=20
>> in
>> http://tools.ietf.org/html/draft-ietf-avt-ports-for-ucast-mcast-
>> rtp-02
>> was discussed for a couple of IETF meeting and on the list. The=20
>> initial approach that was agreed after we had a breakout session in=20
>> previous IETF meetings was to use a cookie based solution even though=20
>> it required more messages and you needed a cookie for every RTP=20
>> session.
>>
>> Before the last IETF meeting there were two proposals to update the=20
>> solution to a token based approach.
>>
>> In the token approach:
>> - Tokens are port-independent (cookies are port-dependent) and only=20
>> depend on the client's IP. That is, once a client gets its token, it=20
>> can use it with any of its ports and with any Feedback Target.
>> - Tokens address the concern raised by Magnus=20
>> (http://www.ietf.org/mail-archive/web/avt/current/msg12617.html
>> )
>> - No need for keep-alives
>>
>> We had a discussion in Maastricht and the outcome is in the meeting=20
>> minutes http://www.ietf.org/proceedings/78/minutes/avt.txt (see the=20
>> section on draft-begen-avt-token-for-portmapping). The token draft=20
>> has Dan Wing (who is also BEHAVE co-chair) as a co-author and he=20
>> looked at the solution verifying that it will work well with NATs
>>
>>
>> At the meeting the preference was for a token based solution but=20
>> there are were two approaches discussed.
>>
>> 1. Use the token only solution
>>
>> 2. Use a combination when if the request has only an IP address the=20
>> response will be a token and if it will have both IP address and port=20
>> the response will be a  cookie.
>>
>> According to the minutes the support was for the second approach.
>> The major reason was a concern that carrier grade NATs can allocate=20
>> more than one IP address as the public address.
>>
>> The authors of
>> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
>> suggest that they addressed the concerns about the token only=20
>> solution (see section 9 of the draft) and prefer to have a token only=20
>> solution.
>> This is a change from the meeting minutes and this is the reason for=20
>> the second question in this consensus call.
>>
>> So here are two questions for the group !!!
>>
>> 1. Do the people in the mailing list support the proposal to use a=20
>> token based solution instead on the cookie one. (this was the=20
>> preference in the consensus call in Maastricht) and we would like to=20
>> confirm it on the list.
>>
>> 2. We have two options for the token approach, please give your=20
>> preferences.
>>
>> 2.1 Token only solution as proposed in
>> http://tools.ietf.org/html/draft-begen-avt-token-for-portmapping-01
>> 2.2 Token + cookie, in this case if the request will include only IP=20
>> address the response is a token, if it also includes a port it will=20
>> be a cookie. This approach is for cases were the NAT may use more=20
>> than one IP address for mapping a local address.
>>
>>
>> Please respond even if you did it on the previous consensus call.
>
>
> The approach in draft-begen-avt-token-for-portmapping-01 looks to be=20
> sufficient to allow a server to determine that multiple requests come=20
> from the same client, provided the client isn't multi-homed and isn't=20
> sharing an IP address with other clients. The problem statement is not=20
> clear enough for me to be sure if that's sufficient.
>
> If it were important to do so, the server could distinguish multiple=20
> clients sharing the same IP address if each client supplied a random=20
> nonce that was taken into account when generating the token in=20
> addition to the client's IP address. This may be an issue with use of=20
> fractional IP addresses in future, and so might be something to=20
> consider adding.
>
>
> TOM: I think this is a good idea. The SSRC of the client associated=20
> with the SSM RTP session that is present in the portmapping request=20
> message could be used for that purpose (hence token is determined by=20
> server on the basis of IP address, time stamp  and client SSRC)

No, because the SSRC can change, and this should be persistent per client.


> Distinguishing multi-homed clients, or clients where requests come=20
> from several different IP addresses, is more complex. Is this an=20
> issue, or is it sufficient to assume that a client requests a new=20
> token when changing the address it uses?
>
> TOM: IMO that will do..I think it is bad practice if a network stack=20
> on a host device/ NAT device allocates different IP addresses towards=20
> a application/end-host
>
> I'd also like to note that the port mapping operation is independent=20
> of the RAMS drafts. It would be good if the draft the group ends up=20
> with is written as a standalone mechanism, with RAMS described as one=20
> example use case, rather than being written in a RAMS-specific manner=20
> as are the current drafts.
>
> TOM: The text is meant to be (only) applicable to scenarios building=20
> on SSM sessions with unicast feedback, no? We wanted a fast -inband-=20
> way for a client to communicate the client receive port for a unicast=20
> connection that is linked to such a SSM. In that case, the text is=20
> sufficiently generic (e.g. RAMS is not even mentioned in the text,=20
> whereas the example is about retransmissions). I take it that any=20
> other parameter signaling (port, any attributes,..) for any other=20
> scenario needs to leverage the SDP offer-answer model.
>
> Cheers,
> Colin
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--
Colin Perkins
http://csperkins.org/




From peter.musgrave@magorcorp.com  Tue Oct 12 05:12:42 2010
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Subject: Re: [AVT] New Version Notification for draft-brandenburg-avt-rtcp-for-idms-02
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Hi Ray,

That makes sense to me but I will defer to the AVT chairs w.r.t. the
ETSI interaction...

Peter Musgrave



On Tue, Oct 12, 2010 at 3:47 AM, Brandenburg, R. (Ray) van
<ray.vanbrandenburg@tno.nl> wrote:
> Hi Peter,
>
> Thanks for your comments.
>
>>
>> Should the relationship of this doc to ETSI TISPAN be mentioned in the
>> Abstract?
>
>
> I agree and will add this in the next version. While we're on the subject=
:
> the main reason we have brought this work to the IETF is because we feel =
the
> AVT WG should be the place where XR blocks are specified. If there is eno=
ugh
> interest around here, one option could be to have the AVT WG take the lea=
d
> on the standardization of this XR block, either as an informational or
> standards track document. The relevant sections of the ETSI TISPAN=A0spec
> could then reference the resulting RFC.=A0How do you feel about this?
>
> Regards,
>
> Ray van Brandenburg
>
>
>
> ________________________________
> From: Ishan Vaishnavi [mailto:ishan.vaishnavi@gmail.com]
> Sent: maandag 11 oktober 2010 17:40
> To: Peter Musgrave
> Cc: Brandenburg, R. (Ray) van; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for
> draft-brandenburg-avt-rtcp-for-idms-02
>
> Hi Peter,
> Thanks for taking time to review this document. Find my comments inline.
> Cheers,
> Ishan
>
>
>
>>
>> I lack background in the IDMS application. =A0Is the Boronat2009
>> reference available online anywhere?
>
> You need either an Elsevier account or a Science Direct one. Here is the
> link. Hope that helps.
>
>>
>> (I am interested in whether there
>> are separate XRs for audio and video and how the lipsync alignment
>> issues are handled within the overall synchronization strategy).
>>
>
> In my opinion lip synchronization is somewhat out of the scope of this
> document. The idea is that each distributed client reports its play-out
> position of its "primary" stream. This primary stream can be a virtual
> stream or a real audio, video, subtitles or images for that matter. =A0Al=
l
> other streams locally on each client synchronize to that stream, this mus=
t
> be done by the respective application. The current XR specification relat=
es
> to how to get these primary streams are synchronized across different
> geographically separated clients. Maybe it can be extended to include thi=
s
> local synchronization as well, but that's another discussion.
>
>>
>> Should the relationship of this doc to ETSI TISPAN be mentioned in the
>> Abstract?
>
> I will leave this one for Ray.
>
>
> This e-mail and its contents are subject to the DISCLAIMER at
> http://www.tno.nl/disclaimer/email.html
>

From tom.van_caenegem@alcatel-lucent.com  Tue Oct 12 07:34:25 2010
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Qin Wu <sunseawq@huawei.com>, "avt@ietf.org" <avt@ietf.org>
Date: Tue, 12 Oct 2010 16:35:23 +0200
Thread-Topic: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Hi Qin,

these are my comments on the draft 3 version:

-you know of course that DVB IPTV has defined a solution to addres NACK sto=
rms. This solution was recently also adopted by OIPF. You propose to define=
 a new dedicated message for NACK suppression, whereas in DVB/OIPF we defin=
ed that when an RTP receiver gets such a NACK message, this receiver should=
 not send a NACK itself (provided the NACK was received before the receiver=
 sends its own NACK). In a similar way, this could also be defined for FIR =
messages. I have no fundamental problem if AVT  decides to define a dedicat=
ed message, but can you provide arguments why a NACK sent by the distributi=
on source/translator in a SSM to the client is not good enough for imposing=
 NACK suppression to a receiver, by definition?

-a general comment: suppressing NACKs from receivers is a very tricky thing=
. Sometimes it may do more harm than do any good, and I feel this discussio=
n and possible negative impact is not really addressed in this draft.

-in your figure 1, the SSM is sourced by the intermediate source which host=
s the feedback target and a "loss reporter". This does not represent a real=
istic architecture as it would mean that for IPTV scenario the head-end rec=
eives all RTCP messages from all receivers. You should include architecture=
s where FT are decoupled from the SSM source.

-the definition of the "loss reporter" is somewhat obscure to me. From vari=
ous text fragments, it appears that a (NACK) FB message sent by the loss re=
porter, will result in the intermediate source/distribution source sending =
 a NACK storm suppression or FFS message to the clients in the group.  This=
 means to me that any packet loss reported by the "loss reporter" is packet=
 loss that will impact all receivers downstream of the intermediate/distrib=
ution source. If this is not the case, you will provide unneeded retransmis=
sions. Is this a correct interpretation? If so, please make this clear in t=
he definition of "loss reporter". If it is not, then mandating that getting=
 a NACK from a loss reporter results in a FFS message, is not recommended I=
MO. =20

-I do not understand why a "loss reporter" MUST not receive the FFS message=
 -which is meant to be sent on the SSM-, and I also do not get how you can =
avoid that the "loss reporter" would receive this FFS message, as a "loss r=
eporter" that is downstream of the intermediate source must be receiving th=
e SSM in order to detect missing packets.

-you introduce the "immediate reporting RTP client" terminology from the DV=
B IPTV spec that is referenced, but its definition does not align with how =
it is defined in the DVB spec. So, please,  either stick to the DVB definit=
ion or use a different term.=20

Regards
Tom=20

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Qin W=
u
Sent: zaterdag 9 oktober 2010 6:49
To: avt@ietf.org
Subject: [AVT] New Version Notification for draft-wu-avt-retransmission-sup=
ression-rtp-03

Folks,

We have updated draft-wu-avt-retransmission-suppression-rtp which should no=
w reflect the discussion we had since IETF78.=20

The feedback suppression solution has also been greatly generalized compari=
ng to the previous version.

Comments are solicited.

Regards!
-Qin
----- Original Message -----
From: <Internet-Drafts@ietf.org>
To: <i-d-announce@ietf.org>
Sent: Saturday, October 09, 2010 12:45 PM
Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt


>A New Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
>=20
> Title           : RTCP Report Extension for Feedback Suppression
> Author(s)       : W. Wu, et al.
> Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> Pages           : 24
> Date            : 2010-10-08
>=20
> This document specifies an extension to the RTCP feedback messages
> defined in the Audio-Visual Profile with Feedback (AVPF).  This
> extension allows an intermediate node in the network (such as an RTP
> translator) inform downstream receivers that sending a feedback
> message concerning a specified set of RTP messages may be
> unnecessary, or even harmful.  For example, in a source-specific
> multicast session with large fan-out, packet loss close to the media
> or distribution source of the session is detected as a loss by a
> large number of receivers.  The RTCP NACK messages used to request
> retransmission of the missing packets are all addressed to the media
> sender, or a designated feedback target.  This may result in a
> phenomenon known variously as a "NACK implosion" or "feedback storm".
> The Feedback Suppression message defined herein is used to notify
> receivers that packet loss was detected and that the sender of the
> report will either proactively recover, or no recovery is possible.
> Receivers respond to receipt of a feedback suppression message by not
> sending a feedback message (e.g. a NACK) that they otherwise would,
> This in turn reduces load on both the feedback target and on the
> network.  This document registers two new RTCP messages for Feedback
> Suppression.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supressio=
n-rtp-03.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>


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-----


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>

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <Even.roni@huawei.com>, "Qin Wu" <sunseawq@huawei.com>, <avt@ietf.org>
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> -----Original Message-----
> From: Roni Even [mailto:Even.roni@huawei.com]
> Sent: Tuesday, October 12, 2010 3:34 AM
> To: Ali C. Begen (abegen); 'Qin Wu'; avt@ietf.org
> Subject: RE: [AVT] New Version Notification =
fordraft-wu-avt-retransmission-supression-rtp-03
>=20
>=20
> Hi,
> >
> > ...
> > > 8) Section 6.1.1 says "If the loss reporter is part of group, the
> > Distribution source MUST filter this packet out and not forward
> > > it back to this loss reporter." How does it filter that loss =
reporter
> > out?
> > >
> > > [Qin]: It is one optimized behavior.
> > > One possible way to do this is that the Distribution Source knows =
the
> > address of loss reporter or know in which port it can
> > > receive the RTCP message from the loss reporter.
> > > Is that reasonable?
> >
> > Are your reports unicast or multicast? In the latter, how do you =
deal
> > with it?
>=20
> I think this has to do with your question about merging reports.

Sure, but either I am missing something or could not explain my question =
clearly. But no hurry we can discuss it in the meeting.
=20
> > > 9) What about the cases where the receiver will actually not send =
a
> > NACK despite the packet loss? Consider where there is
> > > also a FEC stream that can recover the lost packets. Should the =
loss
> > reporter listen to the FEC channel as well and act
> > > accordingly?
> > >
> > > [Qin]: Good question. I think if the reciever know it will not =
send a
> > NACK, then the distribution source should know as well. In
> >
> > Not necessarily. If the FEC is on another distribution channel =
(which
> > is usually the case), the DS does not know what the receivers are
> > getting in terms of FEC.
>=20
> From the DS perspective there are packet losses and letting the =
receiver
> know even if it uses FEC may help the receiver giving him an earlier =
warning
> allowing him to use FEC without waiting.

Early warning is not bad, could be useful but FEC was just one example. =
We probably need to think about this more.
=20
> > > such case, the distribution source may choose not to send feedback
> > suppression message to the receiver.
> >
> > I think we need to think about this a bit further. If FEC recovers =
most
> > of the errors for majority of the receivers, maybe doing this =
(feedback
> > suppression) will not be needed as much.
> >
> > > 10) You seem to assume that different loss reporters will always =
see
> > the same losses. That is not necessarily true. What
> > > should be reported back to the receivers if the reports coming to =
the
> > FT are conflicting? Do you report the union or
> > > intersection of the reports back to the receivers?
> > >
> > > [Qin]: If there are duplication reports, the Distribution source
> > should filter the duplicated report out and only send one copy
> > > of report to the reciever. What the distribution source does is to
> > create one summary report which only append different loss
> > > in this summary report.
> >
> > This does not answer my question. If the reports are identical, yes =
you
> > can remove the duplicate. But if the reports are not identical. Now =
you
> > have a decision to make? And whatever you do may actually change the
> > performance.
>=20
> This is a good feedback. I think that this is application dependent, =
the
> typical behavior can be union but if the DS knows the network topology =
it
> may take another policy. Anyhow even if the receiver gets the union =
and on
> his path the packet were not lost, it does not break since it will =
receive
> the packets. The suppression just tell the receiver to not sent NACKs. =
The
> draft does not address the behavior of the DS in the upstream =
direction and
> leave it open to the implementation.

I think we need some guidelines presented in the draft since different =
reports will show up from time to time and some recommendations would be =
useful.

-acbegen
=20
> Roni Even
>=20
> >
> > -acbegen
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt


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Date: Tue, 12 Oct 2010 17:35:28 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Hi Tom,
The semantics of sending NACK from the sender is not defined in RFC 4585. We
could add the sender semantics to NACK but the consensus so far was to use a
new message and not add new semantics to NACK. Note that the use of the
message is not only for NACK suppression.

See inline

Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Van Caenegem, Tom (Tom)
> Sent: Tuesday, October 12, 2010 4:35 PM
> To: Qin Wu; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Hi Qin,
> 
> these are my comments on the draft 3 version:
> 
> -you know of course that DVB IPTV has defined a solution to addres NACK
> storms. This solution was recently also adopted by OIPF. You propose to
> define a new dedicated message for NACK suppression, whereas in
> DVB/OIPF we defined that when an RTP receiver gets such a NACK message,
> this receiver should not send a NACK itself (provided the NACK was
> received before the receiver sends its own NACK). In a similar way,
> this could also be defined for FIR messages. I have no fundamental
> problem if AVT  decides to define a dedicated message, but can you
> provide arguments why a NACK sent by the distribution source/translator
> in a SSM to the client is not good enough for imposing NACK suppression
> to a receiver, by definition?
> 
> -a general comment: suppressing NACKs from receivers is a very tricky
> thing. Sometimes it may do more harm than do any good, and I feel this
> discussion and possible negative impact is not really addressed in this
> draft.

I am not sure what you mean here. The receivers do not suppress NACKs, the
NACK suppression is sent by intermediary devices.

> 
> -in your figure 1, the SSM is sourced by the intermediate source which
> hosts the feedback target and a "loss reporter". This does not
> represent a realistic architecture as it would mean that for IPTV
> scenario the head-end receives all RTCP messages from all receivers.
> You should include architectures where FT are decoupled from the SSM
> source.

This is not IPTV specific feature so this is an example. There can be other
cases.

> 
> -the definition of the "loss reporter" is somewhat obscure to me. From
> various text fragments, it appears that a (NACK) FB message sent by the
> loss reporter, will result in the intermediate source/distribution
> source sending  a NACK storm suppression or FFS message to the clients
> in the group.  This means to me that any packet loss reported by the
> "loss reporter" is packet loss that will impact all receivers
> downstream of the intermediate/distribution source. If this is not the
> case, you will provide unneeded retransmissions. Is this a correct
> interpretation? If so, please make this clear in the definition of
> "loss reporter". If it is not, then mandating that getting a NACK from
> a loss reporter results in a FFS message, is not recommended IMO.

The draft only says the intermediate device will send NACK suppression and
leaves open the application behavior in the upstream side. The discussion so
far suggested that it should be out of scope for the draft.

> 
> -I do not understand why a "loss reporter" MUST not receive the FFS
> message -which is meant to be sent on the SSM-, and I also do not get
> how you can avoid that the "loss reporter" would receive this FFS
> message, as a "loss reporter" that is downstream of the intermediate
> source must be receiving the SSM in order to detect missing packets.
> 
> -you introduce the "immediate reporting RTP client" terminology from
> the DVB IPTV spec that is referenced, but its definition does not align
> with how it is defined in the DVB spec. So, please,  either stick to
> the DVB definition or use a different term.
> 
> Regards
> Tom
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Qin Wu
> Sent: zaterdag 9 oktober 2010 6:49
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Folks,
> 
> We have updated draft-wu-avt-retransmission-suppression-rtp which
> should now reflect the discussion we had since IETF78.
> 
> The feedback suppression solution has also been greatly generalized
> comparing to the previous version.
> 
> Comments are solicited.
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
> 
> 
> >A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback
> > message concerning a specified set of RTP messages may be
> > unnecessary, or even harmful.  For example, in a source-specific
> > multicast session with large fan-out, packet loss close to the media
> > or distribution source of the session is detected as a loss by a
> > large number of receivers.  The RTCP NACK messages used to request
> > retransmission of the missing packets are all addressed to the media
> > sender, or a designated feedback target.  This may result in a
> > phenomenon known variously as a "NACK implosion" or "feedback storm".
> > The Feedback Suppression message defined herein is used to notify
> > receivers that packet loss was detected and that the sender of the
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by not
> > sending a feedback message (e.g. a NACK) that they otherwise would,
> > This in turn reduces load on both the feedback target and on the
> > network.  This document registers two new RTCP messages for Feedback
> > Suppression.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-
> supression-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >
> 
> 
> -----------------------------------------------------------------------
> ---------
> 
> 
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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Date: Wed, 13 Oct 2010 08:44:52 +0200
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Hi,

 

There is a proposal to swap SIPREC Thursday pm with AVT Wednesday pm. This
swap is not yet confirmed, but please for now ensure that your travel
arrangements don't preclude either slot. The final agenda should be
published this Friday, 2010-10-15.

 

Regards

Roni Even

AVT co-chair


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<p class=MsoPlainText>Hi,<o:p></o:p></p>

<p class=MsoPlainText><o:p>&nbsp;</o:p></p>

<p class=MsoPlainText>There is a proposal to swap SIPREC Thursday pm with AVT
Wednesday pm. This swap is not yet confirmed, but please for now ensure that
your travel arrangements don't preclude either slot. The final agenda should be
published this Friday, 2010-10-15.<o:p></o:p></p>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal>Regards<o:p></o:p></p>

<p class=MsoNormal>Roni Even<o:p></o:p></p>

<p class=MsoNormal>AVT co-chair<o:p></o:p></p>

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Date: Wed, 13 Oct 2010 16:12:46 +0800
From: Qin Wu <sunseawq@huawei.com>
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Hi,:
Thank Tom for your valubable reviews. I agree with Roni's clarification in response to your comments.
I will soon issue new version to address all the comments recieved on the list.
Thank for all.

Regards!
-Qin
----- Original Message ----- 
From: "Roni Even" <Even.roni@huawei.com>
To: "'Van Caenegem, Tom (Tom)'" <tom.van_caenegem@alcatel-lucent.com>; "'Qin Wu'" <sunseawq@huawei.com>; <avt@ietf.org>
Sent: Tuesday, October 12, 2010 11:35 PM
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03


> Hi Tom,
> The semantics of sending NACK from the sender is not defined in RFC 4585. We
> could add the sender semantics to NACK but the consensus so far was to use a
> new message and not add new semantics to NACK. Note that the use of the
> message is not only for NACK suppression.
> 
> See inline
> 
> Roni
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Van Caenegem, Tom (Tom)
>> Sent: Tuesday, October 12, 2010 4:35 PM
>> To: Qin Wu; avt@ietf.org
>> Subject: Re: [AVT] New Version Notification for draft-wu-avt-
>> retransmission-supression-rtp-03
>> 
>> Hi Qin,
>> 
>> these are my comments on the draft 3 version:
>> 
>> -you know of course that DVB IPTV has defined a solution to addres NACK
>> storms. This solution was recently also adopted by OIPF. You propose to
>> define a new dedicated message for NACK suppression, whereas in
>> DVB/OIPF we defined that when an RTP receiver gets such a NACK message,
>> this receiver should not send a NACK itself (provided the NACK was
>> received before the receiver sends its own NACK). In a similar way,
>> this could also be defined for FIR messages. I have no fundamental
>> problem if AVT  decides to define a dedicated message, but can you
>> provide arguments why a NACK sent by the distribution source/translator
>> in a SSM to the client is not good enough for imposing NACK suppression
>> to a receiver, by definition?
>> 
>> -a general comment: suppressing NACKs from receivers is a very tricky
>> thing. Sometimes it may do more harm than do any good, and I feel this
>> discussion and possible negative impact is not really addressed in this
>> draft.
> 
> I am not sure what you mean here. The receivers do not suppress NACKs, the
> NACK suppression is sent by intermediary devices.

[Qin]: It seems to me suppressing NACK from one dedicated reciever is not good choice as I talked with Ali on the list.
Therefore our draft will focus on how to use distribution source to suppress NACK or Feedback message
sent from downstream of distribution source. 

> 
>> 
>> -in your figure 1, the SSM is sourced by the intermediate source which
>> hosts the feedback target and a "loss reporter". This does not
>> represent a realistic architecture as it would mean that for IPTV
>> scenario the head-end receives all RTCP messages from all receivers.
>> You should include architectures where FT are decoupled from the SSM
>> source.
> 
> This is not IPTV specific feature so this is an example. There can be other
> cases.
> 
>> 
>> -the definition of the "loss reporter" is somewhat obscure to me. From
>> various text fragments, it appears that a (NACK) FB message sent by the
>> loss reporter, will result in the intermediate source/distribution
>> source sending  a NACK storm suppression or FFS message to the clients
>> in the group.  This means to me that any packet loss reported by the
>> "loss reporter" is packet loss that will impact all receivers
>> downstream of the intermediate/distribution source. If this is not the
>> case, you will provide unneeded retransmissions. Is this a correct
>> interpretation? If so, please make this clear in the definition of
>> "loss reporter". If it is not, then mandating that getting a NACK from
>> a loss reporter results in a FFS message, is not recommended IMO.
> 
> The draft only says the intermediate device will send NACK suppression and
> leaves open the application behavior in the upstream side. The discussion so
> far suggested that it should be out of scope for the draft.

[Qin]: Yes, the draft will focus on defining new NACK suppression message using RTCP and how to use such new message
to suppression feedback implosion.

> 
>> 
>> -I do not understand why a "loss reporter" MUST not receive the FFS
>> message -which is meant to be sent on the SSM-, and I also do not get
>> how you can avoid that the "loss reporter" would receive this FFS
>> message, as a "loss reporter" that is downstream of the intermediate
>> source must be receiving the SSM in order to detect missing packets.

[Qin]: I think you raise the same issue as Ali. You are right, it does not make sense to avoid the loss reporter
to receive Feedback Suppression message if the loss reporter is part of group.
I will remove the following paragraph from the draft.
"
If the Loss Detector is part of group, the Distribution source MUST 
not send the summary report back to this Loss Detector.
"
 
>> -you introduce the "immediate reporting RTP client" terminology from
>> the DVB IPTV spec that is referenced, but its definition does not align
>> with how it is defined in the DVB spec. So, please,  either stick to
>> the DVB definition or use a different term.

[Qin]: Okay, I will leave "immediate reporting RTP client" out of this draft.

>> Regards
>> Tom
>> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Qin Wu
>> Sent: zaterdag 9 oktober 2010 6:49
>> To: avt@ietf.org
>> Subject: [AVT] New Version Notification for draft-wu-avt-
>> retransmission-supression-rtp-03
>> 
>> Folks,
>> 
>> We have updated draft-wu-avt-retransmission-suppression-rtp which
>> should now reflect the discussion we had since IETF78.
>> 
>> The feedback suppression solution has also been greatly generalized
>> comparing to the previous version.
>> 
>> Comments are solicited.
>> 
>> Regards!
>> -Qin
>> ----- Original Message -----
>> From: <Internet-Drafts@ietf.org>
>> To: <i-d-announce@ietf.org>
>> Sent: Saturday, October 09, 2010 12:45 PM
>> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
>> 
>> 
>> >A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>> >
>> > Title           : RTCP Report Extension for Feedback Suppression
>> > Author(s)       : W. Wu, et al.
>> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
>> > Pages           : 24
>> > Date            : 2010-10-08
>> >
>> > This document specifies an extension to the RTCP feedback messages
>> > defined in the Audio-Visual Profile with Feedback (AVPF).  This
>> > extension allows an intermediate node in the network (such as an RTP
>> > translator) inform downstream receivers that sending a feedback
>> > message concerning a specified set of RTP messages may be
>> > unnecessary, or even harmful.  For example, in a source-specific
>> > multicast session with large fan-out, packet loss close to the media
>> > or distribution source of the session is detected as a loss by a
>> > large number of receivers.  The RTCP NACK messages used to request
>> > retransmission of the missing packets are all addressed to the media
>> > sender, or a designated feedback target.  This may result in a
>> > phenomenon known variously as a "NACK implosion" or "feedback storm".
>> > The Feedback Suppression message defined herein is used to notify
>> > receivers that packet loss was detected and that the sender of the
>> > report will either proactively recover, or no recovery is possible.
>> > Receivers respond to receipt of a feedback suppression message by not
>> > sending a feedback message (e.g. a NACK) that they otherwise would,
>> > This in turn reduces load on both the feedback target and on the
>> > network.  This document registers two new RTCP messages for Feedback
>> > Suppression.
>> >
>> > A URL for this Internet-Draft is:
>> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-
>> supression-rtp-03.txt
>> >
>> > Internet-Drafts are also available by anonymous FTP at:
>> > ftp://ftp.ietf.org/internet-drafts/
>> >
>> > Below is the data which will enable a MIME compliant mail reader
>> > implementation to automatically retrieve the ASCII version of the
>> > Internet-Draft.
>> >
>> 
>> 
>> -----------------------------------------------------------------------
>> ---------
>> 
>> 
>> > _______________________________________________
>> > I-D-Announce mailing list
>> > I-D-Announce@ietf.org
>> > https://www.ietf.org/mailman/listinfo/i-d-announce
>> > Internet-Draft directories: http://www.ietf.org/shadow.html
>> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> >
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>

From sunseawq@huawei.com  Wed Oct 13 01:47:24 2010
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Date: Wed, 13 Oct 2010 16:47:10 +0800
From: Qin Wu <sunseawq@huawei.com>
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Hi,
We have just issued the new version of Feedback Suppression draft. This addressed comments made by Ali, Colin and Tom.
In particular, I've tried to make it clear that this draft specify what feedback suppression message look like and how they are used. 
How the packet loss is detected and repaired is not focus of this draft.
A URL for this Draft is avaiable at:
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
Comments!

Regards!
-Qin
----- Original Message ----- 
From: <Internet-Drafts@ietf.org>
To: <i-d-announce@ietf.org>
Sent: Wednesday, October 13, 2010 4:30 PM
Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-04.txt


>A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> Title           : RTCP Report Extension for Feedback Suppression
> Author(s)       : W. Wu, et al.
> Filename        : draft-wu-avt-retransmission-supression-rtp-04.txt
> Pages           : 23
> Date            : 2010-10-13
> 
> When packet loss close to the media source or intermediary of the
> session is detected as a loss by a large number of receivers, large
> number of feedback requests used to ask for the lost RTP packets are
> all addressed to the same media source, or a designated feedback
> target.  This may result in a phenomenon known variously as a
> "feedback storm " or "feedback implosion ".
> 
> This document specifies an extension to the RTCP feedback messages
> defined in the Audio-Visual Profile with Feedback (AVPF).  This
> extension allows an intermediate node in the network (such as an RTP
> translator) inform downstream receivers that packet loss was detected
> and sending a feedback message concerning a specified set of RTP
> packets may be unnecessary, or even harmful.  Receivers respond to
> receipt of a feedback suppression message by not sending a feedback
> message (e.g. a NACK) that they otherwise would, This in turn reduces
> load on both the feedback target and on the network.  The proposed
> extension is useful for single-source multicast sessions.  In
> addition, it can be applied to any other types of RTP sessions and
> topologies that might benefit from feedback suppression mechanism.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>


--------------------------------------------------------------------------------


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Payload Format for IP-MR Speech Codec
	Author(s)       : D. Yudin
	Filename        : draft-ietf-avt-rtp-ipmr-14.txt
	Pages           : 20
	Date            : 2010-10-13

This document specifies the payload format for packetization of
SPIRIT IP-MR encoded speech signals into the real-time transport
protocol (RTP). The payload format supports transmission of multiple
frames per packet and introduced redundancy for robustness against
packet loss and bit errors.

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Subject: Re: [AVT] Fw: I-D Action:draft-wu-avt-rtcp-xr-quality-monitoring-03.txt
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From ray.vanbrandenburg@tno.nl  Wed Oct 13 04:53:02 2010
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Thread-Topic: RE: [AVT] New Version Notification for draft-brandenburg-avt-rtcp-for-idms-02
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: "Roni Even" <Even.roni@huawei.com>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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=20
Hi Roni, Keith,

As AVT chairs, what is your opinion on this? In other words, what role do y=
ou think AVT can/should play in this?=20

If it helps, I'm planning to attend the upcoming IETF meeting in Beijing an=
d would be willing to give a short introduction into IDMS and the proposed =
XR block.=20

Ray van Brandenburg


-----Original Message-----
From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
Sent: dinsdag 12 oktober 2010 14:13
To: Brandenburg, R. (Ray) van
Cc: avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-brandenburg-avt-rtcp-=
for-idms-02

Hi Ray,

That makes sense to me but I will defer to the AVT chairs w.r.t. the ETSI i=
nteraction...

Peter Musgrave



On Tue, Oct 12, 2010 at 3:47 AM, Brandenburg, R. (Ray) van <ray.vanbrandenb=
urg@tno.nl> wrote:
> Hi Peter,
>
> Thanks for your comments.
>
>>
>> Should the relationship of this doc to ETSI TISPAN be mentioned in=20
>> the Abstract?
>
>
> I agree and will add this in the next version. While we're on the subject:
> the main reason we have brought this work to the IETF is because we=20
> feel the AVT WG should be the place where XR blocks are specified. If=20
> there is enough interest around here, one option could be to have the=20
> AVT WG take the lead on the standardization of this XR block, either=20
> as an informational or standards track document. The relevant sections=20
> of the ETSI TISPAN=A0spec could then reference the resulting RFC.=A0How d=
o you feel about this?
>
> Regards,
>
> Ray van Brandenburg
>
>
>
> ________________________________
> From: Ishan Vaishnavi [mailto:ishan.vaishnavi@gmail.com]
> Sent: maandag 11 oktober 2010 17:40
> To: Peter Musgrave
> Cc: Brandenburg, R. (Ray) van; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for
> draft-brandenburg-avt-rtcp-for-idms-02
>
> Hi Peter,
> Thanks for taking time to review this document. Find my comments inline.
> Cheers,
> Ishan
>
>
>
>>
>> I lack background in the IDMS application. =A0Is the Boronat2009=20
>> reference available online anywhere?
>
> You need either an Elsevier account or a Science Direct one. Here is=20
> the link. Hope that helps.
>
>>
>> (I am interested in whether there
>> are separate XRs for audio and video and how the lipsync alignment=20
>> issues are handled within the overall synchronization strategy).
>>
>
> In my opinion lip synchronization is somewhat out of the scope of this=20
> document. The idea is that each distributed client reports its=20
> play-out position of its "primary" stream. This primary stream can be=20
> a virtual stream or a real audio, video, subtitles or images for that=20
> matter. =A0All other streams locally on each client synchronize to that=20
> stream, this must be done by the respective application. The current=20
> XR specification relates to how to get these primary streams are=20
> synchronized across different geographically separated clients. Maybe=20
> it can be extended to include this local synchronization as well, but tha=
t's another discussion.
>
>>
>> Should the relationship of this doc to ETSI TISPAN be mentioned in=20
>> the Abstract?
>
> I will leave this one for Ray.
>
>
> This e-mail and its contents are subject to the DISCLAIMER at=20
> http://www.tno.nl/disclaimer/email.html
>
This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


From sunseawq@huawei.com  Wed Oct 13 05:32:50 2010
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Date: Wed, 13 Oct 2010 20:32:30 +0800
From: Qin Wu <sunseawq@huawei.com>
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Subject: [AVT] Fw: New Version Notification for draft-wu-avt-rtcp-xr-quality-monitoring-04
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Hi, folks:
04 version is posted which addresses the comments from Colin
* Add some text to General Synchronisation Offset Metrics Block to allow
  more than two stream in the RTP multimedia session.
* Clarify the Picture Type in the terminology section
* Separate TR 101 290 decodability metrics report from this draft
Your comments and suggestions are welcome!

Regards!
-Qin
----- Original Message ----- 
From: "IETF I-D Submission Tool" <idsubmission@ietf.org>
To: <sunseawq@huawei.com>
Cc: <gwz@net-zen.net>; <Roland.Schott@telekom.de>
Sent: Wednesday, October 13, 2010 8:16 PM
Subject: New Version Notification for draft-wu-avt-rtcp-xr-quality-monitoring-04


> 
> A new version of I-D, draft-wu-avt-rtcp-xr-quality-monitoring-04.txt has been successfully submitted by Wenson Wu and posted to the IETF repository.
> 
> Filename: draft-wu-avt-rtcp-xr-quality-monitoring
> Revision: 04
> Title: RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real- time Video Quality Monitoring
> Creation_date: 2010-10-13
> WG ID: Independent Submission
> Number_of_pages: 22
> 
> Abstract:
> This document defines a set of RTP Control Protocol Extended Reports
> (RTCP XR) Report Blocks and associated SDP parameters allowing the
> report of video quality metrics, primarily for video applications of
> RTP.
>                                                                                  
> 
> 
> The IETF Secretariat.
> 
>

From Even.roni@huawei.com  Wed Oct 13 13:38:22 2010
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Date: Wed, 13 Oct 2010 22:36:39 +0200
From: Roni Even <Even.roni@huawei.com>
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Hi,

This is the new schedule for AVT in Beijing

Note that the second slot is on Thursday

 

 

AVT Session 1 Monday, 

Afternoon Session I 1300-1500

Room Name: Valley Ballroom B

----------------------------------------------

AVT Session 2 Thursday, 

Afternoon Session I 1300-1500

Room Name: Garden Ballroom 1

----------------------------------------------

 

Roni Even


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<p class=3DMsoPlainText>Hi,<o:p></o:p></p>

<p class=3DMsoPlainText>This is the new schedule for AVT in =
Beijing<o:p></o:p></p>

<p class=3DMsoPlainText>Note that the second slot is on =
Thursday<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>AVT Session 1 Monday, <o:p></o:p></p>

<p class=3DMsoPlainText>Afternoon Session I 1300-1500<o:p></o:p></p>

<p class=3DMsoPlainText>Room Name: Valley Ballroom B<o:p></o:p></p>

<p =
class=3DMsoPlainText>----------------------------------------------<o:p><=
/o:p></p>

<p class=3DMsoPlainText>AVT Session 2 Thursday, <o:p></o:p></p>

<p class=3DMsoPlainText>Afternoon Session I 1300-1500<o:p></o:p></p>

<p class=3DMsoPlainText>Room Name: Garden Ballroom 1<o:p></o:p></p>

<p =
class=3DMsoPlainText>----------------------------------------------<o:p><=
/o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

</div>

</body>

</html>

--Boundary_(ID_VqnupK+PGH4+DRo/5RyX1A)--

From tom.van_caenegem@alcatel-lucent.com  Thu Oct 14 04:06:02 2010
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Roni Even <Even.roni@huawei.com>, "'Qin Wu'" <sunseawq@huawei.com>, "avt@ietf.org" <avt@ietf.org>
Date: Thu, 14 Oct 2010 13:06:38 +0200
Thread-Topic: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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Roni, thanks for your clarifying comments. See below.=20

-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: dinsdag 12 oktober 2010 17:35
To: Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-03

Hi Tom,
The semantics of sending NACK from the sender is not defined in RFC 4585. W=
e could add the sender semantics to NACK but the consensus so far was to us=
e a new message and not add new semantics to NACK. Note that the use of the=
 message is not only for NACK suppression.


TOM: can you give me a pointer to where that concensus was reached, includi=
ng the rationale? I must have missed it.


See inline

Roni

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of=20
> Van Caenegem, Tom (Tom)
> Sent: Tuesday, October 12, 2010 4:35 PM
> To: Qin Wu; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
>=20
> Hi Qin,
>=20
> these are my comments on the draft 3 version:
>=20
> -you know of course that DVB IPTV has defined a solution to addres=20
> NACK storms. This solution was recently also adopted by OIPF. You=20
> propose to define a new dedicated message for NACK suppression,=20
> whereas in DVB/OIPF we defined that when an RTP receiver gets such a=20
> NACK message, this receiver should not send a NACK itself (provided=20
> the NACK was received before the receiver sends its own NACK). In a=20
> similar way, this could also be defined for FIR messages. I have no=20
> fundamental problem if AVT  decides to define a dedicated message, but=20
> can you provide arguments why a NACK sent by the distribution=20
> source/translator in a SSM to the client is not good enough for=20
> imposing NACK suppression to a receiver, by definition?
>=20
> -a general comment: suppressing NACKs from receivers is a very tricky=20
> thing. Sometimes it may do more harm than do any good, and I feel this=20
> discussion and possible negative impact is not really addressed in=20
> this draft.

I am not sure what you mean here. The receivers do not suppress NACKs, the =
NACK suppression is sent by intermediary devices.

TOM: Sure. The draft basically specifies the Suppression message. But in te=
rms of intermediate source/receiver behaviour, my read-out is as follows:=20
1)a intermediate source can send at any time a NACK suppression
2)there is no normative language for the receiver what to do when this mess=
age is received

In the text, the intermediate source seems to rely on a NACK from a "loss r=
eporter" for sending NACK suppression messages. Is this some kind of truste=
d device? Can it always be guaranteed that the packets this loss reporter r=
eceives or does not receive, are the same packets received/not received for=
 all receivers that get the NACK suppression (and optional the retransmissi=
on afterwards)? Why would a receiver not systematically send the NACK anyho=
w -even if it does understand the suppression message, to maximise its chan=
ce of getting a retransmission.. This is the kind of discussion that should=
 be included IMO.


>=20
> -in your figure 1, the SSM is sourced by the intermediate source which=20
> hosts the feedback target and a "loss reporter". This does not=20
> represent a realistic architecture as it would mean that for IPTV=20
> scenario the head-end receives all RTCP messages from all receivers.
> You should include architectures where FT are decoupled from the SSM=20
> source.

This is not IPTV specific feature so this is an example. There can be other=
 cases.

TOM: The (only) figure in the draft (entitled "system architecture") shows =
that the "intermediate source" is sourcing the SSM, hence is also the distr=
ibution source. The draft is also explicit about this as it says: "The inte=
rmediate source must be able to communicate with all group members in order=
 for either mechanism to be effective at suppressing feedback"=20

This means that scenarios/architectures as addressed with RAMS (where the B=
RS is co-located with the FT which is in general separate from the DS) are =
outside the scope of this work?


>=20
> -the definition of the "loss reporter" is somewhat obscure to me. From=20
> various text fragments, it appears that a (NACK) FB message sent by=20
> the loss reporter, will result in the intermediate source/distribution=20
> source sending  a NACK storm suppression or FFS message to the clients=20
> in the group.  This means to me that any packet loss reported by the=20
> "loss reporter" is packet loss that will impact all receivers=20
> downstream of the intermediate/distribution source. If this is not the=20
> case, you will provide unneeded retransmissions. Is this a correct=20
> interpretation? If so, please make this clear in the definition of=20
> "loss reporter". If it is not, then mandating that getting a NACK from=20
> a loss reporter results in a FFS message, is not recommended IMO.

The draft only says the intermediate device will send NACK suppression and =
leaves open the application behavior in the upstream side. The discussion s=
o far suggested that it should be out of scope for the draft.

TOM: My remark here was about the "loss reporter" concept, not about how -"=
plain normal"- receivers should respond on the NACK suppression message. As=
 written above, the draft's architecture in which you position the NACK sup=
pression seems restricted and not covering RAMS architecture.


>=20
> -I do not understand why a "loss reporter" MUST not receive the FFS=20
> message -which is meant to be sent on the SSM-, and I also do not get=20
> how you can avoid that the "loss reporter" would receive this FFS=20
> message, as a "loss reporter" that is downstream of the intermediate=20
> source must be receiving the SSM in order to detect missing packets.
>=20
> -you introduce the "immediate reporting RTP client" terminology from=20
> the DVB IPTV spec that is referenced, but its definition does not=20
> align with how it is defined in the DVB spec. So, please,  either=20
> stick to the DVB definition or use a different term.


TOM: I hope above two comments will be addressed and further discussed as w=
ell, and I have another question: the draft defines how one can signal in S=
DP, support for the NACK suppression.  Is that for SDP to be received by th=
e RTP receivers? If so, why must that be declared in the SDP?

Thx
Tom

>=20
> Regards
> Tom
>=20
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of=20
> Qin Wu
> Sent: zaterdag 9 oktober 2010 6:49
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
>=20
> Folks,
>=20
> We have updated draft-wu-avt-retransmission-suppression-rtp which=20
> should now reflect the discussion we had since IETF78.
>=20
> The feedback suppression solution has also been greatly generalized=20
> comparing to the previous version.
>=20
> Comments are solicited.
>=20
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
>=20
>=20
> >A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages=20
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This=20
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback=20
> > message concerning a specified set of RTP messages may be=20
> > unnecessary, or even harmful.  For example, in a source-specific=20
> > multicast session with large fan-out, packet loss close to the media=20
> > or distribution source of the session is detected as a loss by a=20
> > large number of receivers.  The RTCP NACK messages used to request=20
> > retransmission of the missing packets are all addressed to the media=20
> > sender, or a designated feedback target.  This may result in a=20
> > phenomenon known variously as a "NACK implosion" or "feedback storm".
> > The Feedback Suppression message defined herein is used to notify=20
> > receivers that packet loss was detected and that the sender of the=20
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by=20
> > not sending a feedback message (e.g. a NACK) that they otherwise=20
> > would, This in turn reduces load on both the feedback target and on=20
> > the network.  This document registers two new RTCP messages for=20
> > Feedback Suppression.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-
> supression-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader=20
> > implementation to automatically retrieve the ASCII version of the=20
> > Internet-Draft.
> >
>=20
>=20
> ----------------------------------------------------------------------
> -
> ---------
>=20
>=20
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or=20
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From sunseawq@huawei.com  Thu Oct 14 20:13:39 2010
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Date: Fri, 15 Oct 2010 11:14:46 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>, Roni Even <Even.roni@huawei.com>, avt@ietf.org
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-03
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I would like to mention that the new version of draft-wu-avt-retransmission-supression-rtp has been issued.
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of 
> Van Caenegem, Tom (Tom)
> Sent: Tuesday, October 12, 2010 4:35 PM
> To: Qin Wu; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Hi Qin,
> 
> these are my comments on the draft 3 version:
> 
> -you know of course that DVB IPTV has defined a solution to addres 
> NACK storms. This solution was recently also adopted by OIPF. You 
> propose to define a new dedicated message for NACK suppression, 
> whereas in DVB/OIPF we defined that when an RTP receiver gets such a 
> NACK message, this receiver should not send a NACK itself (provided 
> the NACK was received before the receiver sends its own NACK). In a 
> similar way, this could also be defined for FIR messages. I have no 
> fundamental problem if AVT  decides to define a dedicated message, but 
> can you provide arguments why a NACK sent by the distribution 
> source/translator in a SSM to the client is not good enough for 
> imposing NACK suppression to a receiver, by definition?
> 
> -a general comment: suppressing NACKs from receivers is a very tricky 
> thing. Sometimes it may do more harm than do any good, and I feel this 
> discussion and possible negative impact is not really addressed in 
> this draft.

I am not sure what you mean here. The receivers do not suppress NACKs, the NACK suppression is sent by intermediary devices.

TOM: Sure. The draft basically specifies the Suppression message. But in terms of intermediate source/receiver behaviour, my read-out is as follows: 
1)a intermediate source can send at any time a NACK suppression

[Qin]: No, the intermediate source follows the timing of RTCP feedback messages
  defined in the Audio-Visual Profile with Feedback (AVPF).  

2)there is no normative language for the receiver what to do when this message is received

[Qin]: You may look at the new version of draft-wu-avt-retransmission-supression-rtp:
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
According to normatve text of section 3 of draft-wu-avt-retransmission-supression-rtp-04:
the receiver will be silent when it receives feedback suppression message.
A receiver may still have sent a Feedback message before receiving a feedback suppression
 message, but further feedback messages for those sequence numbers
 will be suppressed by this technique.

In the text, the intermediate source seems to rely on a NACK from a "loss reporter" for sending NACK suppression messages. Is this some kind of trusted device? Can it always be guaranteed that the packets this loss reporter receives or does not receive, are the same packets received/not received for all receivers that get the NACK suppression (and optional the retransmission afterwards)? Why would a receiver not systematically send the NACK anyhow -even if it does understand the suppression message, to maximise its chance of getting a retransmission.. This is the kind of discussion that should be included IMO.

[Qin]: No, according to discussion so far on the list, relying on dedicated reciever as loss reporter to report packet loss by sending existing NACK may have some impact the performance. 
the intermediate source may has its own loss detection functionality and should not rely on NACK from such dedicated receiver disjointed from intermediate source.
So I have remove the case relying on NACK from a loss report from this draft. Hope this clarfiies.
Also I have moved the text associated with use cases to the example use case section. So they are just one example.


> 
> -in your figure 1, the SSM is sourced by the intermediate source which 
> hosts the feedback target and a "loss reporter". This does not 
> represent a realistic architecture as it would mean that for IPTV 
> scenario the head-end receives all RTCP messages from all receivers.
> You should include architectures where FT are decoupled from the SSM 
> source.

This is not IPTV specific feature so this is an example. There can be other cases.

TOM: The (only) figure in the draft (entitled "system architecture") shows that the "intermediate source" is sourcing the SSM, hence is also the distribution source. The draft is also explicit about this as it says: "The intermediate source must be able to communicate with all group members in order for either mechanism to be effective at suppressing feedback" 

This means that scenarios/architectures as addressed with RAMS (where the BRS is co-located with the FT which is in general separate from the DS) are outside the scope of this work?

[Qin]: Thank you for point out this. You may not realized we have removed this ambiguity in the new version of draft-wu-avt-retransmission-supression-rtp
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
The main changes is:
a. Move the last 6 paragraph to the protocol overview section.
b. Remove the ambiguity text from protocol overview section and add some text in the SSM use cases section.
This architecture will focus on specifying more generic mechanism and discuss how the mechanism works in the different use cases including the scenario as addressed with RAMS.

> 
> -the definition of the "loss reporter" is somewhat obscure to me. From 
> various text fragments, it appears that a (NACK) FB message sent by 
> the loss reporter, will result in the intermediate source/distribution 
> source sending  a NACK storm suppression or FFS message to the clients 
> in the group.  This means to me that any packet loss reported by the 
> "loss reporter" is packet loss that will impact all receivers 
> downstream of the intermediate/distribution source. If this is not the 
> case, you will provide unneeded retransmissions. Is this a correct 
> interpretation? If so, please make this clear in the definition of 
> "loss reporter". If it is not, then mandating that getting a NACK from 
> a loss reporter results in a FFS message, is not recommended IMO.

The draft only says the intermediate device will send NACK suppression and leaves open the application behavior in the upstream side. The discussion so far suggested that it should be out of scope for the draft.

TOM: My remark here was about the "loss reporter" concept, not about how -"plain normal"- receivers should respond on the NACK suppression message. As written above, the draft's architecture in which you position the NACK suppression seems restricted and not covering RAMS architecture.

[Qin] See above.

> 
> -I do not understand why a "loss reporter" MUST not receive the FFS 
> message -which is meant to be sent on the SSM-, and I also do not get 
> how you can avoid that the "loss reporter" would receive this FFS 
> message, as a "loss reporter" that is downstream of the intermediate 
> source must be receiving the SSM in order to detect missing packets.
> 
> -you introduce the "immediate reporting RTP client" terminology from 
> the DVB IPTV spec that is referenced, but its definition does not 
> align with how it is defined in the DVB spec. So, please,  either 
> stick to the DVB definition or use a different term.


TOM: I hope above two comments will be addressed and further discussed as well, 

[Qin]: Based on the discussion with Ingemar, David, Ali and you, it seems the discussion suggest that relying on NACK from dedicated receiver is not a good idea and has some harm. So I have removed this use case from the draft. 
Do it make sense?

and I have another question: the draft defines how one can signal in SDP, support for the NACK suppression.  Is that for SDP to be received by the RTP receivers?

[Qin]: Yes, the reciever should understand feedback suppression message.

 If so, why must that be declared in the SDP?

[Qin]: Any other choice? what's your proposal?


Thx
Tom

> 
> Regards
> Tom
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of 
> Qin Wu
> Sent: zaterdag 9 oktober 2010 6:49
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Folks,
> 
> We have updated draft-wu-avt-retransmission-suppression-rtp which 
> should now reflect the discussion we had since IETF78.
> 
> The feedback suppression solution has also been greatly generalized 
> comparing to the previous version.
> 
> Comments are solicited.
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
> 
> 
> >A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages 
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This 
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback 
> > message concerning a specified set of RTP messages may be 
> > unnecessary, or even harmful.  For example, in a source-specific 
> > multicast session with large fan-out, packet loss close to the media 
> > or distribution source of the session is detected as a loss by a 
> > large number of receivers.  The RTCP NACK messages used to request 
> > retransmission of the missing packets are all addressed to the media 
> > sender, or a designated feedback target.  This may result in a 
> > phenomenon known variously as a "NACK implosion" or "feedback storm".
> > The Feedback Suppression message defined herein is used to notify 
> > receivers that packet loss was detected and that the sender of the 
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by 
> > not sending a feedback message (e.g. a NACK) that they otherwise 
> > would, This in turn reduces load on both the feedback target and on 
> > the network.  This document registers two new RTCP messages for 
> > Feedback Suppression.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-
> supression-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader 
> > implementation to automatically retrieve the ASCII version of the 
> > Internet-Draft.
> >
> 
> 
> ----------------------------------------------------------------------
> -
> ---------
> 
> 
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or 
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From sunseawq@huawei.com  Fri Oct 15 00:17:04 2010
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Date: Fri, 15 Oct 2010 15:17:58 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>, Roni Even <Even.roni@huawei.com>, avt@ietf.org
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Hi, 
Thank for your comments, please see my reply inline.

Regards!
-Qin
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of 
> Van Caenegem, Tom (Tom)
> Sent: Tuesday, October 12, 2010 4:35 PM
> To: Qin Wu; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Hi Qin,
> 
> these are my comments on the draft 3 version:
> 
> -you know of course that DVB IPTV has defined a solution to addres 
> NACK storms. This solution was recently also adopted by OIPF. You 
> propose to define a new dedicated message for NACK suppression, 
> whereas in DVB/OIPF we defined that when an RTP receiver gets such a 
> NACK message, this receiver should not send a NACK itself (provided 
> the NACK was received before the receiver sends its own NACK). In a 
> similar way, this could also be defined for FIR messages. I have no 
> fundamental problem if AVT  decides to define a dedicated message, but 
> can you provide arguments why a NACK sent by the distribution 
> source/translator in a SSM to the client is not good enough for 
> imposing NACK suppression to a receiver, by definition?
> 
> -a general comment: suppressing NACKs from receivers is a very tricky 
> thing. Sometimes it may do more harm than do any good, and I feel this 
> discussion and possible negative impact is not really addressed in 
> this draft.

I am not sure what you mean here. The receivers do not suppress NACKs, the NACK suppression is sent by intermediary devices.

TOM: Sure. The draft basically specifies the Suppression message. But in terms of intermediate source/receiver behaviour, my read-out is as follows: 
1)a intermediate source can send at any time a NACK suppression

[Qin]: No, the intermediate source follows the timing of RTCP feedback messages
  defined in the Audio-Visual Profile with Feedback (AVPF).  

2)there is no normative language for the receiver what to do when this message is received

[Qin]: You may look at the new version of draft-wu-avt-retransmission-supression-rtp:
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
According to normatve text of section 3 of draft-wu-avt-retransmission-supression-rtp-04:
the receiver will be silent when it receives feedback suppression message.
A receiver may still have sent a Feedback message before receiving a feedback suppression
 message, but further feedback messages for those sequence numbers
 will be suppressed by this technique.

In the text, the intermediate source seems to rely on a NACK from a "loss reporter" for sending NACK suppression messages. Is this some kind of trusted device? Can it always be guaranteed that the packets this loss reporter receives or does not receive, are the same packets received/not received for all receivers that get the NACK suppression (and optional the retransmission afterwards)? Why would a receiver not systematically send the NACK anyhow -even if it does understand the suppression message, to maximise its chance of getting a retransmission.. This is the kind of discussion that should be included IMO.

[Qin]: No, according to discussion so far on the list, relying on dedicated reciever as loss reporter to report packet loss by sending existing NACK may have some impact the performance. 
the intermediate source may has its own loss detection functionality and should not rely on NACK from such dedicated receiver disjointed from intermediate source.
So I have remove the case relying on NACK from a loss report from this draft. Hope this clarfiies.
Also I have moved the text associated with use cases to the example use case section. So they are just one example.


> 
> -in your figure 1, the SSM is sourced by the intermediate source which 
> hosts the feedback target and a "loss reporter". This does not 
> represent a realistic architecture as it would mean that for IPTV 
> scenario the head-end receives all RTCP messages from all receivers.
> You should include architectures where FT are decoupled from the SSM 
> source.

This is not IPTV specific feature so this is an example. There can be other cases.

TOM: The (only) figure in the draft (entitled "system architecture") shows that the "intermediate source" is sourcing the SSM, hence is also the distribution source. The draft is also explicit about this as it says: "The intermediate source must be able to communicate with all group members in order for either mechanism to be effective at suppressing feedback" 

This means that scenarios/architectures as addressed with RAMS (where the BRS is co-located with the FT which is in general separate from the DS) are outside the scope of this work?

[Qin]: Thank you for pointing out this. You may not realized we have removed this ambiguity in the new version of draft-wu-avt-retransmission-supression-rtp
http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-04.txt
The main changes is:
a. Move the last 6 paragraph to the protocol overview section.
b. Remove the ambiguity text from protocol overview section and add some text in the SSM use cases section.
This architecture will focus on specifying more generic mechanism and discuss how the mechanism works in the different use cases including the scenario as addressed with RAMS.

> 
> -the definition of the "loss reporter" is somewhat obscure to me. From 
> various text fragments, it appears that a (NACK) FB message sent by 
> the loss reporter, will result in the intermediate source/distribution 
> source sending  a NACK storm suppression or FFS message to the clients 
> in the group.  This means to me that any packet loss reported by the 
> "loss reporter" is packet loss that will impact all receivers 
> downstream of the intermediate/distribution source. If this is not the 
> case, you will provide unneeded retransmissions. Is this a correct 
> interpretation? If so, please make this clear in the definition of 
> "loss reporter". If it is not, then mandating that getting a NACK from 
> a loss reporter results in a FFS message, is not recommended IMO.

The draft only says the intermediate device will send NACK suppression and leaves open the application behavior in the upstream side. The discussion so far suggested that it should be out of scope for the draft.

TOM: My remark here was about the "loss reporter" concept, not about how -"plain normal"- receivers should respond on the NACK suppression message. As written above, the draft's architecture in which you position the NACK suppression seems restricted and not covering RAMS architecture.

[Qin] See above.

> 
> -I do not understand why a "loss reporter" MUST not receive the FFS 
> message -which is meant to be sent on the SSM-, and I also do not get 
> how you can avoid that the "loss reporter" would receive this FFS 
> message, as a "loss reporter" that is downstream of the intermediate 
> source must be receiving the SSM in order to detect missing packets.
> 
> -you introduce the "immediate reporting RTP client" terminology from 
> the DVB IPTV spec that is referenced, but its definition does not 
> align with how it is defined in the DVB spec. So, please,  either 
> stick to the DVB definition or use a different term.


TOM: I hope above two comments will be addressed and further discussed as well, 

[Qin]: Based on the discussion with Ingemar, David, Ali and you, it seems the discussion suggest that relying on NACK from dedicated receiver is not a good idea and has some harm. So I have removed this use case from the draft. 
Do it make sense?

and I have another question: the draft defines how one can signal in SDP, support for the NACK suppression.  Is that for SDP to be received by the RTP receivers?

[Qin]: Yes, the reciever should understand feedback suppression message.

 If so, why must that be declared in the SDP?

[Qin]: Any other choice? what's your proposal?


Thx
Tom

> 
> Regards
> Tom
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of 
> Qin Wu
> Sent: zaterdag 9 oktober 2010 6:49
> To: avt@ietf.org
> Subject: [AVT] New Version Notification for draft-wu-avt-
> retransmission-supression-rtp-03
> 
> Folks,
> 
> We have updated draft-wu-avt-retransmission-suppression-rtp which 
> should now reflect the discussion we had since IETF78.
> 
> The feedback suppression solution has also been greatly generalized 
> comparing to the previous version.
> 
> Comments are solicited.
> 
> Regards!
> -Qin
> ----- Original Message -----
> From: <Internet-Drafts@ietf.org>
> To: <i-d-announce@ietf.org>
> Sent: Saturday, October 09, 2010 12:45 PM
> Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-03.txt
> 
> 
> >A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> > Title           : RTCP Report Extension for Feedback Suppression
> > Author(s)       : W. Wu, et al.
> > Filename        : draft-wu-avt-retransmission-supression-rtp-03.txt
> > Pages           : 24
> > Date            : 2010-10-08
> >
> > This document specifies an extension to the RTCP feedback messages 
> > defined in the Audio-Visual Profile with Feedback (AVPF).  This 
> > extension allows an intermediate node in the network (such as an RTP
> > translator) inform downstream receivers that sending a feedback 
> > message concerning a specified set of RTP messages may be 
> > unnecessary, or even harmful.  For example, in a source-specific 
> > multicast session with large fan-out, packet loss close to the media 
> > or distribution source of the session is detected as a loss by a 
> > large number of receivers.  The RTCP NACK messages used to request 
> > retransmission of the missing packets are all addressed to the media 
> > sender, or a designated feedback target.  This may result in a 
> > phenomenon known variously as a "NACK implosion" or "feedback storm".
> > The Feedback Suppression message defined herein is used to notify 
> > receivers that packet loss was detected and that the sender of the 
> > report will either proactively recover, or no recovery is possible.
> > Receivers respond to receipt of a feedback suppression message by 
> > not sending a feedback message (e.g. a NACK) that they otherwise 
> > would, This in turn reduces load on both the feedback target and on 
> > the network.  This document registers two new RTCP messages for 
> > Feedback Suppression.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-
> supression-rtp-03.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > Below is the data which will enable a MIME compliant mail reader 
> > implementation to automatically retrieve the ASCII version of the 
> > Internet-Draft.
> >
> 
> 
> ----------------------------------------------------------------------
> -
> ---------
> 
> 
> > _______________________________________________
> > I-D-Announce mailing list
> > I-D-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/i-d-announce
> > Internet-Draft directories: http://www.ietf.org/shadow.html or 
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From xiajinwei@huawei.com  Fri Oct 15 19:04:57 2010
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Date: Sat, 16 Oct 2010 10:06:07 +0800
From: Jinwei Xia <xiajinwei@huawei.com>
To: 'IETF AVT WG' <avt@ietf.org>
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Hi,

>From the discussion on IETF 78th, there were some misunderstandings of the
scope of splicing draft. So we discard previous problem statement draft
while initiate new solution draft. In this new draft, We emphysize content
insertion issue is indeed a RTP-generic rather than MPEG2 TS-specific. 

Hope the new draft can address most issues AVT concern, any comments are
very appreciated!


BR!

Jinwei 

> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
> Sent: Saturday, October 16, 2010 9:58 AM
> To: xiajinwei@huawei.com
> Subject: New Version Notification for 
> draft-xia-avt-splicing-for-rtp-00
> 
> 
> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt 
> has been successfully submitted by Jinwei Xia and posted to 
> the IETF repository.
> 
> Filename:	 draft-xia-avt-splicing-for-rtp
> Revision:	 00
> Title:		 Content Splicing for RTP Sessions
> Creation_date:	 2010-10-16
> WG ID:		 Independent Submission
> Number_of_pages: 12
> 
> Abstract:
> This memo outlines RTP splicing.  Splicing is a process that 
> allows a new multimedia stream to be inserted into current 
> multimedia stream and to be conveyed to receiver for a period 
> of time.  This memo discusses the requirements of RTP 
> splicing.  In order to satisfy the requirements, this memo 
> lists several existing intermediary network elements as 
> alternatives and evaluates whether one of alternatives can be 
> used to perform RTP splicing.
>                                                               
>                     
> 
> 
> The IETF Secretariat.
> 
> 


From sunseawq@huawei.com  Fri Oct 15 21:17:41 2010
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Date: Sat, 16 Oct 2010 12:18:49 +0800
From: Qin Wu <sunseawq@huawei.com>
To: geoff.hunt@bt.com, avt@ietf.org
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Cc: philip.arden@bt.com
Subject: Re: [AVT] Audio to discuss monitoring architecture for RTP
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Hi, 
Sorry for late reply. Thank geoff and philip for arranging this discussion. 
I am willing to work on the aspect of draft that describe transport metrics 
and application metrics. I think they are worth more works. I would like
to contribute on these aspect.

Regards!
-Qin
----- Original Message ----- 
From: <geoff.hunt@bt.com>
To: <avt@ietf.org>
Cc: <philip.arden@bt.com>
Sent: Tuesday, September 21, 2010 5:36 PM
Subject: Re: [AVT] Audio to discuss monitoring architecture for RTP


> Here are the notes of the informal AVT call to discuss a monitoring architecture for RTP,
> held on 14-Sep-2010.
> 
> In summary, a number of potential work areas were identified within the broad scope of a monitoring 
> architecture but there was no commitment of resource on the call to work on any of these topics. We 
> agreed that, if a consensus emerges on the list regarding area(s) of immediate interest where people 
> will commit to work, these area(s) could be taken forward.
> 
> Thanks to Philip for taking these notes. If you were on the call and find any errors or omissions, 
> please let us know.
> 
> Participants:
> 
> Geoff Hunt (chair)
> Philip Arden (notes)
> Gunnar Heikkila
> Keith Drage
> Qin Wu
> Glen Zorn
> Marie Josee Drouin
> Dan Romascanu
> Michael Romalho
> Christian Hoene
> 
> Keith said that the audio was an informal IETF call and asked whether 
> there was agreement to adopt the conventions of the IETF Note well for 
> the call.  All were in agreement.
> 
> 1.  Agenda
> 
> Geoff introduced the agenda as follows and asked for any additions - 
> there were none.
> 
> 1. Agenda bashing
> 2. Scene setting
> - Current chartered work
> - Existing drafts and previous work
> 3. What must be defined in an architecture?
>    - candidate topics
>      - transport metrics?
> - application metrics?
>      - how to choose re-usable metrics
>      - source stream identification
>      - measurement point identification (where/what)
>      - measurement correlation keys
>      - destination entities
>      - delivery methods
>      - packaging of metrics (compact? re-usable?)
>      - security
> 4. How much of this is within AVT remit?
> 5. Proposal for scope of AVT deliverable 
> 6. Next steps
> 
> Geoff saw two aims for the call:
> 
> i)  Agree on a short-term AVT deliverable
> ii)  Discuss any longer term deliverables, which may not necessarily be 
> part of AVT.
> 
> Keith - work needs to be defined first and then a decision can be made 
> as to where it fits. Don't distinguish between short- and long-term 
> work on this call
> 
> 2.  Scene Setting
> Geoff gave a brief summary of Monarch 01 and Monarch 00:
> 
> Monarch 01 is narrower in scope and describes how to transport XR 
> metrics (independently of application) in RTCP and how to structure the 
> format of the packets whilst also making sure that the metrics are 
> transportable.
> 
> Monarch 00 describes how to choose metrics for applications and is 
> wider in scope.  It includes application metrics for voice and requires 
> expansion to include video metrics.
> 
> Although AVT has a milestone for a monitoring architecture, neither 
> draft is adopted as an AVT WG work item
> 
> Geoff - do other documents need to be considered?
> 
> Qin - video quality monitoring needed to be addressed, and the 
> following documents need to be considered:
> RFC 4710
> RFC 4712
> RFC 3611
> He also felt that the current scope was very narrow and needed to be 
> extended.  The architecture title implied a broader scope.
> 
> Keith - scope needs to reflect the available resource.  There is no 
> point in having a work item that could cannot be resourced.
> 
> Qin - a more suitable title for the document might be something like 
> "Guidelines for extending metrics blocks using RTCP-XR".
> 
> 3. What must be defined in the architecture?
> 
> i) Transport and application metrics
> 
> Geoff - AVT has a primary obligation to define RTCP transport metric.
> 
> Qin - RFC3611 contains both transport and application metrics and 
> transport metrics should be included as part of the architecture.  The 
> architecture should also include metrics guidelines and how to classify 
> the information in metrics.
> 
> Geoff - should northbound metrics be included as well as RTP peer 
> metrics. Do peer metrics need to be distinguished from northbound 
> metrics?
> 
> Qin - agreed and said that translators play a role in extracting 
> information which may or may not be sent in RTCP.
> 
> ii) Metric re-use
> 
> Geoff - the architecture should describe transport metrics.  When Colin
> (Perkins) asked for a draft he was keen not to invent new metrics and 
> existing metrics should be re-used as far as possible for different 
> applications.
> 
> Qin and Marie both agreed with this view.
> 
> Keith - list of metrics needs to be reviewed and also assuming one 
> needs new metrics information is needed about how they can be made re-usable.
> 
> Dan - agreed.  Should the doc cover scalability in gathering and 
> reporting metrics based on experience gained in XR deployment, eg why 
> MIB development didn't succeed, and examples of probe deployments.
> 
> iii) Security
> 
> Geoff - there is also a related security issue about gathering and 
> transporting metrics.
> 
> Dan - agreed.  The impact of encrypted signals also needs to be 
> considered.
> 
> Geoff - do we need all the topics on the list and are there any 
> omissions?
> 
> iv) Source stream identification and measurement point ID
> 
> Marie - Monarch draft 01 mentioned raised a number of questions.  For 
> example in Section 4 about the identity block there are unresolved 
> questions about the overlap of identity information between XR and RR 
> information.
> 
> Geoff - need to identify the overlaps and then take them to the AVT 
> list. Also, why does RTCP not have some of the identity information 
> mentioned in Monarch 01?
> 
> Marie - missing from Monarch 01 is what could be transported in RTP as 
> opposed to RTCP.
> 
> v) Measurement correlation keys
> 
> Geoff - could we use the SSRC and IP address as a correlation key to ID 
> a stream from end to end?
> 
> Michael - the SIPPING group is working on a session ID for this purpose.
> 
> Geoff - you need a session ID to compare metrics from point A with 
> those from point B.
> 
> Michael - could a hop count identifier be used?
> 
> Geoff - measurements could be made at several nodes and then sent on, 
> but would this technique scale?
> 
> Michael - if you only use RTCP metrics, then one possibility would be 
> to enable them as required.  Ought not to enable all the metrics all 
> the time except for a basic set.  The architecture needs to define a 
> rudimentary set of metrics.
> 
> Geoff - so another aspect of the architecture would be methods for 
> control of metrics collection
> 
> vi) Destination entities and delivery methods
> 
> Geoff - is it appropriate to include a range of destination entities, 
> delivery methods and protocols?
> 
> Michael - this is a hard problem.  If you have separate RTP sessions, 
> all sorts of ID are different.  The DSPs don't have access to all the 
> information required.  Also provider A won't want to give metrics 
> relating to problems in its own network to provider B.
> 
> Geoff - are you saying that it's not worth defining interfaces because 
> of inter-operator politics?
> 
> Michael - one could define metrics for a single network.
> 
> 6. Summary and next steps
> 
> Geoff - next steps are to put out the notes.  Should AVT be doing work 
> in this area?
> 
> Keith - asked whether people are prepared to commit resource.
> 
> Gunnar - the requirements for troubleshooting and for monitoring 
> end-user quality are different. Could collect application metrics from endpoints.
> Should consider both sets of requirements.
> 
> Geoff - need to decide how they fit into the architecture.
> 
> Marie - would like to contribute, but scope needs to be narrowed.
> 
> Keith - guidelines for defining reporting blocks would be useful - or 
> do we need a review?  How much resource would be required to do this?
> 
> Geoff - should we therefore suggest a work item on guidelines for 
> defining new XR blocks?
> 
> Qin - the draft needs to be restructured as the title doesn't reflect 
> the content.
> 
> Keith - keen not to propose work items that have no resource. From AVT 
> charter perspective, key criteria are that work must be useful and that 
> it must be resourced. Just "an idea and an editor" are not enough.
> 
> Geoff - suggest putting the resource question to the AVT list and 
> propose a work item on draft guidelines for XR blocks.  Also note what 
> was discussed on the call and see what topics elicit volunteers.
> 
> Keith - mentioned next meeting in Beijing and asked whether it would be 
> possible to have a face to face meeting? This seemed unlikely for the 
> responses.  He also suggested a follow-on call once responses had been 
> obtained form the list.
> 
> Geoff - will arrange follow on call.
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

From sunseawq@huawei.com  Fri Oct 15 21:28:31 2010
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Date: Sat, 16 Oct 2010 12:29:30 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, geoff.hunt@bt.com, avt@ietf.org
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Cc: philip.arden@bt.com
Subject: Re: [AVT] Audio to discuss monitoring architecture for RTP
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Hi, Keith:
It seems we have two things to do for the next step:
a) contribute to some parts of monitoring architecture draft
or
b) contribute to monitoring topic by writing some new draft
It is not clear to me what one you suggest us to do? Would you 
like to clarify this?

Regards!
-Qin
----- Original Message ----- 
From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: <geoff.hunt@bt.com>; <avt@ietf.org>
Cc: <philip.arden@bt.com>
Sent: Wednesday, September 22, 2010 9:48 PM
Subject: Re: [AVT] Audio to discuss monitoring architecture for RTP


> All,
> 
> Please comment on the notes, and propose where we go next. It would be nice to see words like "I am prepared to work on the parts of a draft that describe this".

> My understanding is that the relevant people will not be in Beijing, so we will not be scheduling it as part of the AVT meeting there. A further conference call (which could potentially be official) has been suggested, but that requires some input before we get to that point. If you want this work, please contribute.
> 
> regards
> 
> Keith 
> 
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
>> Behalf Of geoff.hunt@bt.com
>> Sent: Tuesday, September 21, 2010 10:36 AM
>> To: avt@ietf.org
>> Cc: philip.arden@bt.com
>> Subject: Re: [AVT] Audio to discuss monitoring architecture for RTP
>> 
>> Here are the notes of the informal AVT call to discuss a 
>> monitoring architecture for RTP, held on 14-Sep-2010.
>> 
>> In summary, a number of potential work areas were identified 
>> within the broad scope of a monitoring architecture but there 
>> was no commitment of resource on the call to work on any of 
>> these topics. We agreed that, if a consensus emerges on the 
>> list regarding area(s) of immediate interest where people 
>> will commit to work, these area(s) could be taken forward.
>> 
>> Thanks to Philip for taking these notes. If you were on the 
>> call and find any errors or omissions, please let us know.
>> 
>> Participants:
>> 
>> Geoff Hunt (chair)
>> Philip Arden (notes)
>> Gunnar Heikkila
>> Keith Drage
>> Qin Wu
>> Glen Zorn
>> Marie Josee Drouin
>> Dan Romascanu
>> Michael Romalho
>> Christian Hoene
>> 
>> Keith said that the audio was an informal IETF call and asked 
>> whether there was agreement to adopt the conventions of the 
>> IETF Note well for the call.  All were in agreement.
>> 
>> 1.  Agenda
>> 
>> Geoff introduced the agenda as follows and asked for any 
>> additions - there were none.
>> 
>> 1. Agenda bashing
>> 2. Scene setting
>> - Current chartered work
>> - Existing drafts and previous work
>> 3. What must be defined in an architecture?
>>    - candidate topics
>>       - transport metrics?
>> - application metrics?
>>       - how to choose re-usable metrics
>>       - source stream identification
>>       - measurement point identification (where/what)
>>       - measurement correlation keys
>>       - destination entities
>>       - delivery methods
>>       - packaging of metrics (compact? re-usable?)
>>       - security
>> 4. How much of this is within AVT remit?
>> 5. Proposal for scope of AVT deliverable 
>> 6. Next steps
>> 
>> Geoff saw two aims for the call:
>> 
>> i)  Agree on a short-term AVT deliverable
>> ii)  Discuss any longer term deliverables, which may not 
>> necessarily be part of AVT.
>> 
>> Keith - work needs to be defined first and then a decision 
>> can be made as to where it fits. Don't distinguish between 
>> short- and long-term work on this call
>> 
>> 2.  Scene Setting
>> Geoff gave a brief summary of Monarch 01 and Monarch 00:
>> 
>> Monarch 01 is narrower in scope and describes how to 
>> transport XR metrics (independently of application) in RTCP 
>> and how to structure the format of the packets whilst also 
>> making sure that the metrics are transportable.
>> 
>> Monarch 00 describes how to choose metrics for applications 
>> and is wider in scope.  It includes application metrics for 
>> voice and requires expansion to include video metrics.
>> 
>> Although AVT has a milestone for a monitoring architecture, 
>> neither draft is adopted as an AVT WG work item
>> 
>> Geoff - do other documents need to be considered?
>> 
>> Qin - video quality monitoring needed to be addressed, and 
>> the following documents need to be considered:
>> RFC 4710
>> RFC 4712
>> RFC 3611
>> He also felt that the current scope was very narrow and 
>> needed to be extended.  The architecture title implied a 
>> broader scope.
>> 
>> Keith - scope needs to reflect the available resource.  There 
>> is no point in having a work item that could cannot be resourced.
>> 
>> Qin - a more suitable title for the document might be 
>> something like "Guidelines for extending metrics blocks using 
>> RTCP-XR".
>> 
>> 3. What must be defined in the architecture?
>> 
>> i) Transport and application metrics
>> 
>> Geoff - AVT has a primary obligation to define RTCP transport metric.
>> 
>> Qin - RFC3611 contains both transport and application metrics 
>> and transport metrics should be included as part of the 
>> architecture.  The architecture should also include metrics 
>> guidelines and how to classify the information in metrics.
>> 
>> Geoff - should northbound metrics be included as well as RTP 
>> peer metrics. Do peer metrics need to be distinguished from 
>> northbound metrics?
>> 
>> Qin - agreed and said that translators play a role in 
>> extracting information which may or may not be sent in RTCP.
>> 
>> ii) Metric re-use
>> 
>> Geoff - the architecture should describe transport metrics.  
>> When Colin
>> (Perkins) asked for a draft he was keen not to invent new 
>> metrics and existing metrics should be re-used as far as 
>> possible for different applications.
>> 
>> Qin and Marie both agreed with this view.
>> 
>> Keith - list of metrics needs to be reviewed and also 
>> assuming one needs new metrics information is needed about 
>> how they can be made re-usable.
>> 
>> Dan - agreed.  Should the doc cover scalability in gathering 
>> and reporting metrics based on experience gained in XR 
>> deployment, eg why MIB development didn't succeed, and 
>> examples of probe deployments.
>> 
>> iii) Security
>> 
>> Geoff - there is also a related security issue about 
>> gathering and transporting metrics.
>> 
>> Dan - agreed.  The impact of encrypted signals also needs to 
>> be considered.
>> 
>> Geoff - do we need all the topics on the list and are there 
>> any omissions?
>> 
>> iv) Source stream identification and measurement point ID
>> 
>> Marie - Monarch draft 01 mentioned raised a number of 
>> questions.  For example in Section 4 about the identity block 
>> there are unresolved questions about the overlap of identity 
>> information between XR and RR information.
>> 
>> Geoff - need to identify the overlaps and then take them to 
>> the AVT list. Also, why does RTCP not have some of the 
>> identity information mentioned in Monarch 01?
>> 
>> Marie - missing from Monarch 01 is what could be transported 
>> in RTP as opposed to RTCP.
>> 
>> v) Measurement correlation keys
>> 
>> Geoff - could we use the SSRC and IP address as a correlation 
>> key to ID a stream from end to end?
>> 
>> Michael - the SIPPING group is working on a session ID for 
>> this purpose.
>> 
>> Geoff - you need a session ID to compare metrics from point A 
>> with those from point B.
>> 
>> Michael - could a hop count identifier be used?
>> 
>> Geoff - measurements could be made at several nodes and then 
>> sent on, but would this technique scale?
>> 
>> Michael - if you only use RTCP metrics, then one possibility 
>> would be to enable them as required.  Ought not to enable all 
>> the metrics all the time except for a basic set.  The 
>> architecture needs to define a rudimentary set of metrics.
>> 
>> Geoff - so another aspect of the architecture would be 
>> methods for control of metrics collection
>> 
>> vi) Destination entities and delivery methods
>> 
>> Geoff - is it appropriate to include a range of destination 
>> entities, delivery methods and protocols?
>> 
>> Michael - this is a hard problem.  If you have separate RTP 
>> sessions, all sorts of ID are different.  The DSPs don't have 
>> access to all the information required.  Also provider A 
>> won't want to give metrics relating to problems in its own 
>> network to provider B.
>> 
>> Geoff - are you saying that it's not worth defining 
>> interfaces because of inter-operator politics?
>> 
>> Michael - one could define metrics for a single network.
>> 
>> 6. Summary and next steps
>> 
>> Geoff - next steps are to put out the notes.  Should AVT be 
>> doing work in this area?
>> 
>> Keith - asked whether people are prepared to commit resource.
>> 
>> Gunnar - the requirements for troubleshooting and for 
>> monitoring end-user quality are different. Could collect 
>> application metrics from endpoints.
>> Should consider both sets of requirements.
>> 
>> Geoff - need to decide how they fit into the architecture.
>> 
>> Marie - would like to contribute, but scope needs to be narrowed.
>> 
>> Keith - guidelines for defining reporting blocks would be 
>> useful - or do we need a review?  How much resource would be 
>> required to do this?
>> 
>> Geoff - should we therefore suggest a work item on guidelines 
>> for defining new XR blocks?
>> 
>> Qin - the draft needs to be restructured as the title doesn't 
>> reflect the content.
>> 
>> Keith - keen not to propose work items that have no resource. 
>> From AVT charter perspective, key criteria are that work must 
>> be useful and that it must be resourced. Just "an idea and an 
>> editor" are not enough.
>> 
>> Geoff - suggest putting the resource question to the AVT list 
>> and propose a work item on draft guidelines for XR blocks.  
>> Also note what was discussed on the call and see what topics 
>> elicit volunteers.
>> 
>> Keith - mentioned next meeting in Beijing and asked whether 
>> it would be possible to have a face to face meeting? This 
>> seemed unlikely for the responses.  He also suggested a 
>> follow-on call once responses had been obtained form the list.
>> 
>> Geoff - will arrange follow on call.
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : Unicast-Based Rapid Acquisition of Multicast RTP Sessions
	Author(s)       : B. Ver Steeg, et al.
	Filename        : draft-ietf-avt-rapid-acquisition-for-rtp-16.txt
	Pages           : 56
	Date            : 2010-10-16

When an RTP receiver joins a multicast session, it may need to
acquire and parse certain Reference Information before it can process
any data sent in the multicast session.  Depending on the join time,
length of the Reference Information repetition (or appearance)
interval, size of the Reference Information as well as the
application and transport properties, the time lag before an RTP
receiver can usefully consume the multicast data, which we refer to
as the Acquisition Delay, varies and can be large.  This is an
undesirable phenomenon for receivers that frequently switch among
different multicast sessions, such as video broadcasts.

In this document, we describe a method using the existing RTP and RTP
Control Protocol (RTCP) machinery that reduces the acquisition delay.
In this method, an auxiliary unicast RTP session carrying the
Reference Information to the receiver precedes/accompanies the
multicast stream.  This unicast RTP flow can be transmitted at a
faster than natural bitrate to further accelerate the acquisition.
The motivating use case for this capability is multicast applications
that carry real-time compressed audio and video.  However, the
proposed method can also be used in other types of multicast
applications where the acquisition delay is long enough to be a
problem.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rapid-acquisition-for-rtp-16.txt

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Subject: [AVT] Request for review: ARIA algorithm and its use with SRTP
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--0016e6dd961a0a02130492ddcdea
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AVT experts,

Let me make a request for the AVT WG review
on the draft regarding ARIA cipher algorithm and its use with SRTP.

ARIA is a general-purpose block cipher algorithm developed in 2003
and is published as a RFC (RFC 5794) in March 2010.

As you know, there is another standard block cipher SEED in Korea
and SEED-SRTP is published as a RFC in August 2010.
Both SEED and ARIA were established as KS(Korean Standard)
by the Ministry of Knowledge Economy of Korea.
While SEED is mainly used for for electronic commerce and financial service,
ARIA is for government use and public purpose.

For example, ARIA is used by the followings in Korea.
- Korea e-Government
- electronic identification card
- National Police Agency
- Ministry of Health and Welfare
- Korea Centers for Disease Control and Prevention

In particular, ARIA will be used in VoIP for government
and many international VoIP vendors are developing products supporting ARIA.

We believe it increases the demand for standardization of protocols related
to ARIA.

Please find the internet draft from the below.

http://tools.ietf.org/html/draft-nsri-avt-aria-srtp-00

Best regards,
Woo-Hwan Kim

--0016e6dd961a0a02130492ddcdea
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AVT experts,<br><br>Let me make a request for the AVT WG review <br>on the =
draft regarding ARIA cipher algorithm and its use with SRTP.<br><br>ARIA is=
 a general-purpose block cipher algorithm developed in 2003<br>and is publi=
shed as a RFC (RFC 5794) in March 2010. <br>
<br>As you know, there is another standard block cipher SEED in Korea<br>an=
d SEED-SRTP is published as a RFC in August 2010.<br>Both SEED and ARIA wer=
e established as KS(Korean Standard) <br>by the Ministry of Knowledge Econo=
my of Korea.<br>
While SEED is mainly used for for electronic commerce and financial service=
,<br>ARIA is for government use and public purpose.<br><br>For example, ARI=
A is used by the followings in Korea.<br>- Korea e-Government<br>- electron=
ic identification card<br>
- National Police Agency<br>- Ministry of Health and Welfare<br>- Korea Cen=
ters for Disease Control and Prevention<br><br>In particular, ARIA will be =
used in VoIP for government<br>and many international VoIP vendors are deve=
loping products supporting ARIA. <br>
We believe it increases the demand for standardization of protocols related=
 to ARIA.<br><br>Please find the internet draft from the below.<br><br><a h=
ref=3D"http://tools.ietf.org/html/draft-nsri-avt-aria-srtp-00">http://tools=
.ietf.org/html/draft-nsri-avt-aria-srtp-00</a><br>
<br>Best regards,<br>Woo-Hwan Kim

--0016e6dd961a0a02130492ddcdea--

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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: "avt@ietf.org" <avt@ietf.org>, Qin Wu <sunseawq@huawei.com>
Date: Mon, 18 Oct 2010 18:08:15 +0200
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Hi Qin,

I have some comments (look for TOM below) on the new draft version 4.
Thx for addressing them,
Tom


Section 2 Terminology

 Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.


 The function of the loss reporter may be collocated with
      or integrated into the same entity.

TOM (1): ..same entity as what? i guess you mean the DS?

TOM (2): I do not understand how you associate the port k to a loss detecto=
r.. what do you mean with that?



section 3



(..)

In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.


TOM (3) : ...based on the two above definitions ...is an intermediate node =
hence equal to a "Loss detector"?
+ How does a Loss detector differ from an RTP receiver?

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.

TOM (4) : You probably do not want to have normative language in section 6.=
1, like "How the loss detector SHOULD (..)".

TOM (5) : It is pretty clear how a Loss detector will detect packet loss on=
 its upstream link, I guess.

TOM (6):  What is the rationale for a loss detector to send a Feedback supp=
ression message to the DS.. why NOT a RTCP NACK FB?



Section 6.1.1

If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.


TOM (7) : how will the DS send only a copy of the RTCP FB report packet to =
the group impacted by packet loss. Is this done over the SSM?

In your example you seem to focus only on architectures with a DS which als=
o hosts the FT and the loss reporter.  There should be sections covering ca=
ses where the DS is disjoint from the FT, with several FTs and also several=
 loss reporters disjoint from the FT and the DS.

TOM (8) : what do you mean in above text with "RTCP FB report".. is that th=
e suppression message you define in this draft?

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model" =
it is not clear what your architecture looks like. You talk about multiple =
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on t=
hat?

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  typi=
cally an overlay of unicast is used... will the Feedback suppression messag=
e be sent over unicast as well? If so, what is gain compared to just having=
 each RTCP receiver providing each their NACK to the MCU, even in case of a=
 packet loss impacting all receivers?

Tom (11) : coming back to my previous comment on SDP description in this dr=
aft : if this draft defines a RTCP feedback suppression message, why is the=
re need for a "announcement" in the SDP for the receivers, that indicates t=
hey may receive such feedback suppression messages?  If a receiver supports=
 the message, it will  simply act accordingly as described in the draft, no=
 need to announce that!..

Regards
Tom






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<DIV><FONT face=3DArial size=3D2><SPAN class=3D237051013-18102010>Hi=20
Qin,</SPAN></FONT></DIV>
<DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
<DIV><FONT face=3DCalibri size=3D2><SPAN class=3D237051013-18102010>I have =
some=20
comments (look for TOM below) on the new draft version 4.</SPAN></FONT></DI=
V>
<DIV><FONT face=3DCalibri size=3D2><SPAN class=3D237051013-18102010>Thx for=
 addressing=20
them,</SPAN></FONT></DIV>
<DIV><FONT face=3DCalibri size=3D2><SPAN=20
class=3D237051013-18102010>Tom</SPAN></FONT></DIV>
<DIV><FONT face=3DCalibri size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCalibri size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>Section=
 2=20
Terminology</FONT></SPAN></DIV>
<DIV><PRE><FONT face=3DCalibri size=3D2> Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
</FONT></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DCalibri siz=
e=3D2> The function of the loss reporter may be collocated with
      or integrated into the same entity.</FONT></SPAN></PRE><PRE><SPAN cla=
ss=3D237051013-18102010><FONT face=3DCalibri size=3D2>TOM (1): ..same entit=
y as what? i guess you mean the DS?</FONT></SPAN></PRE><PRE><SPAN class=3D2=
37051013-18102010><FONT face=3DCalibri size=3D2>TOM (2): I do not understan=
d how you associate the port k to a loss detector.. what do you mean with t=
hat?</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3D=
Calibri size=3D2></FONT></SPAN>&nbsp;</PRE><PRE><PRE><FONT face=3DCalibri s=
ize=3D2><SPAN class=3D237051013-18102010>section 3</SPAN></FONT></PRE><PRE>=
<FONT face=3DCalibri size=3D2></FONT>&nbsp;</PRE><PRE><SPAN class=3D2370510=
13-18102010><FONT face=3DCalibri size=3D2>(..)</FONT></SPAN></PRE><PRE><FON=
T face=3DCalibri size=3D2>In order to detect packet loss before the receive=
rs perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.
</FONT></PRE></PRE></DIV>
<DIV><!-- Converted from text/rtf format --><SPAN=20
class=3D237051013-18102010></SPAN><FONT size=3D2><FONT face=3DCalibri><SPAN=
=20
class=3D237051013-18102010>TOM (3) </SPAN>:&nbsp;...<SPAN=20
class=3D237051013-18102010>based on the two above definitions=20
...</SPAN>is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;h<SPAN=20
class=3D237051013-18102010>e</SPAN>nce&nbsp;equal&nbsp;to&nbsp;a&nbsp;"Loss=
&nbsp;detector"?&nbsp;=20
</FONT></FONT></DIV>
<DIV><SPAN class=3D237051013-18102010></SPAN><SPAN=20
class=3D237051013-18102010></SPAN><FONT face=3DCalibri size=3D2>+<SPAN=20
class=3D237051013-18102010> How does a Loss detector differ from an RTP=20
receiver?</SPAN><BR></FONT></DIV>
<DIV><FONT face=3DCalibri size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>Section=
=20
6.1</FONT></SPAN></DIV><PRE><FONT face=3DCalibri size=3D2>In order to avoid=
 the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.</FONT></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><FON=
T size=3D2><FONT face=3DCalibri>T<SPAN class=3D237051013-18102010>OM (4) : =
You probably do not want to have normative language in section 6.1, like "H=
ow the loss detector SHOULD (..)". </SPAN></FONT></FONT></PRE><PRE><SPAN cl=
ass=3D237051013-18102010></SPAN><SPAN class=3D237051013-18102010></SPAN><FO=
NT size=3D2><FONT face=3DCalibri>T<SPAN class=3D237051013-18102010>OM (5) :=
 It is pretty clear how a Loss detector will detect packet loss on its upst=
ream link, I guess.</SPAN></FONT></FONT></PRE><PRE><SPAN class=3D237051013-=
18102010><FONT face=3DCalibri size=3D2>TOM (6):  What is the rationale for =
a loss detector to send a Feedback suppression message to the DS.. why NOT =
a RTCP NACK FB?</FONT></SPAN></PRE><PRE><FONT face=3DCalibri size=3D2></FON=
T>&nbsp;</PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DCalibri si=
ze=3D2>Section 6.1.1</FONT></SPAN></PRE><PRE><FONT face=3DCalibri size=3D2>=
If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.
</FONT></PRE>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>TOM (7)=
 : how will=20
the DS send only a copy of the RTCP FB report packet to the group impacted =
by=20
packet loss. Is this done over the SSM? </FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>In your=
 example=20
you seem to focus only on architectures with a DS which also hosts the FT a=
nd=20
the loss reporter.&nbsp; There should be sections covering cases where the =
DS is=20
disjoint from the FT, with several FTs and also several loss reporters disj=
oint=20
from the FT and the DS.</FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>TOM (8)=
 : what do=20
you mean in above text with "RTCP FB report".. is that the suppression mess=
age=20
you define in this draft?</FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>TOM (9)=
 : In=20
section&nbsp; "6.1.2.&nbsp; Distribution Source Feedback Summary Model" it =
is=20
not clear what your architecture looks like. You talk about multiple=20
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on=20
that?</FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>Tom (10=
) :&nbsp;=20
In section&nbsp; 6.3.&nbsp; Multipoint Control Unit (MCU) use case,&nbsp;=20
typically an overlay of unicast is used... will the Feedback suppression me=
ssage=20
be sent over unicast as well? If so, what is gain compared to just having e=
ach=20
RTCP receiver providing each their NACK to the MCU, even in case of a packe=
t=20
loss impacting all receivers?</FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri size=3D2>Tom (11=
) : coming=20
back to my previous comment on SDP description in this draft&nbsp;:&nbsp;if=
 this=20
draft defines a RTCP feedback suppression message, why is there need for a=
=20
"announcement" in the SDP for the receivers, that indicates they may receiv=
e=20
such feedback suppression messages?&nbsp; If a receiver supports the messag=
e, it=20
will&nbsp; simply&nbsp;act accordingly as described in the draft, no need t=
o=20
announce that!.. </FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DCalibri=20
size=3D2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DArial><FONT face=3DCalib=
ri=20
size=3D2>Tom</FONT></DIV>
<DIV><FONT size=3D2><BR></FONT></DIV></FONT></SPAN>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D237051013-18102010><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV></BODY></HTML>

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From feher@tmit.bme.hu  Mon Oct 18 16:25:25 2010
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Subject: [AVT] Request for review: Approximate authentication in SRTP
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  Dear AVT experts!

Today I have uploaded an initial draft "Using approximate authentication 
with Secure Real-time Transport Protocol (SRTP)"
http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt

I'm a new member of this mailing list, but during the past years I 
already get know RFCs coming from this workgroup. I use some of them.

I'm working on a university and we do research on audio video 
transmission over wireless networks. We consider SRTP for content 
protection. However, unfortunately the current SRTP authentication 
algorithm (HMAC-SHA1) is inappropriate on wireless channels where there 
are natural bit errors. In RFC3711, SRTP allows to use a weak 
authentication in some scenarios. Approximate authentication is a weak 
authentication, but in other hand it is also a solution to have secure 
(even it is not fully secure) audio and video transmission over 
error-prone channels.

The draft I have uploded describes a kind of framework, how to use an 
approximate authentication algorithm within SRTP. There is the 
definition, how to calculate the authentication tag and you can also 
find other requirements regarding the encryption, key derivation, etc.. 
It is intentional, that I suggest no particular approximate 
authentication algorithm in this draft. I plan to submit a draft later, 
when we all agree on the basics of approximate authentication usage 
inside SRTP. In the references you can find publications regarding 
approximate authentication.

Please write your comments! I use/teach SRTP for over 5 years and I 
always thought that approximate authentication is missing. I would like 
to contribute to it. Also, I would like to give a short presentation 
related to the draft in Bejing.

Best Regards,
Gabor Feher

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    Dear AVT experts!<br>
    <br>
    Today I have uploaded an initial draft "Using approximate
    authentication with Secure Real-time Transport Protocol (SRTP)" <br>
    <a class="moz-txt-link-freetext" href="http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt">http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt</a><br>
    <br>
    I'm a new member of this mailing list, but during the past years I
    already get know RFCs coming from this workgroup. I use some of
    them.<br>
    <br>
    I'm working on a university and we do research on audio video
    transmission over wireless networks. We consider SRTP for content
    protection. However, unfortunately the current SRTP authentication
    algorithm (HMAC-SHA1) is inappropriate on wireless channels where
    there are natural bit errors. In RFC3711, SRTP allows to use a weak
    authentication in some scenarios. Approximate authentication is a
    weak authentication, but in other hand it is also a solution to have
    secure (even it is not fully secure) audio and video transmission
    over error-prone channels. <br>
    <br>
    The draft I have uploded describes a kind of framework, how to use
    an approximate authentication algorithm within SRTP. There is the
    definition, how to calculate the authentication tag and you can also
    find other requirements regarding the encryption, key derivation,
    etc.. It is intentional, that I suggest no particular approximate
    authentication algorithm in this draft. I plan to submit a draft
    later, when we all agree on the basics of approximate authentication
    usage inside SRTP. In the references you can find publications
    regarding approximate authentication.<br>
    <br>
    Please write your comments! I use/teach SRTP for over 5 years and I
    always thought that approximate authentication is missing. I would
    like to contribute to it. Also, I would like to give a short
    presentation related to the draft in Bejing.<br>
    <br>
    Best Regards,<br>
    Gabor Feher<br>
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From Even.roni@huawei.com  Mon Oct 18 16:35:57 2010
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Date: Tue, 19 Oct 2010 01:34:02 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] Request for review: Approximate authentication in SRTP
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Hi,

I will look at the draft and I noted your request for agenda time in Beijing

Roni Even

AVT co-chair

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Feh?r
G?bor
Sent: Tuesday, October 19, 2010 1:27 AM
To: avt@ietf.org
Subject: [AVT] Request for review: Approximate authentication in SRTP

 

Dear AVT experts!

Today I have uploaded an initial draft "Using approximate authentication
with Secure Real-time Transport Protocol (SRTP)" 
http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt

I'm a new member of this mailing list, but during the past years I already
get know RFCs coming from this workgroup. I use some of them.

I'm working on a university and we do research on audio video transmission
over wireless networks. We consider SRTP for content protection. However,
unfortunately the current SRTP authentication algorithm (HMAC-SHA1) is
inappropriate on wireless channels where there are natural bit errors. In
RFC3711, SRTP allows to use a weak authentication in some scenarios.
Approximate authentication is a weak authentication, but in other hand it is
also a solution to have secure (even it is not fully secure) audio and video
transmission over error-prone channels. 

The draft I have uploded describes a kind of framework, how to use an
approximate authentication algorithm within SRTP. There is the definition,
how to calculate the authentication tag and you can also find other
requirements regarding the encryption, key derivation, etc.. It is
intentional, that I suggest no particular approximate authentication
algorithm in this draft. I plan to submit a draft later, when we all agree
on the basics of approximate authentication usage inside SRTP. In the
references you can find publications regarding approximate authentication.

Please write your comments! I use/teach SRTP for over 5 years and I always
thought that approximate authentication is missing. I would like to
contribute to it. Also, I would like to give a short presentation related to
the draft in Bejing.

Best Regards,
Gabor Feher


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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I will look at the draft and I noted your request for agenda
time in Beijing<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>AVT co-chair<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> avt-bounces@ietf.org
[mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Feh?r G?bor<br>
<b>Sent:</b> Tuesday, October 19, 2010 1:27 AM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> [AVT] Request for review: Approximate authentication in SRTP<o:p></o:p></span></p>

</div>

</div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal>Dear AVT experts!<br>
<br>
Today I have uploaded an initial draft &quot;Using approximate authentication
with Secure Real-time Transport Protocol (SRTP)&quot; <br>
<a href="http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt">http://www.ietf.org/id/draft-feher-avt-approx-auth-srtp-00.txt</a><br>
<br>
I'm a new member of this mailing list, but during the past years I already get
know RFCs coming from this workgroup. I use some of them.<br>
<br>
I'm working on a university and we do research on audio video transmission over
wireless networks. We consider SRTP for content protection. However,
unfortunately the current SRTP authentication algorithm (HMAC-SHA1) is
inappropriate on wireless channels where there are natural bit errors. In
RFC3711, SRTP allows to use a weak authentication in some scenarios.
Approximate authentication is a weak authentication, but in other hand it is
also a solution to have secure (even it is not fully secure) audio and video
transmission over error-prone channels. <br>
<br>
The draft I have uploded describes a kind of framework, how to use an
approximate authentication algorithm within SRTP. There is the definition, how
to calculate the authentication tag and you can also find other requirements
regarding the encryption, key derivation, etc.. It is intentional, that I
suggest no particular approximate authentication algorithm in this draft. I
plan to submit a draft later, when we all agree on the basics of approximate
authentication usage inside SRTP. In the references you can find publications regarding
approximate authentication.<br>
<br>
Please write your comments! I use/teach SRTP for over 5 years and I always
thought that approximate authentication is missing. I would like to contribute
to it. Also, I would like to give a short presentation related to the draft in
Bejing.<br>
<br>
Best Regards,<br>
Gabor Feher<o:p></o:p></p>

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From sunseawq@huawei.com  Mon Oct 18 20:02:50 2010
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Date: Tue, 19 Oct 2010 11:04:05 +0800
From: Qin Wu <sunseawq@huawei.com>
To: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>, avt@ietf.org
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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Hi, Tom:
Thank for your comments, please see my reply inline.

Regards!
-Qin
  ----- Original Message ----- 
  From: Van Caenegem, Tom (Tom) 
  To: avt@ietf.org ; Qin Wu 
  Sent: Tuesday, October 19, 2010 12:08 AM
  Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04


  Hi Qin,

  I have some comments (look for TOM below) on the new draft version 4.
  Thx for addressing them,
  Tom


  Section 2 Terminology
 Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
 The function of the loss reporter may be collocated with
      or integrated into the same entity.TOM (1): ..same entity as what? i guess you mean the DS?[Qin]: Your are right.  In this draft, the loss detector is just functionality of DS.  Furthermore we only consider the loss detector as part of feedback target within the distribution source.Also this is just one example use case to explain how feedback suppression works in SSM use case.TOM (2): I do not understand how you associate the port k to a loss detector.. what do you mean with that?[Qin]:  Sorry for ambiguity of the description here. It actually means that the loss reporter is part of feedback target and share the same address and port.section 3 (..)In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.
TOM (3) : ...based on the two above definitions ...is an intermediate node hence equal to a "Loss detector"?  
  + How does a Loss detector differ from an RTP receiver?
  [Qin]: The loss detector is just one functionality of Distribution Source in the SSM use case.
  Maybe we should interpret it as "Loss detection functionality". How the packet loss is detected is out of scope of this document.
  Different from dedicated RTP receiver, the intermediate node does not depend on existing NACK message for Retransmission request for packet loss.
  the intermediate node can create message by itself.

  Section 6.1
In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.TOM (4) : You probably do not want to have normative language in section 6.1, like "How the loss detector SHOULD (..)". [Qin]:Sorry, I can't get it.TOM (5) : It is pretty clear how a Loss detector will detect packet loss on its upstream link, I guess.[Qin]: As I said above,  Loss detector is just functionality of some Distribution sources.  Also what it said here in section 6.1 is just one example use case.TOM (6):  What is the rationale for a loss detector to send a Feedback suppression message to the DS.. why NOT a RTCP NACK FB?[Qin]: As I explained in the last IETF presentaion,  this cause NACK scematics mismatch. That's why many people in the last IETF meeeting suggest to create new RTCP message during presentation.Section 6.1.1If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.
TOM (7) : how will the DS send only a copy of the RTCP FB report packet to the group impacted by packet loss. Is this done over the SSM? 

  [Qin]: Sorry, maybe I sould say 
  "
  Section 6.1.1
  If there are multiple Distrbitution Source with the loss detection support looking at the same RTP stream.....
  "
  Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback Suppression Report packet the group via SSM.

  In your example you seem to focus only on architectures with a DS which also hosts the FT and the loss reporter.  There should be sections covering cases where the DS is disjoint from the FT, with several FTs and also several loss reporters disjoint from the FT and the DS.

  [Qin] I am not sure we need to address all the possible SSM use cases. In this draft, we just give some examples of SSM use case.

  TOM (8) : what do you mean in above text with "RTCP FB report".. is that the suppression message you define in this draft?

  [Qin]: Yes,  RTCP Feedback Suppression Report. 

  TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model" it is not clear what your architecture looks like. You talk about multiple distribution sources.. whcih implies multiple SSMs.. Can you elaborate on that?

  [Qin]:  The scenario we are talking about in  the section 6.1.2, is to  allow multiple Distrbituion source between media source and the receiver. I will look at the this case as one special SSM case. because  each of distrbituion source resides at the upstream direction or downstream direction of other distributions souces. Does it make sense?

  Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  typically an overlay of unicast is used... will the Feedback suppression message be sent over unicast as well? If so, what is gain compared to just having each RTCP receiver providing each their NACK to the MCU, even in case of a packet loss impacting all receivers?

  [Qin]: First The MCU case will not cause NACK Implosion, instead, it may cause FIR implosion.
             Second, using Feedback suppression message can restrict the FIR message sent from receivers behind MCU and liberate the media source from  FIR implosion.

  Tom (11) : coming back to my previous comment on SDP description in this draft : if this draft defines a RTCP feedback suppression message, why is there need for a "announcement" in the SDP for the receivers, that indicates they may receive such feedback suppression messages?  If a receiver supports the message, it will  simply act accordingly as described in the draft, no need to announce that!.. 

  [Qin]: Okay, I get your point. I would like to put this as open issue, discuss it in the face to face IETF meeting.

  Regards
  Tom






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Hi Qin,

Thank you for your reply.

Reading your responses,you seem to focus really on the SSM  case with an ar=
chitecture where the single DS (there can be only one DS per SSM) hosts the=
 sole feedback target, which also hosts the loss detector function. My inte=
rpretation is also that in your draft the loss detector sends the FIR or NA=
CK supression message to the DS, which then forwards/relays this message ov=
er the SSM to all receivers, right?

My main comment remains the same: you should at least incorporate the archi=
tecture of RAMS (where there can be several FT  colocated with the BRS, whi=
ch are disjoint from the DS). In this case the FT/BRS should be able to pro=
vide a suppression indication to all receivers that report to this FT (whic=
h is a subset of all receivers that connect to the SSM), which means one sh=
ould not make use of the (original) SSM for providing the suppression indic=
ation.  Your draft provides a possible solution for only a limited use case=
, and not the most interesting one, IMO.

Further down some responses to your answers below.

Tom


________________________________
From: Qin Wu [mailto:sunseawq@huawei.com]
Sent: dinsdag 19 oktober 2010 5:04
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi, Tom:
Thank for your comments, please see my reply inline.

Regards!
-Qin
----- Original Message -----
From: Van Caenegem, Tom (Tom)<mailto:tom.van_caenegem@alcatel-lucent.com>
To: avt@ietf.org<mailto:avt@ietf.org> ; Qin Wu<mailto:sunseawq@huawei.com>
Sent: Tuesday, October 19, 2010 12:08 AM
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Qin,

I have some comments (look for TOM below) on the new draft version 4.
Thx for addressing them,
Tom


Section 2 Terminology

 Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.


 The function of the loss reporter may be collocated with
      or integrated into the same entity.

TOM (1): ..same entity as what? i guess you mean the DS?

[Qin]: Your are right.  In this draft, the loss detector is just functional=
ity of DS.  Furthermore we only consider the loss detector as part of feedb=
ack target within the distribution source.

Also this is just one example use case to explain how feedback suppression =
works in SSM use case.

TOM (2): I do not understand how you associate the port k to a loss detecto=
r.. what do you mean with that?

[Qin]:  Sorry for ambiguity of the description here. It actually means that=
 the loss reporter is part of feedback target and share the same address an=
d port.

section 3



(..)

In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.


TOM (3) : ...based on the two above definitions ...is an intermediate node =
hence equal to a "Loss detector"?
+ How does a Loss detector differ from an RTP receiver?
[Qin]: The loss detector is just one functionality of Distribution Source i=
n the SSM use case.
Maybe we should interpret it as "Loss detection functionality". How the pac=
ket loss is detected is out of scope of this document.
Different from dedicated RTP receiver, the intermediate node does not depen=
d on existing NACK message for Retransmission request for packet loss.
the intermediate node can create message by itself.

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.

TOM (4) : You probably do not want to have normative language in section 6.=
1, like "How the loss detector SHOULD (..)".

[Qin]:Sorry, I can't get it.

TOM: this is really a detail. If you want to keep the sentence, you should =
e.g. say: "How the loss detector can detect the packet loss...





TOM (5) : It is pretty clear how a Loss detector will detect packet loss on=
 its upstream link, I guess.

[Qin]: As I said above,  Loss detector is just functionality of some Distri=
bution sources.  Also what it said here in section 6.1

is just one example use case.

TOM (6):  What is the rationale for a loss detector to send a Feedback supp=
ression message to the DS.. why NOT a RTCP NACK FB?

[Qin]: As I explained in the last IETF presentaion,  this cause NACK scemat=
ics mismatch. That's why many people in the last IETF meeeting suggest to c=
reate new RTCP message during presentation.



TOM: As your assumption is that the loss reporter is part of the FT that is=
 part of the DS, then there is even no need to specify how the loss detecto=
r

reports the packet loss.  In the more generic case, it makes more sense to =
me that the loss reporter just uses a NACK (or a FIR) if it detects packet =
loss, sent towards its FT or to the DS if this one hosts the FT that the lo=
ss detector reports to . It is then up to the FT or the DS to decide whethe=
r this message triggers a NACK or FIR suppression

indication to (s subset of) the receivers.  This suppression indication doe=
s not necessarly require a new message format. A NACK/FIR  that is sent to =
a RTP receiver can be defined as a feedback suppression indication. I do no=
t see a good use case why NACKs from  receivers  would be reflected back in=
 e.g. SSM use case by DS or FT  to all the connected receivers.

If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP rece=
iver, if this is for suppression indication, there is no ambiguity possibly=
.





Section 6.1.1

If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.


TOM (7) : how will the DS send only a copy of the RTCP FB report packet to =
the group impacted by packet loss. Is this done over the SSM?

[Qin]: Sorry, maybe I sould say
"
Section 6.1.1
If there are multiple Distrbitution Source with the loss detection support =
looking at the same RTP stream.....
"
Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback Suppres=
sion Report packet the group via SSM.

In your example you seem to focus only on architectures with a DS which als=
o hosts the FT and the loss reporter.  There should be sections covering ca=
ses where the DS is disjoint from the FT, with several FTs and also several=
 loss reporters disjoint from the FT and the DS.

[Qin] I am not sure we need to address all the possible SSM use cases. In t=
his draft, we just give some examples of SSM use case.

TOM (8) : what do you mean in above text with "RTCP FB report".. is that th=
e suppression message you define in this draft?

[Qin]: Yes,  RTCP Feedback Suppression Report.

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model" =
it is not clear what your architecture looks like. You talk about multiple =
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on t=
hat?

[Qin]:  The scenario we are talking about in  the section 6.1.2, is to  all=
ow multiple Distrbituion source between media source and the receiver. I wi=
ll look at the this case as one special SSM case. because  each of distrbit=
uion source resides at the upstream direction or downstream direction of ot=
her distributions souces. Does it make sense?

TOM: RFC 5760 only defines one DS for a given SSM. If you depart from that,=
 you probably need to address and define this architecture in a better way =
in your draft.

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  typi=
cally an overlay of unicast is used... will the Feedback suppression messag=
e be sent over unicast as well? If so, what is gain compared to just having=
 each RTCP receiver providing each their NACK to the MCU, even in case of a=
 packet loss impacting all receivers?

[Qin]: First The MCU case will not cause NACK Implosion, instead, it may ca=
use FIR implosion.
           Second, using Feedback suppression message can restrict the FIR =
message sent from receivers behind MCU and liberate the media source from  =
FIR implosion.

TOM: it is not important whether this is about FIR or NACK. My point is tha=
t iso having each receiver sending a NACK/FIR, you will now have to send to=
 each receiver a NACK/FIR  suppression in the given scenario. There are no =
differences in terms of Bandwdith overhead or server load in the two cases,=
 so why bother to send suppression messages?


Tom (11) : coming back to my previous comment on SDP description in this dr=
aft : if this draft defines a RTCP feedback suppression message, why is the=
re need for a "announcement" in the SDP for the receivers, that indicates t=
hey may receive such feedback suppression messages?  If a receiver supports=
 the message, it will  simply act accordingly as described in the draft, no=
 need to announce that!..

[Qin]: Okay, I get your point. I would like to put this as open issue, disc=
uss it in the face to face IETF meeting.

Regards
Tom






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<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi Qin,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Thank you for your reply.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Reading your responses,you seem to focus really on=
 the=20
SSM&nbsp; case with an architecture where the single DS (there can be only =
one=20
DS per SSM) hosts the sole feedback target, which also hosts the loss detec=
tor=20
function. My interpretation is also that in your draft the loss detector se=
nds=20
the FIR or NACK supression message to the DS, which then forwards/relays=20
this&nbsp;message over the SSM to all receivers, right?&nbsp;=20
</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>My main comment remains the same: you should at le=
ast=20
incorporate the architecture of RAMS (where&nbsp;there&nbsp;can be several=
=20
FT&nbsp; colocated with the BRS, which are disjoint from the DS). In this c=
ase=20
the FT/BRS should be able to provide a suppression indication to all receiv=
ers=20
that report to this FT (which is a subset of all receivers that connect to =
the=20
SSM), which means one should not make use of the (original) SSM for providi=
ng=20
the suppression indication.&nbsp; Your draft provides a possible solution f=
or=20
only a limited use case, and not the most interesting one,=20
IMO.&nbsp;&nbsp;</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Further down&nbsp;some&nbsp;responses to your answ=
ers=20
below.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Tom</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><FONT=20
style=3D"BACKGROUND-COLOR: #ffffff" face=3DArial color=3D#0000ff size=3D2><=
/FONT><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Qin Wu [mailto:sunseawq@huawei.co=
m]=20
<BR><B>Sent:</B> dinsdag 19 oktober 2010 5:04<BR><B>To:</B> Van Caenegem, T=
om=20
(Tom); avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New Version Notification f=
or=20
draft-wu-avt-retransmission-supression-rtp-04<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Hi, Tom:</DIV>
<DIV>Thank for your comments, please see my reply inline.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Regards!</DIV>
<DIV>-Qin</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LE=
FT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman" =
size=3D3>----- Original=20
  Message ----- </FONT></DIV>
  <DIV style=3D"BACKGROUND: #e4e4e4; FONT: 9pt &#23435;&#20307;; font-color=
: black"><FONT=20
  face=3D"Times New Roman"><FONT size=3D3><B>From:</B> </FONT></FONT><A=20
  title=3Dtom.van_caenegem@alcatel-lucent.com=20
  href=3D"mailto:tom.van_caenegem@alcatel-lucent.com"><FONT face=3D"Times N=
ew Roman"=20
  size=3D3>Van Caenegem, Tom (Tom)</FONT></A><FONT face=3D"Times New Roman"=
 size=3D3>=20
  </FONT></DIV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman">=
<FONT size=3D3><B>To:</B>=20
  </FONT></FONT><A title=3Davt@ietf.org href=3D"mailto:avt@ietf.org"><FONT=
=20
  face=3D"Times New Roman" size=3D3>avt@ietf.org</FONT></A><FONT=20
  face=3D"Times New Roman" size=3D3> ; </FONT><A title=3Dsunseawq@huawei.co=
m=20
  href=3D"mailto:sunseawq@huawei.com"><FONT face=3D"Times New Roman" size=
=3D3>Qin=20
  Wu</FONT></A><FONT face=3D"Times New Roman" size=3D3> </FONT></DIV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman">=
<FONT=20
  size=3D3><B>Sent:</B> Tuesday, October 19, 2010 12:08 AM</FONT></FONT></D=
IV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman">=
<FONT=20
  size=3D3><B>Subject:</B> Re: [AVT] New Version Notification for=20
  draft-wu-avt-retransmission-supression-rtp-04</FONT></FONT></DIV>
  <DIV><FONT face=3D&#23435;&#20307; size=3D2></FONT><BR></DIV>
  <DIV><SPAN class=3D237051013-18102010>Hi Qin,</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>I have some comments (look for TOM =
below)=20
  on the new draft version 4.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Thx for addressing them,</SPAN></DI=
V>
  <DIV><SPAN class=3D237051013-18102010>Tom</SPAN></DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>Section 2 Terminology</SPAN></DIV>
  <DIV><PRE><FONT face=3D"Times New Roman"> Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
</FONT></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3D"Times New =
Roman"> The function of the loss reporter may be collocated with
      or integrated into the same entity.</FONT></SPAN></PRE><PRE><SPAN cla=
ss=3D237051013-18102010><FONT face=3D"Times New Roman">TOM (1): ..same enti=
ty as what? i guess you mean the DS?</FONT></SPAN></PRE><PRE><SPAN class=3D=
237051013-18102010><FONT face=3D"Times New Roman">[Qin]: Your are right.  I=
n this draft, the loss detector is just functionality of DS.  Furthermore w=
e only consider the loss detector as part of feedback target within the dis=
tribution source.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010>=
</SPAN><SPAN class=3D237051013-18102010><FONT face=3D"Times New Roman">Also=
 this is just one example use case to explain how feedback suppression work=
s in SSM use case.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010=
><FONT face=3D"Times New Roman">TOM (2): I do not understand how you associ=
ate the port k to a loss detector.. what do you mean with that?</FONT></SPA=
N></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3D"Times New Roman=
">[Qin]:  Sorry for ambiguity of the description here. It actually means th=
at the loss reporter is part of feedback target and share the same address =
and port.</FONT></SPAN></PRE><PRE><PRE><SPAN class=3D237051013-18102010><FO=
NT face=3D"Times New Roman">section 3</FONT></SPAN></PRE><PRE><FONT face=3D=
"Times New Roman">&nbsp;</FONT></PRE><PRE><SPAN class=3D237051013-18102010>=
<FONT face=3D"Times New Roman">(..)</FONT></SPAN></PRE><PRE><FONT face=3D"T=
imes New Roman">In order to detect packet loss before the receivers perceiv=
e it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.
</FONT></PRE></PRE></DIV>
  <DIV><!-- Converted from text/rtf format --><SPAN>TOM (3)=20
  </SPAN>:&nbsp;...<SPAN class=3D237051013-18102010>based on the two above=
=20
  definitions ...</SPAN>is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;h<SPAN=
=20
  class=3D237051013-18102010>e</SPAN>nce&nbsp;equal&nbsp;to&nbsp;a&nbsp;"Lo=
ss&nbsp;detector"?&nbsp;=20
  </DIV>
  <DIV><SPAN class=3D237051013-18102010></SPAN><SPAN=20
  class=3D237051013-18102010></SPAN>+<SPAN class=3D237051013-18102010> How =
does a=20
  Loss detector differ from an RTP receiver?</SPAN></DIV><SPAN=20
  class=3D237051013-18102010></SPAN></BLOCKQUOTE>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LE=
FT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV><SPAN class=3D237051013-18102010>[Qin]: The loss detector is just on=
e=20
  functionality of Distribution Source in the SSM use case.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Maybe we should&nbsp;interpret it a=
s "Loss=20
  detection functionality". How the packet loss is detected is out of scope=
 of=20
  this document.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Different from dedicated RTP receiv=
er, the=20
  intermediate node does not depend on existing NACK message for Retransmis=
sion=20
  request for packet loss.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>the intermediate node can create me=
ssage=20
  by itself.</SPAN></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>Section 6.1</SPAN></DIV><PRE><FONT =
face=3D"Times New Roman">In order to avoid the forms of NACK implosion desc=
ribed in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.</FONT></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><FON=
T face=3D"Times New Roman">T<SPAN class=3D237051013-18102010>OM (4) : You p=
robably do not want to have normative language in section 6.1, like "How th=
e loss detector SHOULD (..)". </SPAN></FONT></PRE><PRE><SPAN class=3D237051=
013-18102010><FONT face=3D"Times New Roman">[Qin]:Sorry, I can't get it.</F=
ONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-191=
02010>&nbsp;</SPAN></FONT></SPAN></PRE><PRE><SPAN class=3D237051013-1810201=
0><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-19102=
010>TOM: this is really a detail. If you want to keep the sentence, you sho=
uld e.g. say: "How the loss detector can detect the packet loss...</SPAN></=
FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial =
color=3D#0000ff size=3D2><SPAN class=3D289340207-19102010></SPAN></FONT></S=
PAN>&nbsp;</PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial co=
lor=3D#0000ff size=3D2><SPAN class=3D289340207-19102010>&nbsp;</SPAN></FONT=
></SPAN></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><SPAN class=3D23=
7051013-18102010></SPAN><FONT face=3D"Times New Roman">T<SPAN class=3D23705=
1013-18102010>OM (5) : It is pretty clear how a Loss detector will detect p=
acket loss on its upstream link, I guess.</SPAN></FONT></PRE><PRE><SPAN cla=
ss=3D237051013-18102010><FONT face=3D"Times New Roman">[Qin]: As I said abo=
ve,  Loss detector is just functionality of some Distribution sources.  Als=
o what it said here in section 6.1 </FONT></SPAN></PRE><PRE><SPAN class=3D2=
37051013-18102010><FONT face=3D"Times New Roman">is just one example use ca=
se.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><SPAN c=
lass=3D237051013-18102010><FONT face=3D"Times New Roman">TOM (6):  What is =
the rationale for a loss detector to send a Feedback suppression message to=
 the DS.. why NOT a RTCP NACK FB?</FONT></SPAN></PRE><PRE><SPAN class=3D237=
051013-18102010><FONT face=3D"Times New Roman">[Qin]: As I explained in the=
 last IETF presentaion,  this cause NACK scematics mismatch. That's why man=
y people in the last IETF meeeting suggest to create new RTCP message durin=
g presentation.</FONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN cla=
ss=3D289340207-19102010>&nbsp;</SPAN></FONT></SPAN></PRE><PRE><SPAN class=
=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2><SPAN cla=
ss=3D289340207-19102010></SPAN></FONT></SPAN>&nbsp;</PRE><PRE><SPAN class=
=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2><SPAN cla=
ss=3D289340207-19102010>TOM: As your assumption is that the loss reporter i=
s part of the FT that is part of the DS, then there is even no need to spec=
ify how the loss detector </SPAN></FONT></SPAN></PRE><PRE><SPAN class=3D237=
051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D2=
89340207-19102010>reports the packet loss.  In the more generic case, it ma=
kes more sense to me that the loss reporter just uses a NACK (or a FIR) if =
it detects packet loss, sent towards its FT or to the DS if this one hosts =
the FT that the loss detector reports to . It is then up to the FT or the D=
S to decide whether this message triggers a NACK or FIR suppression </SPAN>=
</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DAria=
l color=3D#0000ff size=3D2><SPAN class=3D289340207-19102010>indication to (=
s subset of) the receivers.  This suppression indication does not necessarl=
y require a new message format. A NACK/FIR  that is sent to a RTP receiver =
can be defined as a feedback suppression indication. I do not see a good us=
e case why NACKs from  receivers  would be reflected back in e.g. SSM use c=
ase by DS or FT  to all the connected receivers. </SPAN></FONT></SPAN></PRE=
><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff s=
ize=3D2><SPAN class=3D289340207-19102010>If we define that a NACK/FIR  for =
a SSM may only be sent TOWARDS a RTP receiver, if this is for suppression i=
ndication, there is no ambiguity possibly.   </SPAN></FONT></SPAN></PRE><PR=
E><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=
=3D2><SPAN class=3D289340207-19102010></SPAN></FONT></SPAN>&nbsp;</PRE><PRE=
><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=
=3D2><SPAN class=3D289340207-19102010>&nbsp;</SPAN></FONT></SPAN></PRE><PRE=
><SPAN class=3D237051013-18102010><FONT face=3D"Times New Roman">Section 6.=
1.1</FONT></SPAN></PRE><PRE><FONT face=3D"Times New Roman">If there are mul=
tiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.
</FONT></PRE>
  <DIV><SPAN class=3D237051013-18102010>TOM (7) : how will the DS send only=
 a copy=20
  of the RTCP FB report packet to the group impacted by packet loss. Is thi=
s=20
  done over the SSM? </SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin]: Sorry, maybe I sould say=20
  </SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>"</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Section 6.1.1</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>If there are multiple Distrbitution=
 Source=20
  with the loss detection support looking at the same RTP=20
  stream.....</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>"</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Yes,&nbsp; in the SSM use case, DS =
send=20
  only a copy of the RTCP Feedback Suppression Report packet the group via=
=20
  SSM.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>In your example you seem to focus o=
nly on=20
  architectures with a DS which also hosts the FT and the loss reporter.&nb=
sp;=20
  There should be sections covering cases where the DS is disjoint from the=
 FT,=20
  with several FTs and also several loss reporters disjoint from the FT and=
 the=20
  DS.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin] I am not sure we need to addr=
ess all=20
  the possible SSM use cases. In this draft, we just give&nbsp;some example=
s of=20
  SSM use case.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>TOM (8) : what do you mean in above=
 text=20
  with "RTCP FB report".. is that the suppression message you define in thi=
s=20
  draft?</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin]: Yes,&nbsp; RTCP Feedback=20
  Suppression Report. </SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>TOM (9) : In section&nbsp; "6.1.2.&=
nbsp;=20
  Distribution Source Feedback Summary Model" it is not clear what your=20
  architecture looks like. You talk about multiple distribution sources.. w=
hcih=20
  implies multiple SSMs.. Can you elaborate on that?</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin]:&nbsp; The scenario we are ta=
lking=20
  about in&nbsp; the section 6.1.2, is&nbsp;to &nbsp;allow multiple Distrbi=
tuion=20
  source between media source and the receiver. I will look at the this cas=
e as=20
  one special SSM case. because&nbsp; each of distrbituion source </SPAN><S=
PAN=20
  class=3D237051013-18102010>resides at the upstream direction or downstrea=
m=20
  direction of other distributions souces. Does it make sense?<SPAN=20
  class=3D289340207-19102010><FONT face=3DArial color=3D#0000ff=20
  size=3D2>&nbsp;</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010><F=
ONT=20
  face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010><F=
ONT=20
  face=3DArial color=3D#0000ff size=3D2>TOM: RFC 5760 only defines one DS f=
or a given=20
  SSM. If you depart from that, you probably need to address and=20
  define&nbsp;this architecture in a better way in your=20
  draft.</FONT>&nbsp;</SPAN></SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>Tom (10) :&nbsp; In section&nbsp;=20
  6.3.&nbsp; Multipoint Control Unit (MCU) use case,&nbsp; typically an ove=
rlay=20
  of unicast is used... will the Feedback suppression message be sent over=
=20
  unicast as well? If so, what is gain compared to just having each RTCP=20
  receiver providing each their NACK to the MCU, even in case of a packet l=
oss=20
  impacting all receivers?</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin]: First The MCU case will not =
cause=20
  NACK Implosion, instead, it may cause FIR implosion.</SPAN></DIV>
  <DIV><SPAN=20
  class=3D237051013-18102010>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
  Second, using Feedback suppression message can restrict the FIR message s=
ent=20
  from receivers&nbsp;behind MCU and liberate the media source from&nbsp; F=
IR=20
  implosion.<SPAN class=3D289340207-19102010><FONT face=3DArial color=3D#00=
00ff=20
  size=3D2>&nbsp;</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010><F=
ONT=20
  face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010><F=
ONT=20
  face=3DArial color=3D#0000ff size=3D2>TOM: it is not important whether th=
is is about=20
  FIR or NACK. My point is that iso having each receiver sending a NACK/FIR=
, you=20
  will now have to send to each receiver a NACK/FIR&nbsp; suppression in th=
e=20
  given scenario. There are no differences in terms of Bandwdith overhead o=
r=20
  server load in the two cases, so why bother to send suppression=20
  messages?</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><SPAN=20
  class=3D289340207-19102010>&nbsp;</SPAN></SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>Tom (11) : coming back to my previo=
us=20
  comment on SDP description in this draft&nbsp;:&nbsp;if this draft define=
s a=20
  RTCP feedback suppression message, why is there need for a "announcement"=
 in=20
  the SDP for the receivers, that indicates they may receive such feedback=
=20
  suppression messages?&nbsp; If a receiver supports the message, it will&n=
bsp;=20
  simply&nbsp;act accordingly as described in the draft, no need to announc=
e=20
  that!.. </SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>[Qin]: Okay, I get your point. I wo=
uld=20
  like to put this as open issue, discuss it in the face to face IETF=20
  meeting.</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010>Regards</SPAN></DIV>
  <DIV><SPAN class=3D237051013-18102010>Tom</DIV>
  <DIV><BR></DIV></SPAN>
  <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
  <DIV><SPAN=20
class=3D237051013-18102010></SPAN>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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From: Qin Wu <sunseawq@huawei.com>
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T1/ Combining pkts
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Bob Briscoe skrev 2010-08-06 00:36:
> Magnus, Ingemar, Colin, Piers, Ken,
> 
> Below are my main technical concerns with 
> draft-ietf-avt-ecn-for-rtp-02 as promised.
> I have used the line numbering here:
> <http://tools.ietf.org/idnits?url=http://tools.ietf.org/id/draft-ietf-avt-ecn-for-rtp-02.txt>
> 
> T1/ Combining pkts                      <<<---THIS EMAIL
> T2/ ECN-IP-RTCP
> T3/ Anti-Cheating
> T4/ Logging Not-ECT fall-back
> T5/ Delivery %age ECT vs Not-ECT
> T6/ Pkt Reordering around session start
> 
> 
> T1/ Combining packets: Fragmentation and Reassembly
> -------------------
> "
> 
> 
> 317           *  If the translator modifies the media stream to 
> combine or split
> 318              RTP packets, but does not otherwise transcode the media, it
> 319              must manage the ECN bits in a way analogous to that described
> 320              in Section 5.3 of [RFC3168]:
> 
> 
> "
> Don't agree. The fragmentation and reassembly rules in RFC3168 were 
> designed around TCP. They aren't applicable to RTP. See the argument 
> about RFC3168 at the time it was in last call:
> <http://www.ietf.org/mail-archive/web/tsvwg/current/msg00196.html>
> This email refers to S.11 on fragmentation of this review of 
> draft-ietf-tsvwg-ecn-01 before it became RFC3168:
> <http://www.bobbriscoe.net/projects/2020comms/ecn/protocol/ip/draft-ietf-tsvwg-ecn-ip-00.txt>
> 
> "
> 
> 
> 322              identically to the original; if several ECN marked packets are
> 323              combined into one, the outgoing packet must be either ECN-CE
> 324              marked or dropped if any of the incoming packets are ECN-CE
> 325              marked,
> "
> I suggest:
> "...the outgoing packet SHOULD be ECN-CE marked with a probability 
> proportional to the ratio of ECN-CE bytes to total bytes in the 
> incoming packets being combined. However, for simplicity, the 
> outgoing packet MAY be ECN-CE marked if any of the incoming packets 
> are ECN-CE marked. This latter option will inflate the amount of 
> congestion indicated, but it is at least safe.
> "

Ok, that does make sense. I am fine with that suggestion.

> 
> 325                      and should be ECT marked if any of the 
> incoming packets
> 326              are ECT marked.
> 
> 
> "
> I think this clause should shift to before the preceding one that 
> starts on line 323. It is a more logical order to first check whether 
> all packets being combined are ECN-capable before combining CE markings.
> (I take it that the logic here is that some RTP sources may not be 
> ECN capable, but if ECT packets are being sent by other sources, it 
> implies all potential receivers are ECN-capable.)
> 

Well, the issue here is that if one is in a probing phase and only send
some fraction as ECT marked, and the rest without. If one combines them,
we need to indicate which direction these markings should go. The
current text proposes that if any are ECT, the outgoing packet is ECT.
That way ensuring that any ECT are sent at all. In worst case all
packets during the probing are ECT marked. Thus allowing a non-capable
receiver to only detect this based on RTCP. But at least it is being
detected during the probing phase rather than later which would be if
one had a rule saying that if any of the packets being combined is
Not-ECT then mark it as NOT-ECT would result in.

> "
> 
> 
> 1757       When rounding counter values in the translated RTCP packet, the
> 1758       translator should try to ensure that they sum to the number of RTP
> 1759       packets in the pre-translation sequence number space (numOrig).  The
> 1760       translator should also try to ensure that no non-zero counter is
> 1761       rounded to a zero value, since that will lose information that a
> 1762       particular type of event has occurred.
> 
> 
> "
> An idea: Include a field in the RTP ECN feedback packet for the 
> ratio, R. Then, rather than the translator translating all the 
> numbers in the report, the host receiving the message can do the 
> translations, but it also has the original information.
> 

Hmm, that is possibly a good idea. I guess we need at least 8 bits for
such a ratio, maybe a couple more would be advantageous to get
sufficient resolution. And if I understand you correct you want the
ratio to be measured as:
R= (ECN-CE marked bytes since last report from this SSRC) / (Total bytes
received since last report from this SSRC)?

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Tue Oct 19 06:14:47 2010
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T2/ ECN-IP-RTCP
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Hi Bob,

Thanks for these comments. I am sorry that I haven't responded earlier.
I have finally gotten to the point of providing my view on the comments.

So the benefit is that RTCP loss rates should be lower when congestion
is high on the path the RTCP message takes. As RTCP packets aren't
sequence nor reported on, there isn't a easy way of using them for
congestion control directly. I am not saying impossible, but it will
require significant amount of work and consume extra space in reports
for fields related to RTCP.

For the sender of the RTCP message to be able to influence the
congestion levels on the path from itself to other end-point(s) it needs
to have media flow over that path. If it has media flowing over that
path, it can reduce its transmission rate and hope that it reduced the
congestion level.

If the RTCP sender doesn't have media flowing form it to the report
consumer, which is true for unidirectional media, like streaming and for
RTCP for SSM, or receiver only behavior in ASM. In those case the RTCP
sending end-point can't affect the congestion level. In addition I think
it is wrong to increase the delivery chance for RTCP when the RTCP
sending end-point can't really reduce the rate of RTCP either, nor get
report on that packets where ECN marked.

>From my perspective the benefit is both questionable to utilize and
minor compared to the additional complexities. Especially after we have
improved your proposal, like getting the re-routing case working better.
Because I think the obvious response to having had ECT working and
suddenly detecting ECT failure is to turn of the ECT on RTCP when
reporting this.

If we in the end do want to enable this, we can write an extension or an
update of the spec, where we override the SHALL.

Cheers

Magnus


Bob Briscoe skrev 2010-08-18 18:29:
> Piers,
> 
> At 20:17 17/08/2010, Piers O'Hanlon wrote:
>> Hi Bob,
>>
>> Thanks for the suggestion on ECN for RTCP - some comments in-line
>> where appropriate and the rest here.
>>
>> One observation is that your mechanism generally relies on
>> bi-directional flows to achieve the signalling
> 
> Looking at your inline comments, I think you've misinterpreted my
> attempt at careful wording. I said, "If B is a pure RTP receiver in a
> unidirectional session, it will never get RTCP reports containing
> counts of ECT and CE," which did not mean "it will never get RTCP
> reports". It meant "Any RTCP reports it gets will never contain
> counts of ECT or CE."
> 
> Nonetheless, I agree my proposed variation to the protocol would only
> ever turn on ECT for RTCP packets if there was some bidirectional RTP
> traffic. That's the best I could do without requiring RTCP reports to
> report on RTCP packets in the other direction. But that's better than nothing.
> 
>> but when there are
>> bidirectional RTP flows then the ECN status of the path would be
>> provided already by the respective RTCP reports in each direction.
> 
> Exactly. That's what I'm proposing to use in order to turn on ECT on
> RTCP packets, which the current draft forbids.
> 
>> It's hard to quantify the added benefit of knowing that the reverse
>> path is getting congested and whether that is worth an extra
>> 2xRTCP_reporting_period of loss....
> 
> You only get this amount of loss if there has been a re-route from a
> working ECN path to a non-compliant one. As stated elesewhere in my
> review, I think the draft is rather paranoid about this possibility.
> 1) Some people already question how much effort the IETF should put
> into making protocols robust to non-compliant muddleboxes
> 2) Once a compliant path has been found, would any reasonable person
> really expect us to worry about losses while timing out after being
> re-routed onto a another path that happens to be non-compliant? The
> conditional probability of being re-routed onto a non-compliant path
> having previously been on a compliant path must surely be low?
> 
>> I guess the probability of each
>> should hopefully be quite different but it is still a potentially
>> nasty hit to take if an ugly muddlebox falls in the way.
> 
> I agree all this is hard to quantify, but this double jeopardy is
> like the risk of a pimple growing on a spot.
> 
>> The other snag is with asymmetric paths, as the fact that the RTCP
>> path is congested may not be indicative of the forward [RTP] path.
> 
> The proposed protocol doesn't require or assume symmetric paths.
> 
> I agree the main /motivation/ I gave for ECT on RTCP is not losing
> RTCP reports /if there is/ correlation between congestion in both
> directions. But the protocol doesn't need this correlation in order
> to work. And there are other motivations anyway.
> 
>> On balance I agree with Ken that it probably isn't worth the extra
>> effort, though I can see the attraction.
> 
> There is little extra effort for the gain you get in typical cases
> (see response to Ken). We certainly should not so emphatically
> prohibit ECT on RTCP packets with the ""SHALL NOT" in the draft,
> given it can 'just work' in most deployment cases, with my proposed variation.
> 
> I think we have different assumptions about which obscure
> possibilities should be considered. For instance, I think a lot of
> the complexity dealing with re-routes onto non-compliant paths is overkill.
> 
> Personally I would leave these choices to implementers, which was the
> spirit behind my proposal to modularise the protocol spec more, so
> certain parts could be made optional.
> 
> 
> Bob
> 
> 
>> Thanks,
>>
>> Piers
>>
>> On 5 August 2010 23:39, Bob Briscoe <rbriscoe@jungle.bt.co.uk> wrote:
>>> Magnus, Ingemar, Colin, Piers, Ken,
>>>
>>> Below are my main technical concerns with draft-ietf-avt-ecn-for-rtp-02 as
>>> promised.
>>> I have used the line numbering here:
>>>
>> <http://tools.ietf.org/idnits?url=http://tools.ietf.org/id/draft-ietf-avt-ecn-for-rtp-02.txt>
>>>
>>> T1/ Combining pkts
>>> T2/ ECN-IP-RTCP                         <<<---THIS EMAIL
>>> T3/ Anti-Cheating
>>> T4/ Logging Not-ECT fall-back
>>> T5/ Delivery %age ECT vs Not-ECT
>>> T6/ Pkt Reordering around session start
>>>
>>>
>>> T2/ ECN in the IP header of RTCP reports.
>>> -------------------
>>> "
>>>
>>>
>>> 510        RTCP traffic MUST NOT be ECT marked for the following reason.
>>>
>>>
>>> "
>>> "
>>>
>>>
>>> 1047       The sender SHALL NOT include ECT marks on outgoing RTCP packets,
>>> and
>>> 1048       SHOULD NOT include ECT marks on any other outgoing control
>>> messages
>>> 1049       (e.g.  STUN [RFC5389] packets, DTLS [RFC4347] handshake packets,
>>> or
>>> 1050       ZRTP [I-D.zimmermann-avt-zrtp] control packets) that are
>>> multiplexed
>>> 1051       on the same UDP port.
>>>
>>>
>>> "
>>> If congestion rises on the reverse path it /could/ be a sign that it is
>>> rising on the forward path. Ideally, we want it to be possible
>> for receivers
>>> to mark RTCP reports as ECT so they are not more likely to be
>> lost just when
>>> they are needed to report rising congestion levels. E.g. if the congestion
>>> level on the reverse path is say, 0.5%, then 1 in 200 RTCP reports will be
>>> lost.
>>>
>>> I promised to propose a way to have ECT on RTCP packets that
>> would be robust
>>> to middleboxes dropping ECT packets following a re-route. The
>> solution makes
>>> the following assumptions:
>>> A1) that an ECN-Blocking Muddlebox drops any ECT packet, irrespective of
>>> whether it is RTP or RTCP at the app-layer.
>>> A2) that an RTP source knows how often to expect RTCP reports from
>>> receivers, so it can count when RTCP reporting intervals have
>> passed without
>>> an RTCP report.
>>>
>>> Here's the proposal:* As per the I-D, the session starts without ECT on any
>>> packets.
>>> * First sender(s) introduce ECT on RTP packets through the probing methods
>>> already described in the I-D.
>>> * Consider the unicast case of a session betw A & B.
>>>  - Once A has received 3 successive RTCP reports containing counts of ECT
>>> and CE, it can send its own RTCP reports as ECT.
>>>  - If A subsequently notices that 3 RTCP reporting periods have passed
>>> without a report, or A receives no traffic at all for time [TBA], then A
>>> reverts to sending RTCP reports as Not-ECT.
>>>  - B follows the same two rules as A.
>>>  - If B is a pure RTP receiver in a unidirectional session, it will never
>>> get RTCP reports containing counts of ECT and CE, so it will
>> never switch to
>>> sending its own RTCP reports as ECT. This is because it has no way to test
>>> if ECT packets are getting through.
>>>
>> A 'pure RTP receiver', which doesn't send RTCP isn't compliant with
>> the base RTP spec (rfc3550) - an RTP receiver [only] should still send
>> periodic RTCP Receiver Reports.  Though there are quite a few Voip
>> phones do 'forget' to send RTCP.
>>
>>
>>>  - Then if a re-route sends B>A traffic through an ECN-Blocking Middlebox,
>>> no B>A packets will get through. But after 3 RTCP reporting periods A tells
>>> B in a Not-ECT RTCP report and B reverts to Not-ECT.
>>>  - If a re-route sends both B>A and A>B traffic through an ECN-Blocking
>>> Middlebox, the above behaviour will also recover the session.
>>>
>>> * Multicast multi-source case, A -> B,C,D,E; B -> A,C,D,E; etc.
>>> - Once A has decided it can safely turn on ECT in RTP packets it sends, it
>>> watches for any RTCP reports that report reception of a positive Not-ECT
>>> count but zero ECT or CE count on packets 'A' sent. Once it has received
>>> none of these RTCP reports for three RTCP reporting rounds, 'A' can switch
>>> on ECT for its own RTCP reports. 'A' determines 3 reporting rounds by
>>> picking a couple of receivers and waiting until it receives 3 reports from
>>> each.
>>>  - B,C,D,E follow the same rule.
>>>
>>>  - A cannot test whether its own ECT RTCP reports continue to arrive at
>>> their destinations, because the destinations do not know to expect traffic
>>> from A. So if they are getting no traffic from A because of an
>>> ECN-Blocking  Muddlebox, the transport will never know, although
>>> the application or human may miss the contribution from A and reset
>>> the transport to use Not-ECT on RTCP reports.
>>>
>> Again the receivers should be sending periodic Receiver Reports so A
>> should be able to know the situation.
>>
>>>  - With receiver-initiated multicast, A also cannot test for all sorts of
>>> other blockages that might suddenly appear on paths to some destinations.
>>> Therefore, once an ECN-capable session has been established, it would be
>>> reasonable to take a leap-of-faith that re-routes are likely to continue to
>>> find ECN-capable paths.   - And if the leap-of-faith is not warranted, the
>>> session breaks until the application resets it to Not-ECT - but that's not
>>> surprising if there are broken muddleboxes on some paths.
>>> * Multicast single-source case, A -> P,Q,R,S etc.
>>>  - All follow the same rule as the multi-source case, with the same
>>> limitations.
>>>  - As in the unicast and multi-source cases, if any hosts are pure RTP
>>> receivers (P,Q,R,S), they won't ever get an RTCP report about RTP packets
>>> they have sent, so they will never set ECT on RTCP reports.
>>>
>> There is the option for all parties exchange RTCP report when using
>> SSM by employing RFC5760.
>>
>>> Comparison between ECT-RTCP and Not-ECT-RTCP:
>>> * Session recovery time following a re-route through an ECN-Blocking
>>> Middlebox:
>>>  - ECT-RTCP:           3 RTCP reporting intervals
>>>  - Not-ECT-RTCP:       1 RTCP reporting interval
>>> * ECT-RTCP has more robust RTCP reporting during rising symmetric
>> congestion
>>> * Not-ECT-RTCP involves less implementation (unless using leap-of-faith,
>>> when they are both the same).
>>>
>>> BTW, the assumption that an ECN-Muddlebox will drop any ECT
>> packets, whether
>>> RTP or RTCP, can be relaxed if we add reporting of RTCP packets from A>B in
>>> reports from B>A. But it seems a reasonable assumption so reporting on RTCP
>>> reports should not be necessary.
>>>
>>> Cheers
>>>
>>>
>>> Bob
>>>
>>>
>>>
>>> ________________________________________________________________
>>> Bob Briscoe,                                BT Innovate & Design
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>>
> 
> ________________________________________________________________
> Bob Briscoe,                                BT Innovate & Design
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


-- 

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Qin Wu <sunseawq@huawei.com>, "avt@ietf.org" <avt@ietf.org>
Date: Tue, 19 Oct 2010 15:26:18 +0200
Thread-Topic: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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Qin,

focusing on just SSM arch: I quote from your earlier response:

In this draft, the loss detector is just functionality of DS.  Furthermore =
we only consider the loss detector as part of feedback target within the di=
stribution source. Also this is just one example use case to explain how fe=
edback suppression works in SSM use case.

Again, your draft does explain in detail how feedback suppression works for=
 a SSM as described here above, but does not address the RAMS case (which a=
ssumes a single DS, one or multiple disjoint FTs, where a FT is co-position=
ed with a BRS (same IP address).

Just consider a DS with multiple disjoint FTs (as in RAMS, and also permitt=
ed by RFC 5760). Typically each FT is connected to a different branch of th=
e SSM tree.

Currently your draft imposes that the feedback suppression message is sent =
by the DS over the SSM. This is not a good solution as you either restrict =
the solution to packet losses that are common to all SSM RTP receivers (cov=
ering packet loss between media source and DS ) or you will send suppressio=
n indications (and optionally forthcoming retransmission) to all receivers =
even if some did not suffer from packet loss. Therefore the draft should al=
so define how suppression information is sent to (a subset of) the receiver=
s. It does not make sense to me that IETF would define FB suppression for a=
 architecture that does not take into account deployed SSM architectures fo=
r IPTV.

Even within the SSM architecture you are focusing on, I am very much confus=
ed: you say there is one DS that hosts the FT and loss reporting function. =
What is then the role of the intermediate sources downstream of the DS? I r=
eally would appreciate you illustrate with some more architecture drawings.

The other point I still am not convinced of, is  the use of NACK versus ded=
icated suppression message. There should be more clarity in the different a=
rchitectures you consider before we can make any statements on whether havi=
ng a NACK or FIR  that is used as suppression message towards the SSM recei=
vers, is confusing or not. I do not disagree that NACK as defined in RFC 45=
88 cannot be used as suppression message. But if we define it this way in a=
 this "suppression"  draft/RFC, and no confusions can occur, I do not see t=
he issue, especially as the syntax of the suppression message is completely=
 the same as a NACK/FIR.

Regards
Tom





________________________________
From: Qin Wu [mailto:sunseawq@huawei.com]
Sent: dinsdag 19 oktober 2010 12:50
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi, Tom:

----- Original Message -----
From: Van Caenegem, Tom (Tom)<mailto:tom.van_caenegem@alcatel-lucent.com>
To: Qin Wu<mailto:sunseawq@huawei.com> ; avt@ietf.org<mailto:avt@ietf.org>
Sent: Tuesday, October 19, 2010 4:26 PM
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Qin,

Thank you for your reply.

Reading your responses,you seem to focus really on the SSM  case with an ar=
chitecture where the single DS (there can be only one DS per SSM) hosts the=
 sole feedback target, which also hosts the loss detector function. My inte=
rpretation is also that in your draft the loss detector sends the FIR or NA=
CK supression message to the DS, which then forwards/relays this message ov=
er the SSM to all receivers, right?

[Qin]: SSM case is just one example case in this draft. We have also discus=
sed Transport translator use case and MCU use case. This draft focus on how=
 the feedback suppression works in these different use cases.
For the SSM use case, we may consider multiple distribution sources placed =
between media source and the receiver. The loss detector is just one functi=
onality of Distribution source, or Retransmission Server or MCU, or Transpo=
rt Translator. It does not have to be put as part of feedback target. I pro=
pose to remove the stuff of "Loss detector" from this draft, instead, using=
 Distribution source with loss detection support. What do you think of it?

My main comment remains the same: you should at least incorporate the archi=
tecture of RAMS (where there can be several FT  colocated with the BRS, whi=
ch are disjoint from the DS). In this case the FT/BRS should be able to pro=
vide a suppression indication to all receivers that report to this FT (whic=
h is a subset of all receivers that connect to the SSM), which means one sh=
ould not make use of the (original) SSM for providing the suppression indic=
ation.  Your draft provides a possible solution for only a limited use case=
, and not the most interesting one, IMO.

[Qin]: I think RAMS case is just one special case of SSM use case. how Feed=
back suppression works for SSM use case therfore is quite similar to RAMS, =
the only difference is Distrbituion source in the SSM case is replaced with
BRS in the RAMS. As regarding another RAMS case where several FTs separated=
 from BRS,   we can address this case if there are strong interests.
But to be clear, this draft focus on how Feedback suppression message works=
 in different use case, not for one specific use case.

Further down some responses to your answers below.

Tom


________________________________
From: Qin Wu [mailto:sunseawq@huawei.com]
Sent: dinsdag 19 oktober 2010 5:04
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi, Tom:
Thank for your comments, please see my reply inline.

Regards!
-Qin
----- Original Message -----
From: Van Caenegem, Tom (Tom)<mailto:tom.van_caenegem@alcatel-lucent.com>
To: avt@ietf.org<mailto:avt@ietf.org> ; Qin Wu<mailto:sunseawq@huawei.com>
Sent: Tuesday, October 19, 2010 12:08 AM
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Qin,

I have some comments (look for TOM below) on the new draft version 4.
Thx for addressing them,
Tom


Section 2 Terminology

 Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.


 The function of the loss reporter may be collocated with
      or integrated into the same entity.

TOM (1): ..same entity as what? i guess you mean the DS?

[Qin]: Your are right.  In this draft, the loss detector is just functional=
ity of DS.  Furthermore we only consider the loss detector as part of feedb=
ack target within the distribution source.

Also this is just one example use case to explain how feedback suppression =
works in SSM use case.

TOM (2): I do not understand how you associate the port k to a loss detecto=
r.. what do you mean with that?

[Qin]:  Sorry for ambiguity of the description here. It actually means that=
 the loss reporter is part of feedback target and share the same address an=
d port.

section 3



(..)

In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.


TOM (3) : ...based on the two above definitions ...is an intermediate node =
hence equal to a "Loss detector"?
+ How does a Loss detector differ from an RTP receiver?
[Qin]: The loss detector is just one functionality of Distribution Source i=
n the SSM use case.
Maybe we should interpret it as "Loss detection functionality". How the pac=
ket loss is detected is out of scope of this document.
Different from dedicated RTP receiver, the intermediate node does not depen=
d on existing NACK message for Retransmission request for packet loss.
the intermediate node can create message by itself.

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.

TOM (4) : You probably do not want to have normative language in section 6.=
1, like "How the loss detector SHOULD (..)".

[Qin]:Sorry, I can't get it.

TOM: this is really a detail. If you want to keep the sentence, you should =
e.g. say: "How the loss detector can detect the packet loss...

 [Qin]Okay.

TOM (5) : It is pretty clear how a Loss detector will detect packet loss on=
 its upstream link, I guess.

[Qin]: As I said above,  Loss detector is just functionality of some Distri=
bution sources.  Also what it said here in section 6.1

is just one example use case.

TOM (6):  What is the rationale for a loss detector to send a Feedback supp=
ression message to the DS.. why NOT a RTCP NACK FB?

[Qin]: As I explained in the last IETF presentaion,  this cause NACK scemat=
ics mismatch. That's why many people in the last IETF meeeting suggest to c=
reate new RTCP message during presentation.

TOM: As your assumption is that the loss reporter is part of the FT that is=
 part of the DS, then there is even no need to specify how the loss detecto=
r

reports the packet loss.  In the more generic case, it makes more sense to =
me that the loss reporter just uses a NACK (or a FIR) if it detects packet =
loss, sent towards its FT or to the DS if this one hosts the FT that the lo=
ss detector reports to . It is then up to the FT or the DS to decide whethe=
r this message triggers a NACK or FIR suppression

indication to (s subset of) the receivers.  This suppression indication doe=
s not necessarly require a new message format. A NACK/FIR  that is sent to =
a RTP receiver can be defined as a feedback suppression indication. I do no=
t see a good use case why NACKs from  receivers  would be reflected back in=
 e.g. SSM use case by DS or FT  to all the connected receivers.

If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP rece=
iver, if this is for suppression indication, there is no ambiguity possibly=
.

 [Qin]: To me, NACK is uesed to request packet loss retransmission from the=
 server, NACK has no scemantics for feedback suppression. the intermediate =
node e.g., distribution

source will confus whether the NACK should be forwarded to the server or th=
e receiver.

unlike NACK, Feedback Suppression message defined in this draft is used by =
the intermediate node/middlebox to notify all the receiver that sending NAC=
K may be not necessary and harmful.

the intermediate node will know how to deal with the Feedback Suppression m=
essage and will not cause scemantics confusion like receiving NACK.

One important thing is Feedback suppression can be used without receiving A=
NY NACKs.

that is why many people think we should define new message althugh the new =
message has the similar format as NACK.

Therefore I think using NACK sent from some dedicated receiver is just one =
way for the distribution source to know packet loss happen. But not the onl=
y way.

Section 6.1.1

If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.


TOM (7) : how will the DS send only a copy of the RTCP FB report packet to =
the group impacted by packet loss. Is this done over the SSM?

[Qin]: Sorry, maybe I sould say
"
Section 6.1.1
If there are multiple Distrbitution Source with the loss detection support =
looking at the same RTP stream.....
"
Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback Suppres=
sion Report packet the group via SSM.

In your example you seem to focus only on architectures with a DS which als=
o hosts the FT and the loss reporter.  There should be sections covering ca=
ses where the DS is disjoint from the FT, with several FTs and also several=
 loss reporters disjoint from the FT and the DS.

[Qin] I am not sure we need to address all the possible SSM use cases. In t=
his draft, we just give some examples of SSM use case.

TOM (8) : what do you mean in above text with "RTCP FB report".. is that th=
e suppression message you define in this draft?

[Qin]: Yes,  RTCP Feedback Suppression Report.

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model" =
it is not clear what your architecture looks like. You talk about multiple =
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on t=
hat?

[Qin]:  The scenario we are talking about in  the section 6.1.2, is to  all=
ow multiple Distrbituion source between media source and the receiver. I wi=
ll look at the this case as one special SSM case. because  each of distrbit=
uion source resides at the upstream direction or downstream direction of ot=
her distributions souces. Does it make sense?

TOM: RFC 5760 only defines one DS for a given SSM. If you depart from that,=
 you probably need to address and define this architecture in a better way =
in your draft.

[Qin]: Okay.

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  typi=
cally an overlay of unicast is used... will the Feedback suppression messag=
e be sent over unicast as well? If so, what is gain compared to just having=
 each RTCP receiver providing each their NACK to the MCU, even in case of a=
 packet loss impacting all receivers?

[Qin]: First The MCU case will not cause NACK Implosion, instead, it may ca=
use FIR implosion.
           Second, using Feedback suppression message can restrict the FIR =
message sent from receivers behind MCU and liberate the media source from  =
FIR implosion.

TOM: it is not important whether this is about FIR or NACK. My point is tha=
t iso having each receiver sending a NACK/FIR, you will now have to send to=
 each receiver a NACK/FIR  suppression in the given scenario. There are no =
differences in terms of Bandwdith overhead or server load in the two cases,=
 so why bother to send suppression messages?

[Qin]: No, one effective use case is suppose packet loss can not repaired, =
the MCU can use feedback suppression message to tell the receiver not ask f=
or lost packet.

Tom (11) : coming back to my previous comment on SDP description in this dr=
aft : if this draft defines a RTCP feedback suppression message, why is the=
re need for a "announcement" in the SDP for the receivers, that indicates t=
hey may receive such feedback suppression messages?  If a receiver supports=
 the message, it will  simply act accordingly as described in the draft, no=
 need to announce that!..

[Qin]: Okay, I get your point. I would like to put this as open issue, disc=
uss it in the face to face IETF meeting.

Regards
Tom






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<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.6003" name=3DGENERATOR>
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<BODY bgColor=3D#cce8cf>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Qin,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>focusing on just SSM arch: I quote from your earli=
er=20
response:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010>In this draft,=
 the loss=20
detector is just functionality of DS.&nbsp; Furthermore we only consider th=
e=20
loss detector as part of feedback target within the distribution source. <S=
PAN=20
class=3D237051013-18102010><FONT face=3D"Times New Roman">Also this is just=
 one=20
example use case to explain how feedback suppression works in SSM use=20
case.</FONT></SPAN></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2>Agai=
n, your draft=20
does explain in detail how feedback suppression&nbsp;works for a SSM as=20
described here above, but does not address the RAMS case (which assumes a s=
ingle=20
DS, one or multiple disjoint FTs, where&nbsp;a FT is co-positioned with a B=
RS=20
(same IP address). </FONT></SPAN></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2>Just=
 consider a=20
DS with multiple disjoint&nbsp;FTs (as in RAMS, and also permitted by RFC 5=
760).=20
Typically each&nbsp;FT is connected to a different branch of the SSM tree.=
=20
</FONT></SPAN></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D764462011-19102010><SPAN=20
class=3D237051013-18102010><FONT face=3DArial color=3D#0000ff=20
size=3D2>Currently&nbsp;your draft imposes that the feedback suppression me=
ssage=20
is sent by the DS over the SSM. This is not a good solution as you either=20
restrict the solution to packet losses that are common to all SSM RTP recei=
vers=20
(covering packet loss between media source and DS ) or you will send suppre=
ssion=20
indications (and optionally forthcoming retransmission) to all receivers ev=
en if=20
some did not suffer from packet loss.&nbsp;Therefore&nbsp;the draft should=
=20
also&nbsp;define how suppression information is sent to (a subset of) the=20
receivers. It does not make sense to me that IETF would define FB suppressi=
on=20
for a architecture that does not take&nbsp;into account deployed SSM=20
architectures for IPTV.</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2>Even within the SSM architecture you =
are=20
focusing on, I am very much confused: you say there is one DS that hosts th=
e FT=20
and loss reporting function. What is then the role of the intermediate sour=
ces=20
downstream of the DS? I really would appreciate you illustrate with some mo=
re=20
architecture drawings.</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2>The other point I still am not convin=
ced of, is=20
&nbsp;the use of NACK versus dedicated suppression message.&nbsp;There shou=
ld be=20
more clarity in the different architectures you consider before we can make=
 any=20
statements on whether having a NACK or FIR&nbsp;&nbsp;that is&nbsp;used as=
=20
suppression message towards the SSM receivers, is confusing or not. I do no=
t=20
disagree that NACK as defined in RFC 4588 cannot be used as suppression mes=
sage.=20
But if we define it this way in a&nbsp;this "suppression" &nbsp;draft/RFC, =
and=20
no confusions can occur, I do not see the issue, especially as the syntax o=
f the=20
suppression message is completely the same as a NACK/FIR.=20
</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2>Regards</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2>Tom</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D764462011-19102010><SPAN class=3D237051013-18102010><FON=
T=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Qin Wu [mailto:sunseawq@huawei.co=
m]=20
<BR><B>Sent:</B> dinsdag 19 oktober 2010 12:50<BR><B>To:</B> Van Caenegem, =
Tom=20
(Tom); avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New Version Notification f=
or=20
draft-wu-avt-retransmission-supression-rtp-04<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Hi, Tom:</DIV>
<DIV><FONT face=3D&#23435;&#20307; size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LE=
FT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 9pt &#23435;&#20307;">----- Original Message ----- </=
DIV>
  <DIV style=3D"BACKGROUND: #e4e4e4; FONT: 9pt &#23435;&#20307;; font-color=
: black"><B>From:</B>=20
  <A title=3Dtom.van_caenegem@alcatel-lucent.com=20
  href=3D"mailto:tom.van_caenegem@alcatel-lucent.com">Van Caenegem, Tom (To=
m)</A>=20
  </DIV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><B>To:</B> <A title=3Dsunseawq@=
huawei.com=20
  href=3D"mailto:sunseawq@huawei.com">Qin Wu</A> ; <A title=3Davt@ietf.org=
=20
  href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><B>Sent:</B> Tuesday, October 1=
9, 2010 4:26 PM</DIV>
  <DIV style=3D"FONT: 9pt &#23435;&#20307;"><B>Subject:</B> RE: [AVT] New V=
ersion Notification=20
  for draft-wu-avt-retransmission-supression-rtp-04</DIV>
  <DIV><BR></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Hi Qin,</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Thank you for your reply.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Reading your responses,you seem to focus really =
on the=20
  SSM&nbsp; case with an architecture where the single DS (there can be onl=
y one=20
  DS per SSM) hosts the sole feedback target, which also hosts the loss det=
ector=20
  function. My interpretation is also that in your draft the loss detector =
sends=20
  the FIR or NACK supression message to the DS, which then forwards/relays=
=20
  this&nbsp;message over the SSM to all receivers, right?&nbsp;=20
  </FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  size=3D2>[Qin]: SSM case is just one example case in this draft. We have =
also=20
  discussed Transport translator use case and MCU use case. This draft focu=
s on=20
  how the feedback suppression works in these different use=20
  cases.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  size=3D2>For the SSM use case, we may consider multiple distribution sour=
ces=20
  placed between media source and the receiver. The loss detector is just o=
ne=20
  functionality of Distribution source, or Retransmission Server or MCU, or=
=20
  Transport Translator. It does not have to be put as part of feedback targ=
et. I=20
  propose to remove the stuff&nbsp;of "Loss detector" from this draft, inst=
ead,=20
  using Distribution source with loss detection support. What do you think =
of=20
  it?</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>My main comment remains the same: you should at =
least=20
  incorporate the architecture of RAMS (where&nbsp;there&nbsp;can be severa=
l=20
  FT&nbsp; colocated with the BRS, which are disjoint from the DS). In this=
 case=20
  the FT/BRS should be able to provide a suppression indication to all rece=
ivers=20
  that report to this FT (which is a subset of all receivers that connect t=
o the=20
  SSM), which means one should not make use of the (original) SSM for provi=
ding=20
  the suppression indication.&nbsp; Your draft provides a possible solution=
 for=20
  only a limited use case, and not the most interesting one,=20
  IMO.&nbsp;&nbsp;</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  size=3D2>[Qin]: I think RAMS case&nbsp;is just one special case of&nbsp;S=
SM use=20
  case.&nbsp;how Feedback suppression works for SSM use case therfore is qu=
ite=20
  similar to RAMS, the only difference is Distrbituion source in the SSM ca=
se is=20
  replaced with</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  size=3D2>BRS in the RAMS. As regarding&nbsp;another RAMS case where sever=
al FTs=20
  separated from BRS,&nbsp;&nbsp; </FONT></SPAN><SPAN=20
  class=3D289340207-19102010><FONT face=3DArial size=3D2>we can address thi=
s case if=20
  there&nbsp;are strong&nbsp;interests.&nbsp;</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  size=3D2>But to be clear, this draft focus on how Feedback suppression me=
ssage=20
  works in different use case, not for one specific use=20
case.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Further down&nbsp;some&nbsp;responses to your an=
swers=20
  below.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2>Tom</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D289340207-19102010><FONT face=
=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><FONT=20
  style=3D"BACKGROUND-COLOR: #ffffff" face=3DArial color=3D#0000ff size=3D2=
></FONT><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Qin Wu [mailto:sunseawq@huawei.=
com]=20
  <BR><B>Sent:</B> dinsdag 19 oktober 2010 5:04<BR><B>To:</B> Van Caenegem,=
 Tom=20
  (Tom); avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New Version Notification=
 for=20
  draft-wu-avt-retransmission-supression-rtp-04<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV>Hi, Tom:</DIV>
  <DIV>Thank for your comments, please see my reply inline.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards!</DIV>
  <DIV>-Qin</DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-=
LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
    <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman=
" size=3D3>----- Original=20
    Message ----- </FONT></DIV>
    <DIV style=3D"BACKGROUND: #e4e4e4; FONT: 9pt &#23435;&#20307;; font-col=
or: black"><FONT=20
    face=3D"Times New Roman"><FONT size=3D3><B>From:</B> </FONT></FONT><A=20
    title=3Dtom.van_caenegem@alcatel-lucent.com=20
    href=3D"mailto:tom.van_caenegem@alcatel-lucent.com"><FONT=20
    face=3D"Times New Roman" size=3D3>Van Caenegem, Tom (Tom)</FONT></A><FO=
NT=20
    face=3D"Times New Roman" size=3D3> </FONT></DIV>
    <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman=
"><FONT=20
    size=3D3><B>To:</B> </FONT></FONT><A title=3Davt@ietf.org=20
    href=3D"mailto:avt@ietf.org"><FONT face=3D"Times New Roman"=20
    size=3D3>avt@ietf.org</FONT></A><FONT face=3D"Times New Roman" size=3D3=
> ;=20
    </FONT><A title=3Dsunseawq@huawei.com href=3D"mailto:sunseawq@huawei.co=
m"><FONT=20
    face=3D"Times New Roman" size=3D3>Qin Wu</FONT></A><FONT face=3D"Times =
New Roman"=20
    size=3D3> </FONT></DIV>
    <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman=
"><FONT=20
    size=3D3><B>Sent:</B> Tuesday, October 19, 2010 12:08 AM</FONT></FONT><=
/DIV>
    <DIV style=3D"FONT: 9pt &#23435;&#20307;"><FONT face=3D"Times New Roman=
"><FONT=20
    size=3D3><B>Subject:</B> Re: [AVT] New Version Notification for=20
    draft-wu-avt-retransmission-supression-rtp-04</FONT></FONT></DIV>
    <DIV><FONT face=3D&#23435;&#20307; size=3D2></FONT><BR></DIV>
    <DIV><SPAN class=3D237051013-18102010>Hi Qin,</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>I have some comments (look for TO=
M=20
    below) on the new draft version 4.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Thx for addressing them,</SPAN></=
DIV>
    <DIV><SPAN class=3D237051013-18102010>Tom</SPAN></DIV>
    <DIV>&nbsp;</DIV>
    <DIV>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>Section 2 Terminology</SPAN></DIV=
>
    <DIV><PRE><FONT face=3D"Times New Roman"> Loss Detector:

      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
</FONT></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3D"Times New =
Roman"> The function of the loss reporter may be collocated with
      or integrated into the same entity.</FONT></SPAN></PRE><PRE><SPAN cla=
ss=3D237051013-18102010><FONT face=3D"Times New Roman">TOM (1): ..same enti=
ty as what? i guess you mean the DS?</FONT></SPAN></PRE><PRE><SPAN class=3D=
237051013-18102010><FONT face=3D"Times New Roman">[Qin]: Your are right.  I=
n this draft, the loss detector is just functionality of DS.  Furthermore w=
e only consider the loss detector as part of feedback target within the dis=
tribution source.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010>=
</SPAN><SPAN class=3D237051013-18102010><FONT face=3D"Times New Roman">Also=
 this is just one example use case to explain how feedback suppression work=
s in SSM use case.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010=
><FONT face=3D"Times New Roman">TOM (2): I do not understand how you associ=
ate the port k to a loss detector.. what do you mean with that?</FONT></SPA=
N></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3D"Times New Roman=
">[Qin]:  Sorry for ambiguity of the description here. It actually means th=
at the loss reporter is part of feedback target and share the same address =
and port.</FONT></SPAN></PRE><PRE><PRE><SPAN class=3D237051013-18102010><FO=
NT face=3D"Times New Roman">section 3</FONT></SPAN></PRE><PRE><FONT face=3D=
"Times New Roman">&nbsp;</FONT></PRE><PRE><SPAN class=3D237051013-18102010>=
<FONT face=3D"Times New Roman">(..)</FONT></SPAN></PRE><PRE><FONT face=3D"T=
imes New Roman">In order to detect packet loss before the receivers perceiv=
e it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.
</FONT></PRE></PRE></DIV>
    <DIV><!-- Converted from text/rtf format --><SPAN>TOM (3)=20
    </SPAN>:&nbsp;...<SPAN class=3D237051013-18102010>based on the two abov=
e=20
    definitions ...</SPAN>is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;h<SPA=
N=20
    class=3D237051013-18102010>e</SPAN>nce&nbsp;equal&nbsp;to&nbsp;a&nbsp;"=
Loss&nbsp;detector"?&nbsp;=20
    </DIV>
    <DIV><SPAN class=3D237051013-18102010></SPAN><SPAN=20
    class=3D237051013-18102010></SPAN>+<SPAN class=3D237051013-18102010> Ho=
w does a=20
    Loss detector differ from an RTP receiver?</SPAN></DIV><SPAN=20
    class=3D237051013-18102010></SPAN></BLOCKQUOTE>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-=
LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
    <DIV><SPAN class=3D237051013-18102010>[Qin]: The loss detector is just =
one=20
    functionality of Distribution Source in the SSM use case.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Maybe we should&nbsp;interpret it=
 as=20
    "Loss detection functionality". How the packet loss is detected is out =
of=20
    scope of this document.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Different from dedicated RTP rece=
iver,=20
    the intermediate node does not depend on existing NACK message for=20
    Retransmission request for packet loss.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>the intermediate node can create =
message=20
    by itself.</SPAN></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>Section 6.1</SPAN></DIV><PRE><FON=
T face=3D"Times New Roman">In order to avoid the forms of NACK implosion de=
scribed in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.</FONT></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><FON=
T face=3D"Times New Roman">T<SPAN class=3D237051013-18102010>OM (4) : You p=
robably do not want to have normative language in section 6.1, like "How th=
e loss detector SHOULD (..)". </SPAN></FONT></PRE><PRE><SPAN class=3D237051=
013-18102010><FONT face=3D"Times New Roman">[Qin]:Sorry, I can't get it.</F=
ONT><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-191=
02010>&nbsp;</SPAN></FONT></SPAN></PRE><PRE><SPAN class=3D237051013-1810201=
0><FONT face=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-19102=
010>TOM: this is really a detail. If you want to keep the sentence, you sho=
uld e.g. say: "How the loss detector can detect the packet loss...</SPAN></=
FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial =
color=3D#0000ff size=3D2><SPAN class=3D289340207-19102010></SPAN></FONT></S=
PAN>&nbsp;[Qin]Okay.</PRE><PRE><SPAN class=3D237051013-18102010><FONT face=
=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-19102010></SPAN><=
/FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><SPAN class=
=3D237051013-18102010></SPAN><FONT face=3D"Times New Roman">T<SPAN class=3D=
237051013-18102010>OM (5) : It is pretty clear how a Loss detector will det=
ect packet loss on its upstream link, I guess.</SPAN></FONT></PRE><PRE><SPA=
N class=3D237051013-18102010><FONT face=3D"Times New Roman">[Qin]: As I sai=
d above,  Loss detector is just functionality of some Distribution sources.=
  Also what it said here in section 6.1 </FONT></SPAN></PRE><PRE><SPAN clas=
s=3D237051013-18102010><FONT face=3D"Times New Roman">is just one example u=
se case.</FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010></SPAN><S=
PAN class=3D237051013-18102010><FONT face=3D"Times New Roman">TOM (6):  Wha=
t is the rationale for a loss detector to send a Feedback suppression messa=
ge to the DS.. why NOT a RTCP NACK FB?</FONT></SPAN></PRE><PRE><SPAN class=
=3D237051013-18102010><FONT face=3D"Times New Roman">[Qin]: As I explained =
in the last IETF presentaion,  this cause NACK scematics mismatch. That's w=
hy many people in the last IETF meeeting suggest to create new RTCP message=
 during presentation.</FONT><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN class=3D289340207-19102010>&nbsp;</SPAN></FONT></SPAN></PRE><PRE><SPAN c=
lass=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=
 class=3D289340207-19102010>TOM: As your assumption is that the loss report=
er is part of the FT that is part of the DS, then there is even no need to =
specify how the loss detector </SPAN></FONT></SPAN></PRE><PRE><SPAN class=
=3D237051013-18102010><FONT face=3DArial color=3D#0000ff size=3D2><SPAN cla=
ss=3D289340207-19102010>reports the packet loss.  In the more generic case,=
 it makes more sense to me that the loss reporter just uses a NACK (or a FI=
R) if it detects packet loss, sent towards its FT or to the DS if this one =
hosts the FT that the loss detector reports to . It is then up to the FT or=
 the DS to decide whether this message triggers a NACK or FIR suppression <=
/SPAN></FONT></SPAN></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=
=3DArial color=3D#0000ff size=3D2><SPAN class=3D289340207-19102010>indicati=
on to (s subset of) the receivers.  This suppression indication does not ne=
cessarly require a new message format. A NACK/FIR  that is sent to a RTP re=
ceiver can be defined as a feedback suppression indication. I do not see a =
good use case why NACKs from  receivers  would be reflected back in e.g. SS=
M use case by DS or FT  to all the connected receivers. </SPAN></FONT></SPA=
N></PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0=
000ff size=3D2><SPAN class=3D289340207-19102010>If we define that a NACK/FI=
R  for a SSM may only be sent TOWARDS a RTP receiver, if this is for suppre=
ssion indication, there is no ambiguity possibly.   </SPAN></FONT></SPAN></=
PRE><PRE><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f size=3D2><SPAN class=3D289340207-19102010></SPAN></FONT></SPAN>&nbsp;[Qin=
]: To me, NACK is uesed to request packet loss retransmission from the serv=
er, NACK has no scemantics for feedback suppression. the intermediate node =
e.g., distribution</PRE><PRE>source will confus whether the NACK should be =
forwarded to the server or the receiver.</PRE><PRE>unlike NACK, Feedback Su=
ppression message defined in this draft is used by the intermediate node/mi=
ddlebox to notify all the receiver that sending NACK may be not necessary a=
nd harmful.</PRE><PRE>the intermediate node will know how to deal with the =
Feedback Suppression message and will not cause scemantics confusion like r=
eceiving NACK.</PRE><PRE>One important thing is Feedback suppression can be=
 used without receiving ANY NACKs. </PRE><PRE>that is why many people think=
 we should define new message althugh the new message has the similar forma=
t as NACK.</PRE><PRE>Therefore I think using NACK sent from some dedicated =
receiver is just one way for the distribution source to know packet loss ha=
ppen. But not the only way.</PRE><PRE><SPAN class=3D237051013-18102010><FON=
T face=3D"Times New Roman">Section 6.1.1</FONT></SPAN></PRE><PRE><FONT face=
=3D"Times New Roman">If there are multiple Loss Detectors looking at the sa=
me RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.
</FONT></PRE>
    <DIV><SPAN class=3D237051013-18102010>TOM (7) : how will the DS send on=
ly a=20
    copy of the RTCP FB report packet to the group impacted by packet loss.=
 Is=20
    this done over the SSM? </SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]: Sorry, maybe I sould say=20
    </SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>"</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Section 6.1.1</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>If there are multiple Distrbituti=
on=20
    Source with the loss detection support looking at the same RTP=20
    stream.....</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>"</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Yes,&nbsp; in the SSM use case, D=
S send=20
    only a copy of the RTCP Feedback Suppression Report packet the group vi=
a=20
    SSM.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>In your example you seem to focus=
 only=20
    on architectures with a DS which also hosts the FT and the loss=20
    reporter.&nbsp; There should be sections covering cases where the DS is=
=20
    disjoint from the FT, with several FTs and also several loss reporters=
=20
    disjoint from the FT and the DS.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin] I am not sure we need to ad=
dress=20
    all the possible SSM use cases. In this draft, we just give&nbsp;some=20
    examples of SSM use case.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>TOM (8) : what do you mean in abo=
ve text=20
    with "RTCP FB report".. is that the suppression message you define in t=
his=20
    draft?</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]: Yes,&nbsp; RTCP Feedback=20
    Suppression Report. </SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>TOM (9) : In section&nbsp; "6.1.2=
.&nbsp;=20
    Distribution Source Feedback Summary Model" it is not clear what your=20
    architecture looks like. You talk about multiple distribution sources..=
=20
    whcih implies multiple SSMs.. Can you elaborate on that?</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]:&nbsp; The scenario we are =
talking=20
    about in&nbsp; the section 6.1.2, is&nbsp;to &nbsp;allow multiple=20
    Distrbituion source between media source and the receiver. I will look =
at=20
    the this case as one special SSM case. because&nbsp; each of distrbitui=
on=20
    source </SPAN><SPAN class=3D237051013-18102010>resides at the upstream=
=20
    direction or downstream direction of other distributions souces. Does i=
t=20
    make sense?<SPAN class=3D289340207-19102010><FONT face=3DArial color=3D=
#0000ff=20
    size=3D2>&nbsp;</FONT></SPAN></SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial color=3D#0000ff size=3D2>TOM: RFC 5760 only defines one DS=
 for a=20
    given SSM. If you depart from that, you probably need to address and=20
    define&nbsp;this architecture in a better way in your=20
    draft.</FONT>&nbsp;</SPAN></SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial size=3D2>[Qin]: Okay.</FONT></SPAN></SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>Tom (10) :&nbsp; In section&nbsp;=
=20
    6.3.&nbsp; Multipoint Control Unit (MCU) use case,&nbsp; typically an=20
    overlay of unicast is used... will the Feedback suppression message be =
sent=20
    over unicast as well? If so, what is gain compared to just having each =
RTCP=20
    receiver providing each their NACK to the MCU, even in case of a packet=
 loss=20
    impacting all receivers?</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]: First The MCU case will no=
t cause=20
    NACK Implosion, instead, it may cause FIR implosion.</SPAN></DIV>
    <DIV><SPAN=20
    class=3D237051013-18102010>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;=20
    Second, using Feedback suppression message can restrict the FIR message=
 sent=20
    from receivers&nbsp;behind MCU and liberate the media source from&nbsp;=
 FIR=20
    implosion.<SPAN class=3D289340207-19102010><FONT face=3DArial color=3D#=
0000ff=20
    size=3D2>&nbsp;</FONT></SPAN></SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010><SPAN class=3D289340207-19102010>=
<FONT=20
    face=3DArial color=3D#0000ff size=3D2>TOM: it is not important whether =
this is=20
    about FIR or NACK. My point is that iso having each receiver sending a=
=20
    NACK/FIR, you will now have to send to each receiver a NACK/FIR&nbsp;=20
    suppression in the given scenario. There are no differences in terms of=
=20
    Bandwdith overhead or server load in the two cases, so why bother to se=
nd=20
    suppression messages?</FONT></SPAN></SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT=20
    color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]: No, one effective use case=
 is=20
    suppose packet loss can not repaired, the MCU can use feedback suppress=
ion=20
    message to tell the receiver not ask for lost packet.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3D&#23435;&#20307;=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>Tom (11) : coming back to my prev=
ious=20
    comment on SDP description in this draft&nbsp;:&nbsp;if this draft defi=
nes a=20
    RTCP feedback suppression message, why is there need for a "announcemen=
t" in=20
    the SDP for the receivers, that indicates they may receive such feedbac=
k=20
    suppression messages?&nbsp; If a receiver supports the message, it=20
    will&nbsp; simply&nbsp;act accordingly as described in the draft, no ne=
ed to=20
    announce that!.. </SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>[Qin]: Okay, I get your point. I =
would=20
    like to put this as open issue, discuss it in the face to face IETF=20
    meeting.</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010>Regards</SPAN></DIV>
    <DIV><SPAN class=3D237051013-18102010>Tom</DIV>
    <DIV><BR></DIV></SPAN>
    <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D237051013-18102010></SPAN>&nbsp;</DIV>
    <DIV><SPAN=20
class=3D237051013-18102010></SPAN>&nbsp;</DIV></BLOCKQUOTE></BLOCKQUOTE></B=
ODY></HTML>

--_000_EC3FD58E75D43A4F8807FDE0749175460E497204FRMRSSXCHMBSB1d_--

From csp@csperkins.org  Tue Oct 19 06:57:15 2010
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T1/ Combining pkts
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On 19 Oct 2010, at 13:36, Magnus Westerlund wrote:
> Bob Briscoe skrev 2010-08-06 00:36:
>> Magnus, Ingemar, Colin, Piers, Ken,
>>=20
>> Below are my main technical concerns with=20
>> draft-ietf-avt-ecn-for-rtp-02 as promised.
>> I have used the line numbering here:
>> =
<http://tools.ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-av=
t-ecn-for-rtp-02.txt>
>>=20
>> T1/ Combining pkts                      <<<---THIS EMAIL
>> T2/ ECN-IP-RTCP
>> T3/ Anti-Cheating
>> T4/ Logging Not-ECT fall-back
>> T5/ Delivery %age ECT vs Not-ECT
>> T6/ Pkt Reordering around session start
>>=20
>>=20
>> T1/ Combining packets: Fragmentation and Reassembly
>> -------------------
>> "
>>=20
>>=20
>> 317           *  If the translator modifies the media stream to=20
>> combine or split
>> 318              RTP packets, but does not otherwise transcode the =
media, it
>> 319              must manage the ECN bits in a way analogous to that =
described
>> 320              in Section 5.3 of [RFC3168]:
>>=20
>>=20
>> "
>> Don't agree. The fragmentation and reassembly rules in RFC3168 were=20=

>> designed around TCP. They aren't applicable to RTP. See the argument=20=

>> about RFC3168 at the time it was in last call:
>> <http://www.ietf.org/mail-archive/web/tsvwg/current/msg00196.html>
>> This email refers to S.11 on fragmentation of this review of=20
>> draft-ietf-tsvwg-ecn-01 before it became RFC3168:
>> =
<http://www.bobbriscoe.net/projects/2020comms/ecn/protocol/ip/draft-ietf-t=
svwg-ecn-ip-00.txt>
>>=20
>> "
>>=20
>>=20
>> 322              identically to the original; if several ECN marked =
packets are
>> 323              combined into one, the outgoing packet must be =
either ECN-CE
>> 324              marked or dropped if any of the incoming packets are =
ECN-CE
>> 325              marked,
>> "
>> I suggest:
>> "...the outgoing packet SHOULD be ECN-CE marked with a probability=20
>> proportional to the ratio of ECN-CE bytes to total bytes in the=20
>> incoming packets being combined. However, for simplicity, the=20
>> outgoing packet MAY be ECN-CE marked if any of the incoming packets=20=

>> are ECN-CE marked. This latter option will inflate the amount of=20
>> congestion indicated, but it is at least safe.
>> "
>=20
> Ok, that does make sense. I am fine with that suggestion.

That makes sense, although it seems more aggressive than the rules in =
RFC 3168. What's the rationale for the change? I read =
draft-ietf-tsvwg-byte-pkt-congest-02, but that doesn't help my =
understanding.

>> 325                      and should be ECT marked if any of the=20
>> incoming packets
>> 326              are ECT marked.
>>=20
>>=20
>> "
>> I think this clause should shift to before the preceding one that=20
>> starts on line 323. It is a more logical order to first check whether=20=

>> all packets being combined are ECN-capable before combining CE =
markings.
>> (I take it that the logic here is that some RTP sources may not be=20
>> ECN capable, but if ECT packets are being sent by other sources, it=20=

>> implies all potential receivers are ECN-capable.)
>>=20
>=20
> Well, the issue here is that if one is in a probing phase and only =
send
> some fraction as ECT marked, and the rest without. If one combines =
them,
> we need to indicate which direction these markings should go. The
> current text proposes that if any are ECT, the outgoing packet is ECT.
> That way ensuring that any ECT are sent at all. In worst case all
> packets during the probing are ECT marked. Thus allowing a non-capable
> receiver to only detect this based on RTCP. But at least it is being
> detected during the probing phase rather than later which would be if
> one had a rule saying that if any of the packets being combined is
> Not-ECT then mark it as NOT-ECT would result in.


I'd suggest: "If the outgoing combined packet is not ECN-CE marked, then =
it MUST be ECT marked if any of the incoming packets were ECT marked."

>> "
>>=20
>> 1757       When rounding counter values in the translated RTCP =
packet, the
>> 1758       translator should try to ensure that they sum to the =
number of RTP
>> 1759       packets in the pre-translation sequence number space =
(numOrig).  The
>> 1760       translator should also try to ensure that no non-zero =
counter is
>> 1761       rounded to a zero value, since that will lose information =
that a
>> 1762       particular type of event has occurred.
>>=20
>>=20
>> "
>> An idea: Include a field in the RTP ECN feedback packet for the=20
>> ratio, R. Then, rather than the translator translating all the=20
>> numbers in the report, the host receiving the message can do the=20
>> translations, but it also has the original information.
>>=20
>=20
> Hmm, that is possibly a good idea. I guess we need at least 8 bits for
> such a ratio, maybe a couple more would be advantageous to get
> sufficient resolution. And if I understand you correct you want the
> ratio to be measured as:
> R=3D (ECN-CE marked bytes since last report from this SSRC) / (Total =
bytes received since last report from this SSRC)?


What's the benefit? This introduces complexity to avoid what should only =
be a small rounding error.=20

--=20
Colin Perkins
http://csperkins.org/




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Cc: draft-ietf-avt-ecn-for-rtp <draft-ietf-avt-ecn-for-rtp@tools.ietf.org>, Piers O'Hanlon <p.ohanlon@cs.ucl.ac.uk>, Bob Briscoe <rbriscoe@jungle.bt.co.uk>, avt <avt@ietf.org>
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T2/ ECN-IP-RTCP
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Hi,

I agree with Magnus, Piers and Ken: this extension doesn't seem worth =
the complexity at this time, and can easily be added later if we see a =
need.

I also note that RTCP traffic is not congestion controlled. Putting ECT =
marks on something that won't respond to ECN-CE marks seems undesirable.

Colin



On 19 Oct 2010, at 14:16, Magnus Westerlund wrote:
> Hi Bob,
>=20
> Thanks for these comments. I am sorry that I haven't responded =
earlier.
> I have finally gotten to the point of providing my view on the =
comments.
>=20
> So the benefit is that RTCP loss rates should be lower when congestion
> is high on the path the RTCP message takes. As RTCP packets aren't
> sequence nor reported on, there isn't a easy way of using them for
> congestion control directly. I am not saying impossible, but it will
> require significant amount of work and consume extra space in reports
> for fields related to RTCP.
>=20
> For the sender of the RTCP message to be able to influence the
> congestion levels on the path from itself to other end-point(s) it =
needs
> to have media flow over that path. If it has media flowing over that
> path, it can reduce its transmission rate and hope that it reduced the
> congestion level.
>=20
> If the RTCP sender doesn't have media flowing form it to the report
> consumer, which is true for unidirectional media, like streaming and =
for
> RTCP for SSM, or receiver only behavior in ASM. In those case the RTCP
> sending end-point can't affect the congestion level. In addition I =
think
> it is wrong to increase the delivery chance for RTCP when the RTCP
> sending end-point can't really reduce the rate of RTCP either, nor get
> report on that packets where ECN marked.
>=20
> =46rom my perspective the benefit is both questionable to utilize and
> minor compared to the additional complexities. Especially after we =
have
> improved your proposal, like getting the re-routing case working =
better.
> Because I think the obvious response to having had ECT working and
> suddenly detecting ECT failure is to turn of the ECT on RTCP when
> reporting this.
>=20
> If we in the end do want to enable this, we can write an extension or =
an
> update of the spec, where we override the SHALL.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> Bob Briscoe skrev 2010-08-18 18:29:
>> Piers,
>>=20
>> At 20:17 17/08/2010, Piers O'Hanlon wrote:
>>> Hi Bob,
>>>=20
>>> Thanks for the suggestion on ECN for RTCP - some comments in-line
>>> where appropriate and the rest here.
>>>=20
>>> One observation is that your mechanism generally relies on
>>> bi-directional flows to achieve the signalling
>>=20
>> Looking at your inline comments, I think you've misinterpreted my
>> attempt at careful wording. I said, "If B is a pure RTP receiver in a
>> unidirectional session, it will never get RTCP reports containing
>> counts of ECT and CE," which did not mean "it will never get RTCP
>> reports". It meant "Any RTCP reports it gets will never contain
>> counts of ECT or CE."
>>=20
>> Nonetheless, I agree my proposed variation to the protocol would only
>> ever turn on ECT for RTCP packets if there was some bidirectional RTP
>> traffic. That's the best I could do without requiring RTCP reports to
>> report on RTCP packets in the other direction. But that's better than =
nothing.
>>=20
>>> but when there are
>>> bidirectional RTP flows then the ECN status of the path would be
>>> provided already by the respective RTCP reports in each direction.
>>=20
>> Exactly. That's what I'm proposing to use in order to turn on ECT on
>> RTCP packets, which the current draft forbids.
>>=20
>>> It's hard to quantify the added benefit of knowing that the reverse
>>> path is getting congested and whether that is worth an extra
>>> 2xRTCP_reporting_period of loss....
>>=20
>> You only get this amount of loss if there has been a re-route from a
>> working ECN path to a non-compliant one. As stated elesewhere in my
>> review, I think the draft is rather paranoid about this possibility.
>> 1) Some people already question how much effort the IETF should put
>> into making protocols robust to non-compliant muddleboxes
>> 2) Once a compliant path has been found, would any reasonable person
>> really expect us to worry about losses while timing out after being
>> re-routed onto a another path that happens to be non-compliant? The
>> conditional probability of being re-routed onto a non-compliant path
>> having previously been on a compliant path must surely be low?
>>=20
>>> I guess the probability of each
>>> should hopefully be quite different but it is still a potentially
>>> nasty hit to take if an ugly muddlebox falls in the way.
>>=20
>> I agree all this is hard to quantify, but this double jeopardy is
>> like the risk of a pimple growing on a spot.
>>=20
>>> The other snag is with asymmetric paths, as the fact that the RTCP
>>> path is congested may not be indicative of the forward [RTP] path.
>>=20
>> The proposed protocol doesn't require or assume symmetric paths.
>>=20
>> I agree the main /motivation/ I gave for ECT on RTCP is not losing
>> RTCP reports /if there is/ correlation between congestion in both
>> directions. But the protocol doesn't need this correlation in order
>> to work. And there are other motivations anyway.
>>=20
>>> On balance I agree with Ken that it probably isn't worth the extra
>>> effort, though I can see the attraction.
>>=20
>> There is little extra effort for the gain you get in typical cases
>> (see response to Ken). We certainly should not so emphatically
>> prohibit ECT on RTCP packets with the ""SHALL NOT" in the draft,
>> given it can 'just work' in most deployment cases, with my proposed =
variation.
>>=20
>> I think we have different assumptions about which obscure
>> possibilities should be considered. For instance, I think a lot of
>> the complexity dealing with re-routes onto non-compliant paths is =
overkill.
>>=20
>> Personally I would leave these choices to implementers, which was the
>> spirit behind my proposal to modularise the protocol spec more, so
>> certain parts could be made optional.
>>=20
>>=20
>> Bob
>>=20
>>=20
>>> Thanks,
>>>=20
>>> Piers
>>>=20
>>> On 5 August 2010 23:39, Bob Briscoe <rbriscoe@jungle.bt.co.uk> =
wrote:
>>>> Magnus, Ingemar, Colin, Piers, Ken,
>>>>=20
>>>> Below are my main technical concerns with =
draft-ietf-avt-ecn-for-rtp-02 as
>>>> promised.
>>>> I have used the line numbering here:
>>>>=20
>>> =
<http://tools.ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-av=
t-ecn-for-rtp-02.txt>
>>>>=20
>>>> T1/ Combining pkts
>>>> T2/ ECN-IP-RTCP                         <<<---THIS EMAIL
>>>> T3/ Anti-Cheating
>>>> T4/ Logging Not-ECT fall-back
>>>> T5/ Delivery %age ECT vs Not-ECT
>>>> T6/ Pkt Reordering around session start
>>>>=20
>>>>=20
>>>> T2/ ECN in the IP header of RTCP reports.
>>>> -------------------
>>>> "
>>>>=20
>>>>=20
>>>> 510        RTCP traffic MUST NOT be ECT marked for the following =
reason.
>>>>=20
>>>>=20
>>>> "
>>>> "
>>>>=20
>>>>=20
>>>> 1047       The sender SHALL NOT include ECT marks on outgoing RTCP =
packets,
>>>> and
>>>> 1048       SHOULD NOT include ECT marks on any other outgoing =
control
>>>> messages
>>>> 1049       (e.g.  STUN [RFC5389] packets, DTLS [RFC4347] handshake =
packets,
>>>> or
>>>> 1050       ZRTP [I-D.zimmermann-avt-zrtp] control packets) that are
>>>> multiplexed
>>>> 1051       on the same UDP port.
>>>>=20
>>>>=20
>>>> "
>>>> If congestion rises on the reverse path it /could/ be a sign that =
it is
>>>> rising on the forward path. Ideally, we want it to be possible
>>> for receivers
>>>> to mark RTCP reports as ECT so they are not more likely to be
>>> lost just when
>>>> they are needed to report rising congestion levels. E.g. if the =
congestion
>>>> level on the reverse path is say, 0.5%, then 1 in 200 RTCP reports =
will be
>>>> lost.
>>>>=20
>>>> I promised to propose a way to have ECT on RTCP packets that
>>> would be robust
>>>> to middleboxes dropping ECT packets following a re-route. The
>>> solution makes
>>>> the following assumptions:
>>>> A1) that an ECN-Blocking Muddlebox drops any ECT packet, =
irrespective of
>>>> whether it is RTP or RTCP at the app-layer.
>>>> A2) that an RTP source knows how often to expect RTCP reports from
>>>> receivers, so it can count when RTCP reporting intervals have
>>> passed without
>>>> an RTCP report.
>>>>=20
>>>> Here's the proposal:* As per the I-D, the session starts without =
ECT on any
>>>> packets.
>>>> * First sender(s) introduce ECT on RTP packets through the probing =
methods
>>>> already described in the I-D.
>>>> * Consider the unicast case of a session betw A & B.
>>>> - Once A has received 3 successive RTCP reports containing counts =
of ECT
>>>> and CE, it can send its own RTCP reports as ECT.
>>>> - If A subsequently notices that 3 RTCP reporting periods have =
passed
>>>> without a report, or A receives no traffic at all for time [TBA], =
then A
>>>> reverts to sending RTCP reports as Not-ECT.
>>>> - B follows the same two rules as A.
>>>> - If B is a pure RTP receiver in a unidirectional session, it will =
never
>>>> get RTCP reports containing counts of ECT and CE, so it will
>>> never switch to
>>>> sending its own RTCP reports as ECT. This is because it has no way =
to test
>>>> if ECT packets are getting through.
>>>>=20
>>> A 'pure RTP receiver', which doesn't send RTCP isn't compliant with
>>> the base RTP spec (rfc3550) - an RTP receiver [only] should still =
send
>>> periodic RTCP Receiver Reports.  Though there are quite a few Voip
>>> phones do 'forget' to send RTCP.
>>>=20
>>>=20
>>>> - Then if a re-route sends B>A traffic through an ECN-Blocking =
Middlebox,
>>>> no B>A packets will get through. But after 3 RTCP reporting periods =
A tells
>>>> B in a Not-ECT RTCP report and B reverts to Not-ECT.
>>>> - If a re-route sends both B>A and A>B traffic through an =
ECN-Blocking
>>>> Middlebox, the above behaviour will also recover the session.
>>>>=20
>>>> * Multicast multi-source case, A -> B,C,D,E; B -> A,C,D,E; etc.
>>>> - Once A has decided it can safely turn on ECT in RTP packets it =
sends, it
>>>> watches for any RTCP reports that report reception of a positive =
Not-ECT
>>>> count but zero ECT or CE count on packets 'A' sent. Once it has =
received
>>>> none of these RTCP reports for three RTCP reporting rounds, 'A' can =
switch
>>>> on ECT for its own RTCP reports. 'A' determines 3 reporting rounds =
by
>>>> picking a couple of receivers and waiting until it receives 3 =
reports from
>>>> each.
>>>> - B,C,D,E follow the same rule.
>>>>=20
>>>> - A cannot test whether its own ECT RTCP reports continue to arrive =
at
>>>> their destinations, because the destinations do not know to expect =
traffic
>>>> from A. So if they are getting no traffic from A because of an
>>>> ECN-Blocking  Muddlebox, the transport will never know, although
>>>> the application or human may miss the contribution from A and reset
>>>> the transport to use Not-ECT on RTCP reports.
>>>>=20
>>> Again the receivers should be sending periodic Receiver Reports so A
>>> should be able to know the situation.
>>>=20
>>>> - With receiver-initiated multicast, A also cannot test for all =
sorts of
>>>> other blockages that might suddenly appear on paths to some =
destinations.
>>>> Therefore, once an ECN-capable session has been established, it =
would be
>>>> reasonable to take a leap-of-faith that re-routes are likely to =
continue to
>>>> find ECN-capable paths.   - And if the leap-of-faith is not =
warranted, the
>>>> session breaks until the application resets it to Not-ECT - but =
that's not
>>>> surprising if there are broken muddleboxes on some paths.
>>>> * Multicast single-source case, A -> P,Q,R,S etc.
>>>> - All follow the same rule as the multi-source case, with the same
>>>> limitations.
>>>> - As in the unicast and multi-source cases, if any hosts are pure =
RTP
>>>> receivers (P,Q,R,S), they won't ever get an RTCP report about RTP =
packets
>>>> they have sent, so they will never set ECT on RTCP reports.
>>>>=20
>>> There is the option for all parties exchange RTCP report when using
>>> SSM by employing RFC5760.
>>>=20
>>>> Comparison between ECT-RTCP and Not-ECT-RTCP:
>>>> * Session recovery time following a re-route through an =
ECN-Blocking
>>>> Middlebox:
>>>> - ECT-RTCP:           3 RTCP reporting intervals
>>>> - Not-ECT-RTCP:       1 RTCP reporting interval
>>>> * ECT-RTCP has more robust RTCP reporting during rising symmetric
>>> congestion
>>>> * Not-ECT-RTCP involves less implementation (unless using =
leap-of-faith,
>>>> when they are both the same).
>>>>=20
>>>> BTW, the assumption that an ECN-Muddlebox will drop any ECT
>>> packets, whether
>>>> RTP or RTCP, can be relaxed if we add reporting of RTCP packets =
from A>B in
>>>> reports from B>A. But it seems a reasonable assumption so reporting =
on RTCP
>>>> reports should not be necessary.
>>>>=20
>>>> Cheers
>>>>=20
>>>>=20
>>>> Bob
>>>>=20
>>>>=20
>>>>=20
>>>> ________________________________________________________________
>>>> Bob Briscoe,                                BT Innovate & Design
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>=20
>>=20
>> ________________________________________________________________
>> Bob Briscoe,                                BT Innovate & Design
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From csp@csperkins.org  Tue Oct 19 07:19:09 2010
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To: Bob Briscoe <rbriscoe@jungle.bt.co.uk>
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Cc: draft-ietf-avt-ecn-for-rtp@tools.ietf.org, avt@ietf.org
Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating
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The ECN nonce support is pretty modular. I'm happy to rip it out of this =
draft, since we can always resurrect it in a separate draft if there's a =
need later.

Colin


On 26 Aug 2010, at 00:36, Bob Briscoe wrote:
> Ken,
>=20
> We're violently agreeing I think, modulo just a little more optimism =
on my part about ConEx ;)
>=20
> Colin already said (in a Maastricht corridor discussion) that he would =
be happy to defer receiver misbehaviour stuff. But I don't think I had =
put the specific proposal to him I put in my mail - modularising the =
possibility of two alternative future approaches. I've cut all that text =
from the thread now so, Colin, feel free to respond to Ken's request for =
your view in the previous mail instead.
>=20
> One response inline, altho I think that part of the conversation is =
getting pretty obscure...
>=20
> At 21:39 25/08/2010, ken carlberg wrote:
>> Bob,
>>=20
>> FYI. its been a while since this email was sent, so I'm not snipping =
any of it, but my responses are few and scattered throughout.  And in =
this case, my opinions are not that strong, so I can probably be swayed =
by my co-authors if they feel differently.  That said...
>>=20
>> On Aug 5, 2010, at 6:41 PM, Bob Briscoe wrote:
>> > Magnus, Ingemar, Colin, Piers, Ken,
>=20
> [snip]
>=20
>> > T3c/ Misbehaving end-points
>> > -------------------
>> > "
>> > 1968                                                      It is far =
from certain
>> > 1969          that a receiver would be able to get a significant =
larger share of
>> > 1970          the resources.  That assumes a high enough level of =
aggregation
>> > 1971          that there are flows to acquire shares from.
>> > "
>> > Don't agree. It's completely certain that an unresponsive receiver =
(or sender) can get a significantly larger share of the resources, =
whether or not it uses ECN. Just try it. Send an unresponsive flow along =
with some responsive flows (e.g TCP) and the TCPs reduce their rates =
while the unresponsive flow gets whatever rate it chooses to send at.
>> > "
>>=20
>> well, it depends on context.  if I have an aggregate of flows that =
are only periodically and in a bursty manner triggering a  congestive =
state, as opposed to a sustained level of congestion condition (though =
not collapse), then there are no guarantees that all the other flows =
will be marked or even suffer packet loss.  And...
>=20
>=20
> [BB]: Going back to the para at issue in the I-D that I quoted: could =
I paraphrase it as, "If there are only two flows sharing a bottleneck, a =
cheating receiver cannot get more than twice as much as it would under =
equality, which isn't so bad."? If that's what it means, nah! Capacity =
sharing problems are surely better measured by how little others get, =
rather than how much more the cheater gets.
>=20
> Moving on to your periodic bursts scenario, I'm not quite sure I =
understand what you are getting at or how it's relevant to the para in =
the I-D.
>=20
> Whatever, surely fair shares aren't an issue during any period when a =
link isn't fully utilised. Becasue, if a link isn't fully utilised for =
some period, however brief, it is impossible to take more than your fair =
share over that period (for any definition of fairness) and it's =
impossible to cause others to get less than their fair share, because =
nothing you are doing stops others taking more than they are taking.
>=20
> Then,  during those bursts, it's only relevant that there would have =
marking or loss if someone else had taken a greater share than they were =
taking at that time. But while there's no marking or loss, it implies =
no-one else is wanting more than they are taking. So lack of marking or =
loss indicates there cannot have been any sharing problem.
>=20
> Or am I missing your point?
>=20
>=20
> Bob
>=20
>=20
>=20
> ________________________________________________________________
> Bob Briscoe,                                BT Innovate & Design=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From magnus.westerlund@ericsson.com  Tue Oct 19 07:44:34 2010
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 Other Tech Concerns
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Hi Bob,

See inline for my comments on the three issues.

Bob Briscoe skrev 2010-08-06 00:50:
> Magnus, Ingemar, Colin, Piers, Ken,
> 
> Below are my main technical concerns with 
> draft-ietf-avt-ecn-for-rtp-02 as promised.
> I have used the line numbering here:
> <http://tools.ietf.org/idnits?url=http://tools.ietf.org/id/draft-ietf-avt-ecn-for-rtp-02.txt>
> 
> T1/ Combining pkts
> T2/ ECN-IP-RTCP
> T3/ Anti-Cheating
> T4/ Logging Not-ECT fall-back                   <<<---THIS EMAIL
> T5/ Delivery %age ECT vs Not-ECT                <<<---THIS EMAIL
> T6/ Pkt Reordering around session start <<<---THIS EMAIL
> 
> T4/ Not just fall-back to Not-ECT, but logging or alarms too.
> -------------------
> "
> 783        loss of all packets with ECT or ECN-CE markings.  If the 
> path between
> 784        the sender and the receivers exhibits either of these behaviours one
> 785        needs to stop using ECN immediately to protect both the network and
> 786        the application.
> "
> "
> 1332       Detecting clearing of ECN field:
> 1337       Dropping of ECT packets:
> "
> As well as detecting ECN black holes and falling back to Not-ECT, the 
> source ought to log the IP address pair exhibiting this problem. This 
> could be used for later diagnostics, if someone ever tries to 
> systematically clear up this middlemess (e.g. in a corporate 
> environment, or anywhere that a network operator has access to these logs).
> The logs might also be used as a database for selective leaps-of-faith.
> 
> If an appropriate management alarm can be raised, it should be too.

Agreed, I have no issues including that in the spec.

> 
> T5/ Comparing Delivery percentage of ECT vs Not-ECT
> -------------------
> "
> 1337       Dropping of ECT packets: To determine if the packet drop ratio is
> 1338       different between not-ECT and ECT marked transmission requires a mix
> 1339       of transmitted traffic.  The sender should compare if the delivery
> 1340       percentage (delivered / transmitted) between ECT and not-ECT is
> 1341       significantly different.  Care must be taken if the number 
> of packets
> 1342       are low in either of the categories.
> "
> Eh? This surely won't work. Under normal circumstances (ie no 
> ECN-Blocking Muddleboxes) Not-ECT packets will be dropped more due to 
> congestion and ECT packets will hardly ever be dropped. So the sender 
> cannot detect if ECT packets are getting some extra drop due to 
> Muddleboxes, by comparing with Not-ECT which will have extra drop for 
> another valid reason (congestion).

Well, that depends on the relative frequency between the ECT and non-ECT
and the actual packet loss rates. But, yes I think you might have
spotted a short coming in our thinking. But I think this is not
completely wrong but is lacking the factor from CE marking, see below.

A very direct method of resolving this issue is to use explicit loss
vectors by using RTCP XR packet loss vectors. The ECN Nonce one doesn't
separate between ECN-CE and packet loss and although possible requires a
bit of going around to find the packet loss rate for ECT. What I can
understand the sender would need to keep its actual transmission
pattern. First count and eliminate the not-ect packets that are lost
from the NONCE vector. Then the remaining marked in the NONCE vector are
CE and packet loss of ECT packets. By removing the number of CE marked
ones from the same report interval based on the ECN report blocks you
would arrive at an approximate number of lost ECT packets. I say
approximate, because you still have the reordering issue that make the
sender uncertain on exactly which packets has been reported in the ECN
report between two RTCP packets.

But lets look at what level of statistics is needed to be able to draw
some conclusions. I agree that for certain operation points there will
be a clear difference between ECT and non-ECT traffic. But lets look at
some examples:

No congestion at all: NON-ECT drop rate should be 0. ECT on capable
transport drop rate 0, one path(s) that are incapable there will be a
drop rate >0.

Low levels of congestion: Non-ECT drop rate at a low level (0.5%). ECT
on capable transport drop rate 0, with CE marking at low level (0.5%),
ECT on incapable path will have a drop rate >0.5.

High level of congestion: NON-ECT drop rate at high level (10%). ECT on
capable transport will have some drop rate x, where 10 > x > 0, The CE
marking rate will be approx 10 - x. ECT on incapable path will see a
drop rate that is > 10.

Based on the above, I think you can determine when there drops on ECT
packets that are not related. This as you can take the CE marking rates
into account. NON-ECT droped packets - CE marked packets should
approximately equal the number of ECT marked drop packets. What I am
unclear on in this case is how big fudge factor you do need. But that
needs to be derived based on statics.

Comments on the above reasoning.


> 
> T6/ Pkt Reordering around session start-------------------
> [Just for the record - I've already sent this point off list.]
> 
> I will use the term "RTCP ECN Reports" for both the RTCP AVPF NACK 
> transport feedback format and the RTCP XR ECN summary report.
> "
> 1281    4.4.2.  Interpretation of ECN Summary information
> 
> 1306       The counters will be initiated to zero to provide value for the RTP
> 1307       stream sender from the very first report.  After the first 
> report the
> 1308       changes between the latest received and the previous one is
> 1309       determined by simply taking the values of the latest minus the
> 1310       previous one, taking field wrapping into account.  This 
> definition is
> 1311       also robust to packet losses, since if one report is missing, the
> 1312       reporting interval becomes longer, but is otherwise equally valid.
> "
> Both proposed RTCP ECN Reports insufficiently define the range of 
> packets they cover. They both define the range by the highest 
> sequence number and the number of packets covered by the report (the 
> sum of the various couters in the report, all counted since the start 
> of the receiver's session).
> 
> This approach makes report accuracy vulnerable to reordering around 
> the start of the session. For instance, suppose teh receiver receives 
> the following packets, and imagine the receiver starts its session at 
> sequence #126. Then  sends its first report after seq #135:
> 
> ...119,122,123,124,125,126,127,128,120,121,129,130,132,133,134,135,...
> 
> Highest seq no  = #135
> Number of losses        = 1 (#131)
> packets received        = 11 (which will be broken down into CE, ECT etc)
> (Total pkts in report   = 12)
> 
> The sender will calculate that this report refers to the 12 packets 
> starting at 135-12=#123 and ending at #135.
> 
> Then, the sender cannot accurately check whether the number of ECT & 
> CE (or Not-ECT) packets reported by the receiver matches the number 
> of ECT (or Not-ECT) it sent, because they are both talking about a 
> different set of packets. For instance, if the sender sets #123 and 
> #124 as ECT probes, these will be counted as ECT by the sender, but 
> the receiver puts them outside the range it is reporting on.
> 
> This error will be repeated each time the receiver sends a report, 
> because it is always cumulative from the start of the session, and 
> the problem is an ill-defined session start.
> 
> There are three possible solutions:
> a) Add text to mandate that the receiver considers the session start 
> to be the lowest sequence number after it initialises its counters. 
> In the example, the receiver would count the session start as #120 
> not #126, even though it initialised its counters at #126. When it 
> receives packet #120 it immediately rolls back the start seqno to 
> #120 and considers 121-125 as 5 losses. Ultimately, because it later 
> receives #121 it only reports #122-125 as 4 losses - even if these 
> packets did actually arrive before the counter was initialised.
> 
> b) The receiver does not include any packets with seqnos below the 
> seqno when it initialised its variables (unless there has been a wrap 
> of course). In the example, the receever would not include packets 
> #120 & #121 in its RTCP ECN Reports.
> 
> c) The RTCP ECN Report formats include an explicit start seqno. I 
> don't think this is necessary, given either a) or b) are easy to do.
> 
> Am I correct, or have I overlooked something?

Yes, you are correct that it is important that a receiver do keep track
of which packet is the first to count from and thus either doing A or B.
I think we should add a clarification on this issue. As long as you do
either of A or B the sender wouldn't be fooled about the set.

I think from a probing point of view doing B would actually be better.
If one take this as an example the packet sender may could come to the
conclusion that the ECT probes are lost due to incapability rather than
re-ordering. The potential down side is that a probe that has been
received may fall outside of the window and need an additional round.

There are an alternative to B that I think will work, but most likely
are more complex to implement. That is that the sender will move its
start number back as long as possible from the initially received number
as long as there is no hole.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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Date: Tue, 19 Oct 2010 16:45:50 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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Hi Tom,

I want to clarify your example since I am not sure what is special here. The
BRS sends separate unicast streams to each participant so there is reason
for suppression on these unicast streams. As for the primary multicast
stream from the DS it goes to all receivers. Now if the BRS identify packet
loss on the primary stream it should send suppression indication to all
members of the multicast group regardless to which FT they report.

Can you clarify

 

Roni Even

 

 

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Van
Caenegem, Tom (Tom)
Sent: Tuesday, October 19, 2010 10:27 AM
To: Qin Wu; avt@ietf.org
Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

 

Hi Qin,

 

Thank you for your reply.

 

Reading your responses,you seem to focus really on the SSM  case with an
architecture where the single DS (there can be only one DS per SSM) hosts
the sole feedback target, which also hosts the loss detector function. My
interpretation is also that in your draft the loss detector sends the FIR or
NACK supression message to the DS, which then forwards/relays this message
over the SSM to all receivers, right?  

 

My main comment remains the same: you should at least incorporate the
architecture of RAMS (where there can be several FT  colocated with the BRS,
which are disjoint from the DS). In this case the FT/BRS should be able to
provide a suppression indication to all receivers that report to this FT
(which is a subset of all receivers that connect to the SSM), which means
one should not make use of the (original) SSM for providing the suppression
indication.  Your draft provides a possible solution for only a limited use
case, and not the most interesting one, IMO.  

 

Further down some responses to your answers below.

 

Tom

 

 

  _____  

From: Qin Wu [mailto:sunseawq@huawei.com] 
Sent: dinsdag 19 oktober 2010 5:04
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

Hi, Tom:

Thank for your comments, please see my reply inline.

 

Regards!

-Qin

----- Original Message ----- 

From:  <mailto:tom.van_caenegem@alcatel-lucent.com> Van Caenegem, Tom (Tom) 

To:  <mailto:avt@ietf.org> avt@ietf.org ;  <mailto:sunseawq@huawei.com> Qin
Wu 

Sent: Tuesday, October 19, 2010 12:08 AM

Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

 

Hi Qin,

 

I have some comments (look for TOM below) on the new draft version 4.

Thx for addressing them,

Tom

 

 

Section 2 Terminology

 Loss Detector:
 
      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
 The function of the loss reporter may be collocated with
      or integrated into the same entity.
TOM (1): ..same entity as what? i guess you mean the DS?
[Qin]: Your are right.  In this draft, the loss detector is just
functionality of DS.  Furthermore we only consider the loss detector as part
of feedback target within the distribution source.
Also this is just one example use case to explain how feedback suppression
works in SSM use case.
TOM (2): I do not understand how you associate the port k to a loss
detector.. what do you mean with that?
[Qin]:  Sorry for ambiguity of the description here. It actually means that
the loss reporter is part of feedback target and share the same address and
port.
section 3
 
(..)
In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.

TOM (3) : ...based on the two above definitions ...is an intermediate node
hence equal to a "Loss detector"?  

+ How does a Loss detector differ from an RTP receiver?

[Qin]: The loss detector is just one functionality of Distribution Source in
the SSM use case.

Maybe we should interpret it as "Loss detection functionality". How the
packet loss is detected is out of scope of this document.

Different from dedicated RTP receiver, the intermediate node does not depend
on existing NACK message for Retransmission request for packet loss.

the intermediate node can create message by itself.

 

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.
TOM (4) : You probably do not want to have normative language in section
6.1, like "How the loss detector SHOULD (..)". 
[Qin]:Sorry, I can't get it. 
TOM: this is really a detail. If you want to keep the sentence, you should
e.g. say: "How the loss detector can detect the packet loss...
 
 
TOM (5) : It is pretty clear how a Loss detector will detect packet loss on
its upstream link, I guess.
[Qin]: As I said above,  Loss detector is just functionality of some
Distribution sources.  Also what it said here in section 6.1 
is just one example use case.
TOM (6):  What is the rationale for a loss detector to send a Feedback
suppression message to the DS.. why NOT a RTCP NACK FB?
[Qin]: As I explained in the last IETF presentaion,  this cause NACK
scematics mismatch. That's why many people in the last IETF meeeting suggest
to create new RTCP message during presentation. 
 
TOM: As your assumption is that the loss reporter is part of the FT that is
part of the DS, then there is even no need to specify how the loss detector 
reports the packet loss.  In the more generic case, it makes more sense to
me that the loss reporter just uses a NACK (or a FIR) if it detects packet
loss, sent towards its FT or to the DS if this one hosts the FT that the
loss detector reports to . It is then up to the FT or the DS to decide
whether this message triggers a NACK or FIR suppression 
indication to (s subset of) the receivers.  This suppression indication does
not necessarly require a new message format. A NACK/FIR  that is sent to a
RTP receiver can be defined as a feedback suppression indication. I do not
see a good use case why NACKs from  receivers  would be reflected back in
e.g. SSM use case by DS or FT  to all the connected receivers. 
If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP
receiver, if this is for suppression indication, there is no ambiguity
possibly.   
 
 
Section 6.1.1
If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.

TOM (7) : how will the DS send only a copy of the RTCP FB report packet to
the group impacted by packet loss. Is this done over the SSM? 

 

[Qin]: Sorry, maybe I sould say 

"

Section 6.1.1

If there are multiple Distrbitution Source with the loss detection support
looking at the same RTP stream.....

"

Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback
Suppression Report packet the group via SSM.

 

In your example you seem to focus only on architectures with a DS which also
hosts the FT and the loss reporter.  There should be sections covering cases
where the DS is disjoint from the FT, with several FTs and also several loss
reporters disjoint from the FT and the DS.

 

[Qin] I am not sure we need to address all the possible SSM use cases. In
this draft, we just give some examples of SSM use case.

 

TOM (8) : what do you mean in above text with "RTCP FB report".. is that the
suppression message you define in this draft?

 

[Qin]: Yes,  RTCP Feedback Suppression Report. 

 

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model"
it is not clear what your architecture looks like. You talk about multiple
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on
that?

 

[Qin]:  The scenario we are talking about in  the section 6.1.2, is to
allow multiple Distrbituion source between media source and the receiver. I
will look at the this case as one special SSM case. because  each of
distrbituion source resides at the upstream direction or downstream
direction of other distributions souces. Does it make sense? 

 

TOM: RFC 5760 only defines one DS for a given SSM. If you depart from that,
you probably need to address and define this architecture in a better way in
your draft. 

 

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,
typically an overlay of unicast is used... will the Feedback suppression
message be sent over unicast as well? If so, what is gain compared to just
having each RTCP receiver providing each their NACK to the MCU, even in case
of a packet loss impacting all receivers?

 

[Qin]: First The MCU case will not cause NACK Implosion, instead, it may
cause FIR implosion.

           Second, using Feedback suppression message can restrict the FIR
message sent from receivers behind MCU and liberate the media source from
FIR implosion. 

 

TOM: it is not important whether this is about FIR or NACK. My point is that
iso having each receiver sending a NACK/FIR, you will now have to send to
each receiver a NACK/FIR  suppression in the given scenario. There are no
differences in terms of Bandwdith overhead or server load in the two cases,
so why bother to send suppression messages?

 

 

Tom (11) : coming back to my previous comment on SDP description in this
draft : if this draft defines a RTCP feedback suppression message, why is
there need for a "announcement" in the SDP for the receivers, that indicates
they may receive such feedback suppression messages?  If a receiver supports
the message, it will  simply act accordingly as described in the draft, no
need to announce that!.. 

 

[Qin]: Okay, I get your point. I would like to put this as open issue,
discuss it in the face to face IETF meeting.

 

Regards

Tom

 

 

 

 

 


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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Tom,<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I want to clarify your example since I am not sure what is
special here. The BRS sends separate unicast streams to each participant so
there is reason for suppression on these unicast streams. As for the primary
multicast stream from the DS it goes to all receivers. Now if the BRS identify
packet loss on the primary stream it should send suppression indication to all
members of the multicast group regardless to which FT they report.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Can you clarify<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'>

<div>

<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>

<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Van
Caenegem, Tom (Tom)<br>
<b>Sent:</b> Tuesday, October 19, 2010 10:27 AM<br>
<b>To:</b> Qin Wu; avt@ietf.org<br>
<b>Subject:</b> Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></span></p>

</div>

</div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi Qin,</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Thank you for your reply.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Reading your responses,you seem to focus really on the SSM&nbsp;
case with an architecture where the single DS (there can be only one DS per
SSM) hosts the sole feedback target, which also hosts the loss detector
function. My interpretation is also that in your draft the loss detector sends
the FIR or NACK supression message to the DS, which then forwards/relays
this&nbsp;message over the SSM to all receivers, right?&nbsp; </span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>My main comment remains the same: you should at least incorporate
the architecture of RAMS (where&nbsp;there&nbsp;can be several FT&nbsp;
colocated with the BRS, which are disjoint from the DS). In this case the
FT/BRS should be able to provide a suppression indication to all receivers that
report to this FT (which is a subset of all receivers that connect to the SSM),
which means one should not make use of the (original) SSM for providing the
suppression indication.&nbsp; Your draft provides a possible solution for only
a limited use case, and not the most interesting one, IMO.&nbsp;&nbsp;</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Further down&nbsp;some&nbsp;responses to your answers below.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Tom</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<div class=MsoNormal align=center style='text-align:center'>

<hr size=2 width="100%" align=center>

</div>

<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Qin Wu [mailto:sunseawq@huawei.com] <br>
<b>Sent:</b> dinsdag 19 oktober 2010 5:04<br>
<b>To:</b> Van Caenegem, Tom (Tom); avt@ietf.org<br>
<b>Subject:</b> Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04</span><o:p></o:p></p>

<div>

<p class=MsoNormal>Hi, Tom:<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Thank for your comments, please see my reply inline.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Regards!<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>-Qin<o:p></o:p></p>

</div>

<blockquote style='border:none;border-left:solid black 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

<div>

<p class=MsoNormal>----- Original Message ----- <span style='font-size:9.0pt;
font-family:SimSun'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal style='background:#E4E4E4'><b>From:</b> <span
style='font-size:9.0pt;font-family:SimSun'><a
href="mailto:tom.van_caenegem@alcatel-lucent.com"
title="tom.van_caenegem@alcatel-lucent.com"><span style='font-size:12.0pt;
font-family:"Times New Roman","serif"'>Van Caenegem, Tom (Tom)</span></a></span>
<span style='font-size:9.0pt;font-family:SimSun'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>To:</b> <span style='font-size:9.0pt;font-family:SimSun'><a
href="mailto:avt@ietf.org" title="avt@ietf.org"><span style='font-size:12.0pt;
font-family:"Times New Roman","serif"'>avt@ietf.org</span></a></span> ; <span
style='font-size:9.0pt;font-family:SimSun'><a href="mailto:sunseawq@huawei.com"
title="sunseawq@huawei.com"><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>Qin
Wu</span></a></span> <span style='font-size:9.0pt;font-family:SimSun'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>Sent:</b> Tuesday, October 19, 2010 12:08 AM<span
style='font-size:9.0pt;font-family:SimSun'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>Subject:</b> Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04<span style='font-size:9.0pt;
font-family:SimSun'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=MsoNormal>Hi Qin,<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>I have some comments (look for TOM below) on the new draft
version 4.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Thx for addressing them,<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 2 Terminology<o:p></o:p></p>

</div>

<div><pre><span style='font-family:"Times New Roman","serif"'>&nbsp;Loss Detector:<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'><o:p>&nbsp;</o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector is one logical function which is used to detect<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the packet loss at the RTP layer and report it to the distribution<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; source.&nbsp; The function of the loss reporter may be collocated with<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into the same entity.&nbsp; In this case, for a session<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; defined as having a distribution source A, on ports n for the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channel and k for the RTCP channel, the Loss Detector are<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; identified by an IP address of distribution source A on port k.<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector MAY also be implemented in one or more entities<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; different from the distribution source.<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'> The function of the loss reporter may be collocated with<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into the same entity.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (1): ..same entity as what? i guess you mean the DS?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: Your are right.&nbsp; In this draft, the loss detector is just functionality of DS.&nbsp; Furthermore we only consider the loss detector as part of feedback target within the distribution source.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>Also this is just one example use case to explain how feedback suppression works in SSM use case.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (2): I do not understand how you associate the port k to a loss detector.. what do you mean with that?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]:&nbsp; Sorry for ambiguity of the description here. It actually means that the loss reporter is part of feedback target and share the same address and port.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>section 3</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>(..)</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>In order to detect packet loss before the receivers perceive it, one<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; or more intermediate nodes are placed between the media source and<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receiver (e.g., Distribution server, MCU, RTP translator).&nbsp; These<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; intermediaries monitor for packet loss upstream of themselves by<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; checking RTP sequence numbers, just as receivers do.&nbsp; Upon detecting<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; (or suspecting) an upstream loss, the intermediary may send Feedback<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; Suppression message towards the receivers as defined in this<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; specification.<o:p></o:p></span></pre></div>

<div>

<p class=MsoNormal>TOM (3) :&nbsp;...based on the two above definitions
...is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;hence&nbsp;equal&nbsp;to&nbsp;a&nbsp;&quot;Loss&nbsp;detector&quot;?&nbsp;
<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>+ How does a Loss detector differ from an RTP receiver?<o:p></o:p></p>

</div>

</blockquote>

<blockquote style='border:none;border-left:solid black 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

<div>

<p class=MsoNormal>[Qin]: The loss detector is just one functionality of Distribution
Source in the SSM use case.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Maybe we should&nbsp;interpret it as &quot;Loss detection
functionality&quot;. How the packet loss is detected is out of scope of this
document.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Different from dedicated RTP receiver, the intermediate node
does not depend on existing NACK message for Retransmission request for packet
loss.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>the intermediate node can create message by itself.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 6.1<o:p></o:p></p>

</div>

<pre><span style='font-family:"Times New Roman","serif"'>In order to avoid the forms of NACK implosion described in section 1,<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; the Loss Detector is introduced.&nbsp; The Loss Detector detects and<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; reports packet loss, on both the upstream link and the downstream<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; aggregate link.&nbsp; How the loss detector SHOULD detect the packet loss<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; is beyond of scope of this document.&nbsp; When upstream link or<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; downstream aggregate link packet loss occurs, the Loss Detector MAY<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; inform distribution source of the detected packet loss using Feedback<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; Suppression messages.&nbsp; In response, the distribution source either<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; forwards packet loss suppression report received from Loss Detector<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; or creates a Feedback Suppression report and sends it to all the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receivers, over the multicast channel.&nbsp; This loss suppression report<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; tells the receivers that the lost packet will either be forthcoming<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; from distribution source, or it irretrievably lost such that there is<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; nothing to be gained by the receiver sending a NACK to the media<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; source.&nbsp; The distribution source then can (optionally) ask for the<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; lost packets from the media source on behalf of all the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receivers.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (4) : You probably do not want to have normative language in section 6.1, like &quot;How the loss detector SHOULD (..)&quot;. </span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]:Sorry, I can't get it.</span><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>TOM: this is really a detail. If you want to keep the sentence, you should e.g. say: &quot;How the loss detector can detect the packet loss...</span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (5) : It is pretty clear how a Loss detector will detect packet loss on its upstream link, I guess.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: As I said above,&nbsp; Loss detector is just functionality of some Distribution sources.&nbsp; Also what it said here in section 6.1 </span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>is just one example use case.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (6):&nbsp; What is the rationale for a loss detector to send a Feedback suppression message to the DS.. why NOT a RTCP NACK FB?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: As I explained in the last IETF presentaion,&nbsp; this cause NACK scematics mismatch. That's why many people in the last IETF meeeting suggest to create new RTCP message during presentation.</span><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>TOM: As your assumption is that the loss reporter is part of the FT that is part of the DS, then there is even no need to specify how the loss detector </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>reports the packet loss.&nbsp; In the more generic case, it makes more sense to me that the loss reporter just uses a NACK (or a FIR) if it detects packet loss, sent towards its FT or to the DS if this one hosts the FT that the loss detector reports to . It is then up to the FT or the DS to decide whether this message triggers a NACK or FIR suppression </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>indication to (s subset of) the receivers.&nbsp; This suppression indication does not necessarly require a new message format. A NACK/FIR&nbsp; that is sent to a RTP receiver can be defined as a feedback suppression indication. I do not see a good use case why NACKs from&nbsp; receivers&nbsp; would be reflected back in e.g. SSM use case by DS or FT&nbsp; to all the connected receivers. </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>If we define that a NACK/FIR&nbsp; for a SSM may only be sent TOWARDS a RTP receiver, if this is for suppression indication, there is no ambiguity possibly.&nbsp;&nbsp; </span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>Section 6.1.1</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>If there are multiple Loss Detectors looking at the same RTP stream,<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; then the loss may be identified by more than one and those detecting<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; the loss will all send requests for the same packet loss.&nbsp; In this<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; case, the distribution source MUST filter the duplicated packet loss<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; request out and only forward one copy of the RTCP Feedback report<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; packet from the first Loss Detector to the group impacted by packet<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; loss.<o:p></o:p></span></pre>

<div>

<p class=MsoNormal>TOM (7) : how will the DS send only a copy of the RTCP FB
report packet to the group impacted by packet loss. Is this done over the SSM? <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Sorry, maybe I sould say <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&quot;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 6.1.1<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>If there are multiple Distrbitution Source with the loss
detection support looking at the same RTP stream.....<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&quot;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Yes,&nbsp; in the SSM use case, DS send only a copy of the
RTCP Feedback Suppression Report packet the group via SSM.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>In your example you seem to focus only on architectures with
a DS which also hosts the FT and the loss reporter.&nbsp; There should be
sections covering cases where the DS is disjoint from the FT, with several FTs
and also several loss reporters disjoint from the FT and the DS.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin] I am not sure we need to address all the possible SSM
use cases. In this draft, we just give&nbsp;some examples of SSM use case.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>TOM (8) : what do you mean in above text with &quot;RTCP FB
report&quot;.. is that the suppression message you define in this draft?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Yes,&nbsp; RTCP Feedback Suppression Report. <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>TOM (9) : In section&nbsp; &quot;6.1.2.&nbsp; Distribution
Source Feedback Summary Model&quot; it is not clear what your architecture
looks like. You talk about multiple distribution sources.. whcih implies
multiple SSMs.. Can you elaborate on that?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]:&nbsp; The scenario we are talking about in&nbsp; the
section 6.1.2, is&nbsp;to &nbsp;allow multiple Distrbituion source between
media source and the receiver. I will look at the this case as one special SSM
case. because&nbsp; each of distrbituion source resides at the upstream
direction or downstream direction of other distributions souces. Does it make
sense?<span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>TOM: RFC 5760 only defines one DS for a given SSM. If you depart
from that, you probably need to address and define&nbsp;this architecture in a
better way in your draft.</span>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom (10) :&nbsp; In section&nbsp; 6.3.&nbsp; Multipoint
Control Unit (MCU) use case,&nbsp; typically an overlay of unicast is used...
will the Feedback suppression message be sent over unicast as well? If so, what
is gain compared to just having each RTCP receiver providing each their NACK to
the MCU, even in case of a packet loss impacting all receivers?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: First The MCU case will not cause NACK Implosion,
instead, it may cause FIR implosion.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Second, using Feedback suppression message can restrict the FIR message sent
from receivers&nbsp;behind MCU and liberate the media source from&nbsp; FIR
implosion.<span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>TOM: it is not important whether this is about FIR or NACK. My
point is that iso having each receiver sending a NACK/FIR, you will now have to
send to each receiver a NACK/FIR&nbsp; suppression in the given scenario. There
are no differences in terms of Bandwdith overhead or server load in the two
cases, so why bother to send suppression messages?</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom (11) : coming back to my previous comment on SDP
description in this draft&nbsp;:&nbsp;if this draft defines a RTCP feedback
suppression message, why is there need for a &quot;announcement&quot; in the
SDP for the receivers, that indicates they may receive such feedback
suppression messages?&nbsp; If a receiver supports the message, it will&nbsp;
simply&nbsp;act accordingly as described in the draft, no need to announce
that!.. <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Okay, I get your point. I would like to put this as
open issue, discuss it in the face to face IETF meeting.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Regards<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

</blockquote>

</div>

</div>

</body>

</html>

--Boundary_(ID_AyywcT0J2eS5iDLucO/JEw)--

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Subject: [AVT] RFC 5993 on RTP Payload Format for Global System for Mobile Communications Half Rate (GSM-HR)
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5993

        Title:      RTP Payload Format for Global 
                    System for Mobile Communications Half Rate 
                    (GSM-HR) 
        Author:     X. Duan, S. Wang,
                    M. Westerlund, K. Hellwig,
                    I. Johansson
        Status:     Standards Track
        Stream:     IETF
        Date:       October 2010
        Mailbox:    duanxiaodong@chinamobile.com, 
                    wangshuaiyu@chinamobile.com, 
                    magnus.westerlund@ericsson.com,  
                    karl.hellwig@ericsson.com, 
                    ingemar.s.johansson@ericsson.com
        Pages:      18
        Characters: 37824
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-avt-rtp-gsm-hr-03.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5993.txt

This document specifies the payload format for packetization of
Global System for Mobile Communications Half Rate (GSM-HR) speech
codec data into the Real-time Transport Protocol (RTP).  The payload
format supports transmission of multiple frames per payload and
packet loss robustness methods using redundancy.  [STANDARDS-TRACK]

This document is a product of the Audio/Video Transport Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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From tom.van_caenegem@alcatel-lucent.com  Tue Oct 19 08:29:33 2010
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From: "Van Caenegem, Tom (Tom)" <tom.van_caenegem@alcatel-lucent.com>
To: Roni Even <Even.roni@huawei.com>, "'Qin Wu'" <sunseawq@huawei.com>, "avt@ietf.org" <avt@ietf.org>
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Thread-Topic: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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--_000_EC3FD58E75D43A4F8807FDE0749175460E497311FRMRSSXCHMBSB1d_
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Hi Roni,

In the typical RAMS architecture, the BRS will receive the SSM stream from =
the DS. If there are several BRS-es, there may be just one BRS that detects=
 packet loss on its upstream link, but the others will perhaps not, as the =
packet loss took place on SSM tree branch that does not impact the other BR=
Ses. In that case those RTP receivers associated to the other BRS-es in gen=
eral will not be impacted and do not need to receive the suppression indica=
tion, while the one BRS that did sense the packet loss should probably best=
 send a suppression indication (although there is a chance that only this B=
RS is impacted by packet loss, and not a single RTP receiver in the field r=
eporting to this BRS/FT... this all depends on the considered topologies).

As a BRS is 1-to-1 associated with a FT, this calls for consideration of ar=
chitectures with (multiple) disjoint FT (and associated BRS in RAMS applica=
tions)  in the suppression draft. Such disjoint FT/BRS in general cannot ac=
t as so-called "intermediate nodes" or "translators" because they are not o=
n the direct SSM path between the DS and the RTP receivers.

Another aspect I want to highlight is that feedback suppression is closely =
aligned with retransmission. In the case described above, the one BRS that =
detected packet loss on its upstream link, may try to recover the missing p=
acket (outside the scope of the draft) and retransmit over multicast to the=
 receivers that report to the FT. This retransmission towards the receivers=
 is also outside the current scope of the current draft, but the two are so=
mehow linked.  In DVB IPTV, we introduced dedicated SSMs sourced by the BRS=
 for sending NACK suppression and the (optional) retransmission. In the con=
sidered scenario, having a DS sending a missing packet -reported by a FT do=
wnstream in the SSM tree- over the original SSM is not a good approach IMO,=
 as it results in general in unnecessary retransmissions for most receivers=
.

Hope this clarifies.

Tom

________________________________
From: Roni Even [mailto:Even.roni@huawei.com]
Sent: dinsdag 19 oktober 2010 16:46
To: Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org
Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Tom,
I want to clarify your example since I am not sure what is special here. Th=
e BRS sends separate unicast streams to each participant so there is reason=
 for suppression on these unicast streams. As for the primary multicast str=
eam from the DS it goes to all receivers. Now if the BRS identify packet lo=
ss on the primary stream it should send suppression indication to all membe=
rs of the multicast group regardless to which FT they report.
Can you clarify

Roni Even



From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Van C=
aenegem, Tom (Tom)
Sent: Tuesday, October 19, 2010 10:27 AM
To: Qin Wu; avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Qin,

Thank you for your reply.

Reading your responses,you seem to focus really on the SSM  case with an ar=
chitecture where the single DS (there can be only one DS per SSM) hosts the=
 sole feedback target, which also hosts the loss detector function. My inte=
rpretation is also that in your draft the loss detector sends the FIR or NA=
CK supression message to the DS, which then forwards/relays this message ov=
er the SSM to all receivers, right?

My main comment remains the same: you should at least incorporate the archi=
tecture of RAMS (where there can be several FT  colocated with the BRS, whi=
ch are disjoint from the DS). In this case the FT/BRS should be able to pro=
vide a suppression indication to all receivers that report to this FT (whic=
h is a subset of all receivers that connect to the SSM), which means one sh=
ould not make use of the (original) SSM for providing the suppression indic=
ation.  Your draft provides a possible solution for only a limited use case=
, and not the most interesting one, IMO.

Further down some responses to your answers below.

Tom


________________________________
From: Qin Wu [mailto:sunseawq@huawei.com]
Sent: dinsdag 19 oktober 2010 5:04
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04
Hi, Tom:
Thank for your comments, please see my reply inline.

Regards!
-Qin
----- Original Message -----
From: Van Caenegem, Tom (Tom)<mailto:tom.van_caenegem@alcatel-lucent.com>
To: avt@ietf.org<mailto:avt@ietf.org> ; Qin Wu<mailto:sunseawq@huawei.com>
Sent: Tuesday, October 19, 2010 12:08 AM
Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission=
-supression-rtp-04

Hi Qin,

I have some comments (look for TOM below) on the new draft version 4.
Thx for addressing them,
Tom


Section 2 Terminology

 Loss Detector:



      The Loss Detector is one logical function which is used to detect

      the packet loss at the RTP layer and report it to the distribution

      source.  The function of the loss reporter may be collocated with

      or integrated into the same entity.  In this case, for a session

      defined as having a distribution source A, on ports n for the RTP

      channel and k for the RTCP channel, the Loss Detector are

      identified by an IP address of distribution source A on port k.

      The Loss Detector MAY also be implemented in one or more entities

      different from the distribution source.

 The function of the loss reporter may be collocated with

      or integrated into the same entity.

TOM (1): ..same entity as what? i guess you mean the DS?

[Qin]: Your are right.  In this draft, the loss detector is just functional=
ity of DS.  Furthermore we only consider the loss detector as part of feedb=
ack target within the distribution source.

Also this is just one example use case to explain how feedback suppression =
works in SSM use case.

TOM (2): I do not understand how you associate the port k to a loss detecto=
r.. what do you mean with that?

[Qin]:  Sorry for ambiguity of the description here. It actually means that=
 the loss reporter is part of feedback target and share the same address an=
d port.

section 3



(..)

In order to detect packet loss before the receivers perceive it, one

   or more intermediate nodes are placed between the media source and

   receiver (e.g., Distribution server, MCU, RTP translator).  These

   intermediaries monitor for packet loss upstream of themselves by

   checking RTP sequence numbers, just as receivers do.  Upon detecting

   (or suspecting) an upstream loss, the intermediary may send Feedback

   Suppression message towards the receivers as defined in this

   specification.
TOM (3) : ...based on the two above definitions ...is an intermediate node =
hence equal to a "Loss detector"?
+ How does a Loss detector differ from an RTP receiver?
[Qin]: The loss detector is just one functionality of Distribution Source i=
n the SSM use case.
Maybe we should interpret it as "Loss detection functionality". How the pac=
ket loss is detected is out of scope of this document.
Different from dedicated RTP receiver, the intermediate node does not depen=
d on existing NACK message for Retransmission request for packet loss.
the intermediate node can create message by itself.

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,

   the Loss Detector is introduced.  The Loss Detector detects and

   reports packet loss, on both the upstream link and the downstream

   aggregate link.  How the loss detector SHOULD detect the packet loss

   is beyond of scope of this document.  When upstream link or

   downstream aggregate link packet loss occurs, the Loss Detector MAY

   inform distribution source of the detected packet loss using Feedback

   Suppression messages.  In response, the distribution source either

   forwards packet loss suppression report received from Loss Detector

   or creates a Feedback Suppression report and sends it to all the RTP

   receivers, over the multicast channel.  This loss suppression report

   tells the receivers that the lost packet will either be forthcoming

   from distribution source, or it irretrievably lost such that there is

   nothing to be gained by the receiver sending a NACK to the media

   source.  The distribution source then can (optionally) ask for the

   lost packets from the media source on behalf of all the RTP

   receivers.

TOM (4) : You probably do not want to have normative language in section 6.=
1, like "How the loss detector SHOULD (..)".

[Qin]:Sorry, I can't get it.

TOM: this is really a detail. If you want to keep the sentence, you should =
e.g. say: "How the loss detector can detect the packet loss...





TOM (5) : It is pretty clear how a Loss detector will detect packet loss on=
 its upstream link, I guess.

[Qin]: As I said above,  Loss detector is just functionality of some Distri=
bution sources.  Also what it said here in section 6.1

is just one example use case.

TOM (6):  What is the rationale for a loss detector to send a Feedback supp=
ression message to the DS.. why NOT a RTCP NACK FB?

[Qin]: As I explained in the last IETF presentaion,  this cause NACK scemat=
ics mismatch. That's why many people in the last IETF meeeting suggest to c=
reate new RTCP message during presentation.



TOM: As your assumption is that the loss reporter is part of the FT that is=
 part of the DS, then there is even no need to specify how the loss detecto=
r

reports the packet loss.  In the more generic case, it makes more sense to =
me that the loss reporter just uses a NACK (or a FIR) if it detects packet =
loss, sent towards its FT or to the DS if this one hosts the FT that the lo=
ss detector reports to . It is then up to the FT or the DS to decide whethe=
r this message triggers a NACK or FIR suppression

indication to (s subset of) the receivers.  This suppression indication doe=
s not necessarly require a new message format. A NACK/FIR  that is sent to =
a RTP receiver can be defined as a feedback suppression indication. I do no=
t see a good use case why NACKs from  receivers  would be reflected back in=
 e.g. SSM use case by DS or FT  to all the connected receivers.

If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP rece=
iver, if this is for suppression indication, there is no ambiguity possibly=
.





Section 6.1.1

If there are multiple Loss Detectors looking at the same RTP stream,

   then the loss may be identified by more than one and those detecting

   the loss will all send requests for the same packet loss.  In this

   case, the distribution source MUST filter the duplicated packet loss

   request out and only forward one copy of the RTCP Feedback report

   packet from the first Loss Detector to the group impacted by packet

   loss.
TOM (7) : how will the DS send only a copy of the RTCP FB report packet to =
the group impacted by packet loss. Is this done over the SSM?

[Qin]: Sorry, maybe I sould say
"
Section 6.1.1
If there are multiple Distrbitution Source with the loss detection support =
looking at the same RTP stream.....
"
Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback Suppres=
sion Report packet the group via SSM.

In your example you seem to focus only on architectures with a DS which als=
o hosts the FT and the loss reporter.  There should be sections covering ca=
ses where the DS is disjoint from the FT, with several FTs and also several=
 loss reporters disjoint from the FT and the DS.

[Qin] I am not sure we need to address all the possible SSM use cases. In t=
his draft, we just give some examples of SSM use case.

TOM (8) : what do you mean in above text with "RTCP FB report".. is that th=
e suppression message you define in this draft?

[Qin]: Yes,  RTCP Feedback Suppression Report.

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model" =
it is not clear what your architecture looks like. You talk about multiple =
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on t=
hat?

[Qin]:  The scenario we are talking about in  the section 6.1.2, is to  all=
ow multiple Distrbituion source between media source and the receiver. I wi=
ll look at the this case as one special SSM case. because  each of distrbit=
uion source resides at the upstream direction or downstream direction of ot=
her distributions souces. Does it make sense?

TOM: RFC 5760 only defines one DS for a given SSM. If you depart from that,=
 you probably need to address and define this architecture in a better way =
in your draft.

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  typi=
cally an overlay of unicast is used... will the Feedback suppression messag=
e be sent over unicast as well? If so, what is gain compared to just having=
 each RTCP receiver providing each their NACK to the MCU, even in case of a=
 packet loss impacting all receivers?

[Qin]: First The MCU case will not cause NACK Implosion, instead, it may ca=
use FIR implosion.
           Second, using Feedback suppression message can restrict the FIR =
message sent from receivers behind MCU and liberate the media source from  =
FIR implosion.

TOM: it is not important whether this is about FIR or NACK. My point is tha=
t iso having each receiver sending a NACK/FIR, you will now have to send to=
 each receiver a NACK/FIR  suppression in the given scenario. There are no =
differences in terms of Bandwdith overhead or server load in the two cases,=
 so why bother to send suppression messages?


Tom (11) : coming back to my previous comment on SDP description in this dr=
aft : if this draft defines a RTCP feedback suppression message, why is the=
re need for a "announcement" in the SDP for the receivers, that indicates t=
hey may receive such feedback suppression messages?  If a receiver supports=
 the message, it will  simply act accordingly as described in the draft, no=
 need to announce that!..

[Qin]: Okay, I get your point. I would like to put this as open issue, disc=
uss it in the face to face IETF meeting.

Regards
Tom






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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue bgColor=3D#cce8cf>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D783445214-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi Roni,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D783445214-19102010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff size=3D2>In t=
he typical=20
RAMS architecture, the BRS will receive the SSM stream from the DS. If ther=
e are=20
several BRS-es, there may be just one BRS that detects packet loss on its=20
upstream link, but the others will perhaps not, as the packet loss took pla=
ce on=20
SSM tree branch&nbsp;that does not impact the other BRSes. In that case tho=
se=20
RTP receivers associated to the other BRS-es in general will not be impacte=
d and=20
do not need to receive the suppression indication, while the one BRS that d=
id=20
sense the packet loss should probably best send a suppression indication=20
(although there is a chance that only&nbsp;this BRS is impacted by packet l=
oss,=20
and not a single RTP receiver in the field reporting to this BRS/FT... this=
 all=20
depends on the considered topologies). </FONT></SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff size=3D2>As a=
 BRS is=20
1-to-1 associated with a FT, this calls for consideration of architectures =
with=20
(multiple) disjoint FT (and associated BRS in RAMS applications)&nbsp; in t=
he=20
suppression draft. Such disjoint FT/BRS in general cannot act as so-called=
=20
"intermediate nodes" or "translators" because they are not on the direct SS=
M=20
path between the DS and the RTP receivers. </FONT></SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff size=3D2>Anot=
her aspect I=20
want to highlight is that feedback suppression is closely aligned with=20
retransmission. In the case described above, the one BRS that detected pack=
et=20
loss on its upstream link, may try to recover the missing packet (outside t=
he=20
scope of the draft) and retransmit over multicast to the receivers that rep=
ort=20
to the FT. This retransmission towards the receivers is also outside the cu=
rrent=20
scope of the current draft, but the two are somehow linked.&nbsp; In DVB IP=
TV,=20
we introduced dedicated SSMs sourced by the BRS for&nbsp;sending NACK=20
suppression and the (optional) retransmission.&nbsp;In the considered scena=
rio,=20
having a DS sending a missing packet -reported by a FT downstream in the SS=
M=20
tree- over the original SSM&nbsp;is not a good approach IMO, as it results =
in=20
general in unnecessary retransmissions for most receivers.</FONT></SPAN></D=
IV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff size=3D2>Hope=
 this=20
clarifies.</FONT></SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010><FONT face=3DArial color=3D#0000ff=20
size=3D2>Tom</FONT></SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><SPAN=
=20
class=3D783445214-19102010>&nbsp;</SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
</DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft><FONT=
 face=3DTahoma=20
size=3D2><B>From:</B> Roni Even [mailto:Even.roni@huawei.com] <BR><B>Sent:<=
/B>=20
dinsdag 19 oktober 2010 16:46<BR><B>To:</B> Van Caenegem, Tom (Tom); 'Qin W=
u';=20
avt@ietf.org<BR><B>Subject:</B> RE: [AVT] New Version Notification for=20
draft-wu-avt-retransmission-supression-rtp-04<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DWordSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'">Hi=20
Tom,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'">I=20
want to clarify your example since I am not sure what is special here. The =
BRS=20
sends separate unicast streams to each participant so there is reason for=20
suppression on these unicast streams. As for the primary multicast stream f=
rom=20
the DS it goes to all receivers. Now if the BRS identify packet loss on the=
=20
primary stream it should send suppression indication to all members of the=
=20
multicast group regardless to which FT they report.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'">Can=20
you clarify<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'">Roni=20
Even<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid=
; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df=
 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium non=
e; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">=20
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Van=
=20
Caenegem, Tom (Tom)<BR><B>Sent:</B> Tuesday, October 19, 2010 10:27=20
AM<BR><B>To:</B> Qin Wu; avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New Vers=
ion=20
Notification for=20
draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></SPAN></P></DIV></=
DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">H=
i=20
Qin,</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">T=
hank=20
you for your reply.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">R=
eading=20
your responses,you seem to focus really on the SSM&nbsp; case with an=20
architecture where the single DS (there can be only one DS per SSM) hosts t=
he=20
sole feedback target, which also hosts the loss detector function. My=20
interpretation is also that in your draft the loss detector sends the FIR o=
r=20
NACK supression message to the DS, which then forwards/relays this&nbsp;mes=
sage=20
over the SSM to all receivers, right?&nbsp; </SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">M=
y main=20
comment remains the same: you should at least incorporate the architecture =
of=20
RAMS (where&nbsp;there&nbsp;can be several FT&nbsp; colocated with the BRS,=
=20
which are disjoint from the DS). In this case the FT/BRS should be able to=
=20
provide a suppression indication to all receivers that report to this FT (w=
hich=20
is a subset of all receivers that connect to the SSM), which means one shou=
ld=20
not make use of the (original) SSM for providing the suppression=20
indication.&nbsp; Your draft provides a possible solution for only a limite=
d use=20
case, and not the most interesting one, IMO.&nbsp;&nbsp;</SPAN><o:p></o:p><=
/P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">F=
urther=20
down&nbsp;some&nbsp;responses to your answers below.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">T=
om</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" align=3Dcenter>
<HR align=3Dcenter width=3D"100%" SIZE=3D2>
</DIV>
<P class=3DMsoNormal style=3D"MARGIN-BOTTOM: 12pt"><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Qin Wu=20
[mailto:sunseawq@huawei.com] <BR><B>Sent:</B> dinsdag 19 oktober 2010=20
5:04<BR><B>To:</B> Van Caenegem, Tom (Tom); avt@ietf.org<BR><B>Subject:</B>=
 Re:=20
[AVT] New Version Notification for=20
draft-wu-avt-retransmission-supression-rtp-04</SPAN><o:p></o:p></P>
<DIV>
<P class=3DMsoNormal>Hi, Tom:<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>Thank for your comments, please see my reply=20
inline.<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>Regards!<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>-Qin<o:p></o:p></P></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; MARGIN: 5pt 0in 5pt 3.75pt; B=
ORDER-LEFT: black 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: medium none=
">
  <DIV>
  <P class=3DMsoNormal>----- Original Message ----- <SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><o:p></o:p></SPAN></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal style=3D"BACKGROUND: #e4e4e4"><B>From:</B> <SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><A=20
  title=3Dtom.van_caenegem@alcatel-lucent.com=20
  href=3D"mailto:tom.van_caenegem@alcatel-lucent.com"><SPAN=20
  style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman','serif'">Van Cae=
negem,=20
  Tom (Tom)</SPAN></A></SPAN> <SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><o:p></o:p></SPAN></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal><B>To:</B> <SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><A title=3Davt@ietf.org=20
  href=3D"mailto:avt@ietf.org"><SPAN=20
  style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman','serif'">avt@iet=
f.org</SPAN></A></SPAN>=20
  ; <SPAN style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><A=20
  title=3Dsunseawq@huawei.com href=3D"mailto:sunseawq@huawei.com"><SPAN=20
  style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman','serif'">Qin=20
  Wu</SPAN></A></SPAN> <SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><o:p></o:p></SPAN></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal><B>Sent:</B> Tuesday, October 19, 2010 12:08 AM<SPAN=
=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><o:p></o:p></SPAN></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal><B>Subject:</B> Re: [AVT] New Version Notification f=
or=20
  draft-wu-avt-retransmission-supression-rtp-04<SPAN=20
  style=3D"FONT-SIZE: 9pt; FONT-FAMILY: SimSun"><o:p></o:p></SPAN></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Hi Qin,<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>I have some comments (look for TOM below) on the new=
 draft=20
  version 4.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Thx for addressing them,<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Tom<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Section 2 Terminology<o:p></o:p></P></DIV>
  <DIV><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;Lo=
ss Detector:<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times=
 New Roman','serif'"><o:p>&nbsp;</o:p></SPAN></PRE><PRE><SPAN style=3D"FONT=
-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss=
 Detector is one logical function which is used to detect<o:p></o:p></SPAN>=
</PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; the packet loss at the RTP layer and report it to the=
 distribution<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Time=
s New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; source.&nbsp; The func=
tion of the loss reporter may be collocated with<o:p></o:p></SPAN></PRE><PR=
E><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp; or integrated into the same entity.&nbsp; In this case, for a =
session<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New =
Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; defined as having a distribu=
tion source A, on ports n for the RTP<o:p></o:p></SPAN></PRE><PRE><SPAN sty=
le=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; channel and k for the RTCP channel, the Loss Detector are<o:p></o:p></SPA=
N></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; identified by an IP address of distribution source =
A on port k.<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times=
 New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector MAY a=
lso be implemented in one or more entities<o:p></o:p></SPAN></PRE><PRE><SPA=
N style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; different from the distribution source.<o:p></o:p></SPAN></PRE><PRE>=
<SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'"> The function of the=
 loss reporter may be collocated with<o:p></o:p></SPAN></PRE><PRE><SPAN sty=
le=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; or integrated into the same entity.</SPAN><o:p></o:p></PRE><PRE><SPAN sty=
le=3D"FONT-FAMILY: 'Times New Roman','serif'">TOM (1): ..same entity as wha=
t? i guess you mean the DS?</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT=
-FAMILY: 'Times New Roman','serif'">[Qin]: Your are right.&nbsp; In this dr=
aft, the loss detector is just functionality of DS.&nbsp; Furthermore we on=
ly consider the loss detector as part of feedback target within the distrib=
ution source.</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Time=
s New Roman','serif'">Also this is just one example use case to explain how=
 feedback suppression works in SSM use case.</SPAN><o:p></o:p></PRE><PRE><S=
PAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">TOM (2): I do not unde=
rstand how you associate the port k to a loss detector.. what do you mean w=
ith that?</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times Ne=
w Roman','serif'">[Qin]:&nbsp; Sorry for ambiguity of the description here.=
 It actually means that the loss reporter is part of feedback target and sh=
are the same address and port.</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"F=
ONT-FAMILY: 'Times New Roman','serif'">section 3</SPAN><o:p></o:p></PRE><PR=
E><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;</SPAN><o:p>=
</o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">(..=
)</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman'=
,'serif'">In order to detect packet loss before the receivers perceive it, =
one<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roma=
n','serif'">&nbsp;&nbsp; or more intermediate nodes are placed between the =
media source and<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'T=
imes New Roman','serif'">&nbsp;&nbsp; receiver (e.g., Distribution server, =
MCU, RTP translator).&nbsp; These<o:p></o:p></SPAN></PRE><PRE><SPAN style=
=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; intermediaries mon=
itor for packet loss upstream of themselves by<o:p></o:p></SPAN></PRE><PRE>=
<SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; checkin=
g RTP sequence numbers, just as receivers do.&nbsp; Upon detecting<o:p></o:=
p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">=
&nbsp;&nbsp; (or suspecting) an upstream loss, the intermediary may send Fe=
edback<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New R=
oman','serif'">&nbsp;&nbsp; Suppression message towards the receivers as de=
fined in this<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Time=
s New Roman','serif'">&nbsp;&nbsp; specification.<o:p></o:p></SPAN></PRE></=
DIV>
  <DIV>
  <P class=3DMsoNormal>TOM (3) :&nbsp;...based on the two above definitions=
=20
  ...is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;hence&nbsp;equal&nbsp;to&n=
bsp;a&nbsp;"Loss&nbsp;detector"?&nbsp;=20
  <o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>+ How does a Loss detector differ from an RTP=20
  receiver?<o:p></o:p></P></DIV></BLOCKQUOTE>
<BLOCKQUOTE=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; MARGIN: 5pt 0in 5pt 3.75pt; B=
ORDER-LEFT: black 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: medium none=
">
  <DIV>
  <P class=3DMsoNormal>[Qin]: The loss detector is just one functionality o=
f=20
  Distribution Source in the SSM use case.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Maybe we should&nbsp;interpret it as "Loss detection=
=20
  functionality". How the packet loss is detected is out of scope of this=20
  document.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Different from dedicated RTP receiver, the intermedi=
ate=20
  node does not depend on existing NACK message for Retransmission request =
for=20
  packet loss.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>the intermediate node can create message by=20
  itself.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Section 6.1<o:p></o:p></P></DIV><PRE><SPAN style=3D"=
FONT-FAMILY: 'Times New Roman','serif'">In order to avoid the forms of NACK=
 implosion described in section 1,<o:p></o:p></SPAN></PRE><PRE><SPAN style=
=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; the Loss Detector =
is introduced.&nbsp; The Loss Detector detects and<o:p></o:p></SPAN></PRE><=
PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; rep=
orts packet loss, on both the upstream link and the downstream<o:p></o:p></=
SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbs=
p;&nbsp; aggregate link.&nbsp; How the loss detector SHOULD detect the pack=
et loss<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New =
Roman','serif'">&nbsp;&nbsp; is beyond of scope of this document.&nbsp; Whe=
n upstream link or<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: =
'Times New Roman','serif'">&nbsp;&nbsp; downstream aggregate link packet lo=
ss occurs, the Loss Detector MAY<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D=
"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; inform distribution s=
ource of the detected packet loss using Feedback<o:p></o:p></SPAN></PRE><PR=
E><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; Suppr=
ession messages.&nbsp; In response, the distribution source either<o:p></o:=
p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">=
&nbsp;&nbsp; forwards packet loss suppression report received from Loss Det=
ector<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Ro=
man','serif'">&nbsp;&nbsp; or creates a Feedback Suppression report and sen=
ds it to all the RTP<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY=
: 'Times New Roman','serif'">&nbsp;&nbsp; receivers, over the multicast cha=
nnel.&nbsp; This loss suppression report<o:p></o:p></SPAN></PRE><PRE><SPAN =
style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; tells the rec=
eivers that the lost packet will either be forthcoming<o:p></o:p></SPAN></P=
RE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp;=
 from distribution source, or it irretrievably lost such that there is<o:p>=
</o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','seri=
f'">&nbsp;&nbsp; nothing to be gained by the receiver sending a NACK to the=
 media<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New R=
oman','serif'">&nbsp;&nbsp; source.&nbsp; The distribution source then can =
(optionally) ask for the<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FA=
MILY: 'Times New Roman','serif'">&nbsp;&nbsp; lost packets from the media s=
ource on behalf of all the RTP<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"F=
ONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; receivers.</SPAN><o:p><=
/o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">TOM =
(4) : You probably do not want to have normative language in section 6.1, l=
ike "How the loss detector SHOULD (..)". </SPAN><o:p></o:p></PRE><PRE><SPAN=
 style=3D"FONT-FAMILY: 'Times New Roman','serif'">[Qin]:Sorry, I can't get =
it.</SPAN><SPAN style=3D"COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">&n=
bsp;</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"COLOR: blue; FONT-FAMILY: '=
Arial','sans-serif'">TOM: this is really a detail. If you want to keep the =
sentence, you should e.g. say: "How the loss detector can detect the packet=
 loss...</SPAN><o:p></o:p></PRE><PRE>&nbsp;<o:p></o:p></PRE><PRE><SPAN styl=
e=3D"COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;</SPAN><o:p></o:=
p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">TOM (5)=
 : It is pretty clear how a Loss detector will detect packet loss on its up=
stream link, I guess.</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMIL=
Y: 'Times New Roman','serif'">[Qin]: As I said above,&nbsp; Loss detector i=
s just functionality of some Distribution sources.&nbsp; Also what it said =
here in section 6.1 </SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY=
: 'Times New Roman','serif'">is just one example use case.</SPAN><o:p></o:p=
></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">TOM (6):=
&nbsp; What is the rationale for a loss detector to send a Feedback suppres=
sion message to the DS.. why NOT a RTCP NACK FB?</SPAN><o:p></o:p></PRE><PR=
E><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">[Qin]: As I explai=
ned in the last IETF presentaion,&nbsp; this cause NACK scematics mismatch.=
 That's why many people in the last IETF meeeting suggest to create new RTC=
P message during presentation.</SPAN><SPAN style=3D"COLOR: blue; FONT-FAMIL=
Y: 'Arial','sans-serif'">&nbsp;</SPAN><o:p></o:p></PRE><PRE>&nbsp;<o:p></o:=
p></PRE><PRE><SPAN style=3D"COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'"=
>TOM: As your assumption is that the loss reporter is part of the FT that i=
s part of the DS, then there is even no need to specify how the loss detect=
or </SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"COLOR: blue; FONT-FAMILY: 'A=
rial','sans-serif'">reports the packet loss.&nbsp; In the more generic case=
, it makes more sense to me that the loss reporter just uses a NACK (or a F=
IR) if it detects packet loss, sent towards its FT or to the DS if this one=
 hosts the FT that the loss detector reports to . It is then up to the FT o=
r the DS to decide whether this message triggers a NACK or FIR suppression =
</SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"COLOR: blue; FONT-FAMILY: 'Aria=
l','sans-serif'">indication to (s subset of) the receivers.&nbsp; This supp=
ression indication does not necessarly require a new message format. A NACK=
/FIR&nbsp; that is sent to a RTP receiver can be defined as a feedback supp=
ression indication. I do not see a good use case why NACKs from&nbsp; recei=
vers&nbsp; would be reflected back in e.g. SSM use case by DS or FT&nbsp; t=
o all the connected receivers. </SPAN><o:p></o:p></PRE><PRE><SPAN style=3D"=
COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">If we define that a NACK/FI=
R&nbsp; for a SSM may only be sent TOWARDS a RTP receiver, if this is for s=
uppression indication, there is no ambiguity possibly.&nbsp;&nbsp; </SPAN><=
o:p></o:p></PRE><PRE>&nbsp;<o:p></o:p></PRE><PRE><SPAN style=3D"COLOR: blue=
; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;</SPAN><o:p></o:p></PRE><PRE><SP=
AN style=3D"FONT-FAMILY: 'Times New Roman','serif'">Section 6.1.1</SPAN><o:=
p></o:p></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">I=
f there are multiple Loss Detectors looking at the same RTP stream,<o:p></o=
:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'"=
>&nbsp;&nbsp; then the loss may be identified by more than one and those de=
tecting<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New =
Roman','serif'">&nbsp;&nbsp; the loss will all send requests for the same p=
acket loss.&nbsp; In this<o:p></o:p></SPAN></PRE><PRE><SPAN style=3D"FONT-F=
AMILY: 'Times New Roman','serif'">&nbsp;&nbsp; case, the distribution sourc=
e MUST filter the duplicated packet loss<o:p></o:p></SPAN></PRE><PRE><SPAN =
style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; request out a=
nd only forward one copy of the RTCP Feedback report<o:p></o:p></SPAN></PRE=
><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">&nbsp;&nbsp; p=
acket from the first Loss Detector to the group impacted by packet<o:p></o:=
p></SPAN></PRE><PRE><SPAN style=3D"FONT-FAMILY: 'Times New Roman','serif'">=
&nbsp;&nbsp; loss.<o:p></o:p></SPAN></PRE>
  <DIV>
  <P class=3DMsoNormal>TOM (7) : how will the DS send only a copy of the RT=
CP FB=20
  report packet to the group impacted by packet loss. Is this done over the=
 SSM?=20
  <o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin]: Sorry, maybe I sould say <o:p></o:p></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal>"<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Section 6.1.1<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>If there are multiple Distrbitution Source with the =
loss=20
  detection support looking at the same RTP stream.....<o:p></o:p></P></DIV=
>
  <DIV>
  <P class=3DMsoNormal>"<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Yes,&nbsp; in the SSM use case, DS send only a copy =
of the=20
  RTCP Feedback Suppression Report packet the group via=20
SSM.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>In your example you seem to focus only on architectu=
res=20
  with a DS which also hosts the FT and the loss reporter.&nbsp; There shou=
ld be=20
  sections covering cases where the DS is disjoint from the FT, with severa=
l FTs=20
  and also several loss reporters disjoint from the FT and the=20
  DS.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin] I am not sure we need to address all the possi=
ble SSM=20
  use cases. In this draft, we just give&nbsp;some examples of SSM use=20
  case.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>TOM (8) : what do you mean in above text with "RTCP =
FB=20
  report".. is that the suppression message you define in this=20
  draft?<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin]: Yes,&nbsp; RTCP Feedback Suppression Report.=
=20
  <o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>TOM (9) : In section&nbsp; "6.1.2.&nbsp; Distributio=
n=20
  Source Feedback Summary Model" it is not clear what your architecture loo=
ks=20
  like. You talk about multiple distribution sources.. whcih implies multip=
le=20
  SSMs.. Can you elaborate on that?<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin]:&nbsp; The scenario we are talking about in&nb=
sp; the=20
  section 6.1.2, is&nbsp;to &nbsp;allow multiple Distrbituion source betwee=
n=20
  media source and the receiver. I will look at the this case as one specia=
l SSM=20
  case. because&nbsp; each of distrbituion source resides at the upstream=20
  direction or downstream direction of other distributions souces. Does it =
make=20
  sense?<SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'"=
>&nbsp;</SPAN><o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'"=
>TOM:=20
  RFC 5760 only defines one DS for a given SSM. If you depart from that, yo=
u=20
  probably need to address and define&nbsp;this architecture in a better wa=
y in=20
  your draft.</SPAN>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Tom (10) :&nbsp; In section&nbsp; 6.3.&nbsp; Multipo=
int=20
  Control Unit (MCU) use case,&nbsp; typically an overlay of unicast is use=
d...=20
  will the Feedback suppression message be sent over unicast as well? If so=
,=20
  what is gain compared to just having each RTCP receiver providing each th=
eir=20
  NACK to the MCU, even in case of a packet loss impacting all=20
  receivers?<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin]: First The MCU case will not cause NACK Implos=
ion,=20
  instead, it may cause FIR implosion.<o:p></o:p></P></DIV>
  <DIV>
  <P=20
  class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;=20
  Second, using Feedback suppression message can restrict the FIR message s=
ent=20
  from receivers&nbsp;behind MCU and liberate the media source from&nbsp; F=
IR=20
  implosion.<SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'"=
>&nbsp;</SPAN><o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'"=
>TOM:=20
  it is not important whether this is about FIR or NACK. My point is that i=
so=20
  having each receiver sending a NACK/FIR, you will now have to send to eac=
h=20
  receiver a NACK/FIR&nbsp; suppression in the given scenario. There are no=
=20
  differences in terms of Bandwdith overhead or server load in the two case=
s, so=20
  why bother to send suppression messages?</SPAN><o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Tom (11) : coming back to my previous comment on SDP=
=20
  description in this draft&nbsp;:&nbsp;if this draft defines a RTCP feedba=
ck=20
  suppression message, why is there need for a "announcement" in the SDP fo=
r the=20
  receivers, that indicates they may receive such feedback suppression=20
  messages?&nbsp; If a receiver supports the message, it will&nbsp;=20
  simply&nbsp;act accordingly as described in the draft, no need to announc=
e=20
  that!.. <o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>[Qin]: Okay, I get your point. I would like to put t=
his as=20
  open issue, discuss it in the face to face IETF meeting.<o:p></o:p></P></=
DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Regards<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>Tom<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
  <DIV>
  <P=20
class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV></BLOCKQUOTE></DIV></DIV></BOD=
Y></HTML>

--_000_EC3FD58E75D43A4F8807FDE0749175460E497311FRMRSSXCHMBSB1d_--

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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating
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Hi Bob,

I definitely see some points about what you write. I also think ECN for
RTP has made all effort it can to enable a future redefining of one of
the ECT values. As one of the ADs that was involved in chartering CONEX
I am pretty well aware of it. However, this is experimental work that
needs to prove its benefit and maybe more than anything else have a
deployment story. Based on this we have take a reasonable approach on
dealing with ECT(1).

I am the one that thought we should included the NONCE as a module in
the solution. I kind of agree that in the general case you simply have
to not support it to avoid it. In addition I agree that pushing TCP
flows out of the way is reasonably simple and does provide a benefit for
the app doing it. A lot of todays non-congestion controlled UDP based
real-time applications are based on doing exactly that. As long as you
are running at rates that are almost always supported you simply go grab
the space you need. Well, that work up to a certain limit and a certain
traffic mix. When sufficient many uncontrolled UDP flows share a
bottleneck you end up in heavy congestion. Fortunately most of these
flow are related to an actual human that will terminate that flow if it
is unusable.

I did perceive Nonce as mechanism some infrastructure devices may need.
Especially where there is incentive to lie and the traffic mix is RTP
heavy and have large flows. Take IPTV as an example. In those cases
there is a sender that desires to not over tax the infrastructure
because that brings down the quality of experience for several users.

However, I am fine with moving out the Nonce stuff into a separate
draft. I might come back to particular points later.

Cheers

Magnus



Bob Briscoe skrev 2010-08-06 00:41:
> Magnus, Ingemar, Colin, Piers, Ken,
> 
> Below are my main technical concerns with 
> draft-ietf-avt-ecn-for-rtp-02 as promised.
> I have used the line numbering here:
> <http://tools.ietf.org/idnits?url=http://tools.ietf.org/id/draft-ietf-avt-ecn-for-rtp-02.txt>
> 
> T1/ Combining pkts
> T2/ ECN-IP-RTCP
> T3/ Anti-Cheating                       <<<---THIS EMAIL
> T4/ Logging Not-ECT fall-back
> T5/ Delivery %age ECT vs Not-ECT
> T6/ Pkt Reordering around session start
> 
> 
> T3a/ Nonce must be non-negotiable for a receiver
> -------------------
> The addition of ECN nonce support to RTP/RTCP adds considerable 
> complexity. In return we get protection against cheating in just one 
> rather contrived scenario - we have to assume the sender wants to 
> altruistically protect the network and other users against a cheating receiver.
> "
> 1204       The assumption about the senders acting on the behalf of the network
> 1205       may be reduced due to the nature of peer-to-peer use of 
> RTP.  Still a
> 1206       significant portion of RTP senders are infrastructure devices (for
> 1207       example, streaming media servers) that do have an interest in
> 1208       protecting both service quality and the network.
> "
> OK, perhaps not so contrived. We'll return to this in a second...
> 
> "
> 1208                                                         In addition, as
> 1209       real-time media is commonly sensitive to increased delay and packet
> 1210       loss, it will be in both media sender and receivers interest to
> 1211       minimise the number and duration of any congestion events as they
> 1212       will affect media quality.
> "
> "
> 1348       behavior.  We note that it appears counter-productive for a receiver
> 1349       to attempt to cheat as it most likely will have negative impact on
> 1350       its media quality.
> "
> Not so. Point to any evidence that cheating makes things worse for 
> yourself. This argument is one of those fallacies that keeps hanging 
> around from the 1990s. If others are responding to congestion, I get 
> higher rate and/or better quality media by not responding to 
> congestion. And on the current Internet most other traffic is 
> responding to congestion (TCP or BitTorrent uTP), so I can take 
> advantage of that.
> 
> Let's return to the infrastructure devices argument, which seems more 
> plausible...
> "
> 619        o  The "nonce" parameter may be used to signal whether the ECN nonce
> 620           is to be used in the session.  This parameter takes two values;
> 621           "nonce=1" for nonce proposed or shall be used, and 
> "nonce=0" for622             no nonce.  If this parameter is not 
> specified, the default is no
> 623           nonce."A receiver that intends to cheat will just claim 
> it does not support the nonce. Therefore, for any sender to have the 
> option to use the nonce, ideally it should have always been mandatory 
> for all receivers to implement nonce support.
> 
> However, many RTP receiver implementations already exist without ECN 
> support, let alone nonce support. Therefore any receiver that wishes 
> to cheat can just hide among the crowd who do not include or 
> recognise an "a=ecn-capable-rtp" attribute at all.
> 
> A sender could punish receivers that do not support the nonce, for 
> instance with a lower rate. But that makes initial deployment hard, 
> as few receivers support it to start with, so you cannot punish 
> nearly everyone who says they don't support ECN just in case they are lying.
> 
> Perhaps instead, a sender could reward receivers that support the 
> nonce. For instance the reward could be a higher rate. But higher 
> than what? Presumably higher than the rate that is 'fair' to everyone 
> else on the network.
> 
> But, let's go back over logic so far. A sender only uses the ECN 
> nonce if it is trying to altruistically protect the interests of 
> others on the network. But in order to create the necessary 
> incentives for receivers to use the nonce, a sender has to first take 
> more rate from others than its 'fair' entitlement, thus negating its 
> intended altruism.
> 
> If a bloke approached you saying, "Lend me $100 so I can reward any 
> thieves who can prove they have not stolen from you," would you give 
> the bloke $100?
> 
> T3b/ Avoid precluding future uses of the experimental ECT(1) codepoint.
> -------------------
> The IETF has just chartered a new working group on Congestion 
> Exposure (ConEx) as a way to more comprehensively handle cheating 
> with respect to congestion: ConEx is a proposed change to IP to give 
> a sender a field in which it can expose expected congestion to the 
> network. Those working on ConEx claim networks can negate any gain a 
> receiver might get from suppressing congestion feedback as well as 
> negating any gain a sender might get from suppressing congestion 
> exposure back to the network. Once truthful congestion is exposed to 
> the network by the sender, the network can encourage and ultimately 
> force senders to respond to congestion.
> 
> The aim is to allow and encourage a whole range of responses to 
> congestion, not to force a particular response (ie. not just TCP). So 
> low bit-rate inelastic sessions will typically be able to be 
> unresponsive to congestion, while high bit-rate inelastic sessions 
> would come under more pressure to be adaptive.
> 
> This 6pp paper describes how ConEx could be used to encourage one 
> site's elastic flows to compensate for its inelastic flows, but 
> ultimately, if high congestion persists, making even the inelastic 
> flows adapt or stop.
>          "Policing Freedom to Use the Internet Resource Pool"
>          <http://www.bobbriscoe.net/projects/refb/#polfree>
> 
> ConEx is currently chartered to produce an experimental protocol, so 
> we cannot yet know whether it will be successful in its goals (I led 
> the research behind ConEx, so I can only say that so far it has been 
> robust to all the attacks the research community has thrown at it).
> 
> The ECN nonce RFC is also experimental. Use of the nonce has been 
> specified in TCP and DCCP, but not implemented as far as anyone knows.
> 
> ConEx might well need to take over the ECT(1) codepoint that is 
> currently assigned to the ECN nonce experiment.
> 
> Therefore, I suggest the RTP-ECN spec suspends judgement between the 
> two approaches. Rather than building use of ECT(1) into RTP-ECN by 
> default, I suggest RTP-ECN is specified in two stages:
> i) without committing to either anti-cheating approach
> ii) an additional option (in a separate I-D) comitting to either the 
> ECN Nonce or ConEx
> 
> The downside is that this will create additional deployment options. 
> But the RTP-ECN draft already implies that implementation of nonce 
> support is optional.
> 
> If an RTP-ECN implementation does not use the ECN nonce for 
> anti-cheating, we need to ensure it will not use the ECT(1) codepoint 
> at all. Otherwise this would unnecessarily preclude using ECT(1) for 
> ConEx, or for some other purpose.
> 
> There is a considerable amount of text woven into the RTP-ECN draft 
> around using ECT(1) and the nonce. If you agree with this suggestion, 
> I can mark up which text needs removing.
> 
> T3c/ Misbehaving end-points
> -------------------
> "
> 1968                                                      It is far 
> from certain
> 1969          that a receiver would be able to get a significant 
> larger share of
> 1970          the resources.  That assumes a high enough level of aggregation
> 1971          that there are flows to acquire shares from.
> "
> Don't agree. It's completely certain that an unresponsive receiver 
> (or sender) can get a significantly larger share of the resources, 
> whether or not it uses ECN. Just try it. Send an unresponsive flow 
> along with some responsive flows (e.g TCP) and the TCPs reduce their 
> rates while the unresponsive flow gets whatever rate it chooses to send at.
> "
> 1971                                                        The risk 
> of cheating
> 1972          is that failure to react to congestion results in packet loss and
> 1973          increased path delay.
> "
> If a cheating unresponsive flow leaves N times less bandwidth 
> available than it would have done were it responsive, it will force 
> long-running TCP flows to each reduce their bit-rate by N times, 
> making their transfers last N times longer.
> These TCP flows will raise the ECN marking probability by about N^2 
> times (even if the TCP flows are not ECN-capable). So the marking 
> /rate/ will increase by N times. Any competing adaptive real-time 
> flows will reduce their rates in response to the increased 
> loss/marking probability as well.
> 
> If the bottleneck queue is ECN-capable the cheating flow won't care 
> about a higher ECN marking rate, as its own risk of loss will still 
> be extremely low if it is sending ECT packets.
> 
> Considering any short flows competing for the same bottleneck, the 
> amount of congestion they cause by overshooting as each leaves TCP 
> slow start should reduce by about N times. Therefore, by hogging more 
> capacity the unresponsive flow also reduces the risk that it will 
> suffer any losses from other TCPs overshooting slow start.
> If the bottleneck queue is not ECN-capable, delay to the cheating 
> flow should not be significantly increased by its own cheating, 
> whether the queue is drop-tail or AQM."
> 1973                                 To mitigate the risk of cheating receivers
> 1974          the solution include ECN-Nonce that makes it probabilistically
> 1975          unlikely that a receiver can cheat for more than a few packets
> 1976          before being found out.
> "
> See earlier point about the a nonce sender relying on a cheating 
> receiver choosing to volunteer to support the nonce. I really would 
> rather Internet Drafts stopped making unsubstantiated references to 
> the ECN nonce as if it solves the problem of a cheating receiver.
> 
> "
> 1979       Receivers misbehaving:  A receiver may
> 1980                                                         [...]  simple
> 1981          provide invalid ECN-nonce values.
> "
> This is not a workable attack. If a receiver provides invalid 
> ECN-nonce values, the sender will consider it has failed the nonce 
> test and be entitled to stop sending to it.
> 
> "
> 1979       Receivers misbehaving:  A receiver may prevent the usage 
> of ECN in an
> 1980          RTP session by reporting itself as non ECN capable [...]
> 1981                                             Thus forcing the 
> sender to turn
> 1982          off usage of ECN.  In a point-to-point scenario there is little
> 1983          incentive to do this as it will only affect the receiver.  Thus
> 1984          failing to utilise an optimisation.
> "
> Unlikely to be true. ECN provides very little benefit - it is a tiny 
> optimisation compared to the potential benefit to a receiver from not 
> reporting congestion events, which might allow it to receive at the 
> highest codec rate. Having just above normal loss levels might be a 
> little worse than no losses by using ECN. But getting a high codec 
> rate will surely be worth it, while other r-t flows will have to run 
> at the lowest codec, or other TCP flows will have to reduce to 
> whatever bandwidth is left.
> 
> "
> 1965       Receivers cheating
> 1979       Receivers misbehaving:
> 1997       Misbehaving Senders:
> "
> In all these cases, ConEx is intended to solve the problem, 
> particularly when ECN is used, whether with RTP or any other transport.
> 
> Cheers
> 
> Bob
> 
> 
> ________________________________________________________________
> Bob Briscoe,                                BT Innovate & Design  
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Date: Tue, 19 Oct 2010 18:00:33 +0200
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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Hi Tom,

In the current RAMS there is no way for the BRS to send a suppression
indication , the only available way is in the main SSM stream. I think that
such BRS will be reporting nodes to the DS in the simple case. Are you
suggesting that we define a new SSM session that will be initiated by the
BRS. I think this will make more sense if we also address the retransmission
support you described since the SSM session will serve for the
retransmission.

Currently this part is not in scope but we may ask the WG if they want to
add this specific scenario

 

Roni Even

 

From: Van Caenegem, Tom (Tom) [mailto:tom.van_caenegem@alcatel-lucent.com] 
Sent: Tuesday, October 19, 2010 5:31 PM
To: Roni Even; 'Qin Wu'; avt@ietf.org
Subject: RE: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

 

Hi Roni,

 

In the typical RAMS architecture, the BRS will receive the SSM stream from
the DS. If there are several BRS-es, there may be just one BRS that detects
packet loss on its upstream link, but the others will perhaps not, as the
packet loss took place on SSM tree branch that does not impact the other
BRSes. In that case those RTP receivers associated to the other BRS-es in
general will not be impacted and do not need to receive the suppression
indication, while the one BRS that did sense the packet loss should probably
best send a suppression indication (although there is a chance that only
this BRS is impacted by packet loss, and not a single RTP receiver in the
field reporting to this BRS/FT... this all depends on the considered
topologies). 

 

As a BRS is 1-to-1 associated with a FT, this calls for consideration of
architectures with (multiple) disjoint FT (and associated BRS in RAMS
applications)  in the suppression draft. Such disjoint FT/BRS in general
cannot act as so-called "intermediate nodes" or "translators" because they
are not on the direct SSM path between the DS and the RTP receivers. 

 

Another aspect I want to highlight is that feedback suppression is closely
aligned with retransmission. In the case described above, the one BRS that
detected packet loss on its upstream link, may try to recover the missing
packet (outside the scope of the draft) and retransmit over multicast to the
receivers that report to the FT. This retransmission towards the receivers
is also outside the current scope of the current draft, but the two are
somehow linked.  In DVB IPTV, we introduced dedicated SSMs sourced by the
BRS for sending NACK suppression and the (optional) retransmission. In the
considered scenario, having a DS sending a missing packet -reported by a FT
downstream in the SSM tree- over the original SSM is not a good approach
IMO, as it results in general in unnecessary retransmissions for most
receivers.

 

Hope this clarifies.

 

Tom

 

  _____  

From: Roni Even [mailto:Even.roni@huawei.com] 
Sent: dinsdag 19 oktober 2010 16:46
To: Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org
Subject: RE: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

Hi Tom,

I want to clarify your example since I am not sure what is special here. The
BRS sends separate unicast streams to each participant so there is reason
for suppression on these unicast streams. As for the primary multicast
stream from the DS it goes to all receivers. Now if the BRS identify packet
loss on the primary stream it should send suppression indication to all
members of the multicast group regardless to which FT they report.

Can you clarify

 

Roni Even

 

 

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Van
Caenegem, Tom (Tom)
Sent: Tuesday, October 19, 2010 10:27 AM
To: Qin Wu; avt@ietf.org
Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

 

Hi Qin,

 

Thank you for your reply.

 

Reading your responses,you seem to focus really on the SSM  case with an
architecture where the single DS (there can be only one DS per SSM) hosts
the sole feedback target, which also hosts the loss detector function. My
interpretation is also that in your draft the loss detector sends the FIR or
NACK supression message to the DS, which then forwards/relays this message
over the SSM to all receivers, right?  

 

My main comment remains the same: you should at least incorporate the
architecture of RAMS (where there can be several FT  colocated with the BRS,
which are disjoint from the DS). In this case the FT/BRS should be able to
provide a suppression indication to all receivers that report to this FT
(which is a subset of all receivers that connect to the SSM), which means
one should not make use of the (original) SSM for providing the suppression
indication.  Your draft provides a possible solution for only a limited use
case, and not the most interesting one, IMO.  

 

Further down some responses to your answers below.

 

Tom

 

 

  _____  

From: Qin Wu [mailto:sunseawq@huawei.com] 
Sent: dinsdag 19 oktober 2010 5:04
To: Van Caenegem, Tom (Tom); avt@ietf.org
Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

Hi, Tom:

Thank for your comments, please see my reply inline.

 

Regards!

-Qin

----- Original Message ----- 

From:  <mailto:tom.van_caenegem@alcatel-lucent.com> Van Caenegem, Tom (Tom) 

To:  <mailto:avt@ietf.org> avt@ietf.org ;  <mailto:sunseawq@huawei.com> Qin
Wu 

Sent: Tuesday, October 19, 2010 12:08 AM

Subject: Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04

 

Hi Qin,

 

I have some comments (look for TOM below) on the new draft version 4.

Thx for addressing them,

Tom

 

 

Section 2 Terminology

 Loss Detector:
 
      The Loss Detector is one logical function which is used to detect
      the packet loss at the RTP layer and report it to the distribution
      source.  The function of the loss reporter may be collocated with
      or integrated into the same entity.  In this case, for a session
      defined as having a distribution source A, on ports n for the RTP
      channel and k for the RTCP channel, the Loss Detector are
      identified by an IP address of distribution source A on port k.
      The Loss Detector MAY also be implemented in one or more entities
      different from the distribution source.
 The function of the loss reporter may be collocated with
      or integrated into the same entity.
TOM (1): ..same entity as what? i guess you mean the DS?
[Qin]: Your are right.  In this draft, the loss detector is just
functionality of DS.  Furthermore we only consider the loss detector as part
of feedback target within the distribution source.
Also this is just one example use case to explain how feedback suppression
works in SSM use case.
TOM (2): I do not understand how you associate the port k to a loss
detector.. what do you mean with that?
[Qin]:  Sorry for ambiguity of the description here. It actually means that
the loss reporter is part of feedback target and share the same address and
port.
section 3
 
(..)
In order to detect packet loss before the receivers perceive it, one
   or more intermediate nodes are placed between the media source and
   receiver (e.g., Distribution server, MCU, RTP translator).  These
   intermediaries monitor for packet loss upstream of themselves by
   checking RTP sequence numbers, just as receivers do.  Upon detecting
   (or suspecting) an upstream loss, the intermediary may send Feedback
   Suppression message towards the receivers as defined in this
   specification.

TOM (3) : ...based on the two above definitions ...is an intermediate node
hence equal to a "Loss detector"?  

+ How does a Loss detector differ from an RTP receiver?

[Qin]: The loss detector is just one functionality of Distribution Source in
the SSM use case.

Maybe we should interpret it as "Loss detection functionality". How the
packet loss is detected is out of scope of this document.

Different from dedicated RTP receiver, the intermediate node does not depend
on existing NACK message for Retransmission request for packet loss.

the intermediate node can create message by itself.

 

Section 6.1

In order to avoid the forms of NACK implosion described in section 1,
   the Loss Detector is introduced.  The Loss Detector detects and
   reports packet loss, on both the upstream link and the downstream
   aggregate link.  How the loss detector SHOULD detect the packet loss
   is beyond of scope of this document.  When upstream link or
   downstream aggregate link packet loss occurs, the Loss Detector MAY
   inform distribution source of the detected packet loss using Feedback
   Suppression messages.  In response, the distribution source either
   forwards packet loss suppression report received from Loss Detector
   or creates a Feedback Suppression report and sends it to all the RTP
   receivers, over the multicast channel.  This loss suppression report
   tells the receivers that the lost packet will either be forthcoming
   from distribution source, or it irretrievably lost such that there is
   nothing to be gained by the receiver sending a NACK to the media
   source.  The distribution source then can (optionally) ask for the
   lost packets from the media source on behalf of all the RTP
   receivers.
TOM (4) : You probably do not want to have normative language in section
6.1, like "How the loss detector SHOULD (..)". 
[Qin]:Sorry, I can't get it. 
TOM: this is really a detail. If you want to keep the sentence, you should
e.g. say: "How the loss detector can detect the packet loss...
 
 
TOM (5) : It is pretty clear how a Loss detector will detect packet loss on
its upstream link, I guess.
[Qin]: As I said above,  Loss detector is just functionality of some
Distribution sources.  Also what it said here in section 6.1 
is just one example use case.
TOM (6):  What is the rationale for a loss detector to send a Feedback
suppression message to the DS.. why NOT a RTCP NACK FB?
[Qin]: As I explained in the last IETF presentaion,  this cause NACK
scematics mismatch. That's why many people in the last IETF meeeting suggest
to create new RTCP message during presentation. 
 
TOM: As your assumption is that the loss reporter is part of the FT that is
part of the DS, then there is even no need to specify how the loss detector 
reports the packet loss.  In the more generic case, it makes more sense to
me that the loss reporter just uses a NACK (or a FIR) if it detects packet
loss, sent towards its FT or to the DS if this one hosts the FT that the
loss detector reports to . It is then up to the FT or the DS to decide
whether this message triggers a NACK or FIR suppression 
indication to (s subset of) the receivers.  This suppression indication does
not necessarly require a new message format. A NACK/FIR  that is sent to a
RTP receiver can be defined as a feedback suppression indication. I do not
see a good use case why NACKs from  receivers  would be reflected back in
e.g. SSM use case by DS or FT  to all the connected receivers. 
If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP
receiver, if this is for suppression indication, there is no ambiguity
possibly.   
 
 
Section 6.1.1
If there are multiple Loss Detectors looking at the same RTP stream,
   then the loss may be identified by more than one and those detecting
   the loss will all send requests for the same packet loss.  In this
   case, the distribution source MUST filter the duplicated packet loss
   request out and only forward one copy of the RTCP Feedback report
   packet from the first Loss Detector to the group impacted by packet
   loss.

TOM (7) : how will the DS send only a copy of the RTCP FB report packet to
the group impacted by packet loss. Is this done over the SSM? 

 

[Qin]: Sorry, maybe I sould say 

"

Section 6.1.1

If there are multiple Distrbitution Source with the loss detection support
looking at the same RTP stream.....

"

Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback
Suppression Report packet the group via SSM.

 

In your example you seem to focus only on architectures with a DS which also
hosts the FT and the loss reporter.  There should be sections covering cases
where the DS is disjoint from the FT, with several FTs and also several loss
reporters disjoint from the FT and the DS.

 

[Qin] I am not sure we need to address all the possible SSM use cases. In
this draft, we just give some examples of SSM use case.

 

TOM (8) : what do you mean in above text with "RTCP FB report".. is that the
suppression message you define in this draft?

 

[Qin]: Yes,  RTCP Feedback Suppression Report. 

 

TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary Model"
it is not clear what your architecture looks like. You talk about multiple
distribution sources.. whcih implies multiple SSMs.. Can you elaborate on
that?

 

[Qin]:  The scenario we are talking about in  the section 6.1.2, is to
allow multiple Distrbituion source between media source and the receiver. I
will look at the this case as one special SSM case. because  each of
distrbituion source resides at the upstream direction or downstream
direction of other distributions souces. Does it make sense? 

 

TOM: RFC 5760 only defines one DS for a given SSM. If you depart from that,
you probably need to address and define this architecture in a better way in
your draft. 

 

Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,
typically an overlay of unicast is used... will the Feedback suppression
message be sent over unicast as well? If so, what is gain compared to just
having each RTCP receiver providing each their NACK to the MCU, even in case
of a packet loss impacting all receivers?

 

[Qin]: First The MCU case will not cause NACK Implosion, instead, it may
cause FIR implosion.

           Second, using Feedback suppression message can restrict the FIR
message sent from receivers behind MCU and liberate the media source from
FIR implosion. 

 

TOM: it is not important whether this is about FIR or NACK. My point is that
iso having each receiver sending a NACK/FIR, you will now have to send to
each receiver a NACK/FIR  suppression in the given scenario. There are no
differences in terms of Bandwdith overhead or server load in the two cases,
so why bother to send suppression messages?

 

 

Tom (11) : coming back to my previous comment on SDP description in this
draft : if this draft defines a RTCP feedback suppression message, why is
there need for a "announcement" in the SDP for the receivers, that indicates
they may receive such feedback suppression messages?  If a receiver supports
the message, it will  simply act accordingly as described in the draft, no
need to announce that!.. 

 

[Qin]: Okay, I get your point. I would like to put this as open issue,
discuss it in the face to face IETF meeting.

 

Regards

Tom

 

 

 

 

 


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<div class=WordSection1>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Tom,<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In the current RAMS there is no way for the BRS to send a
suppression indication , the only available way is in the main SSM stream. I
think that such BRS will be reporting nodes to the DS in the simple case. Are
you suggesting that we define a new SSM session that will be initiated by the
BRS. I think this will make more sense if we also address the retransmission
support you described since the SSM session will serve for the retransmission.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Currently this part is not in scope but we may ask the WG if
they want to add this specific scenario<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'>

<div>

<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>

<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Van Caenegem, Tom
(Tom) [mailto:tom.van_caenegem@alcatel-lucent.com] <br>
<b>Sent:</b> Tuesday, October 19, 2010 5:31 PM<br>
<b>To:</b> Roni Even; 'Qin Wu'; avt@ietf.org<br>
<b>Subject:</b> RE: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></span></p>

</div>

</div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi Roni,</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>In the typical RAMS architecture, the BRS will receive the SSM
stream from the DS. If there are several BRS-es, there may be just one BRS that
detects packet loss on its upstream link, but the others will perhaps not, as
the packet loss took place on SSM tree branch&nbsp;that does not impact the
other BRSes. In that case those RTP receivers associated to the other BRS-es in
general will not be impacted and do not need to receive the suppression
indication, while the one BRS that did sense the packet loss should probably
best send a suppression indication (although there is a chance that
only&nbsp;this BRS is impacted by packet loss, and not a single RTP receiver in
the field reporting to this BRS/FT... this all depends on the considered
topologies). </span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>As a BRS is 1-to-1 associated with a FT, this calls for
consideration of architectures with (multiple) disjoint FT (and associated BRS
in RAMS applications)&nbsp; in the suppression draft. Such disjoint FT/BRS in
general cannot act as so-called &quot;intermediate nodes&quot; or &quot;translators&quot;
because they are not on the direct SSM path between the DS and the RTP
receivers. </span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Another aspect I want to highlight is that feedback suppression is
closely aligned with retransmission. In the case described above, the one BRS
that detected packet loss on its upstream link, may try to recover the missing
packet (outside the scope of the draft) and retransmit over multicast to the
receivers that report to the FT. This retransmission towards the receivers is
also outside the current scope of the current draft, but the two are somehow
linked.&nbsp; In DVB IPTV, we introduced dedicated SSMs sourced by the BRS
for&nbsp;sending NACK suppression and the (optional) retransmission.&nbsp;In
the considered scenario, having a DS sending a missing packet -reported by a FT
downstream in the SSM tree- over the original SSM&nbsp;is not a good approach
IMO, as it results in general in unnecessary retransmissions for most
receivers.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hope this clarifies.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Tom</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<div class=MsoNormal align=center style='text-align:center'>

<hr size=2 width="100%" align=center>

</div>

<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Roni Even [mailto:Even.roni@huawei.com] <br>
<b>Sent:</b> dinsdag 19 oktober 2010 16:46<br>
<b>To:</b> Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org<br>
<b>Subject:</b> RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04</span><o:p></o:p></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Tom,<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I want to clarify your example since I am not sure what is
special here. The BRS sends separate unicast streams to each participant so
there is reason for suppression on these unicast streams. As for the primary
multicast stream from the DS it goes to all receivers. Now if the BRS identify
packet loss on the primary stream it should send suppression indication to all
members of the multicast group regardless to which FT they report.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Can you clarify<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'>

<div>

<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>

<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Van
Caenegem, Tom (Tom)<br>
<b>Sent:</b> Tuesday, October 19, 2010 10:27 AM<br>
<b>To:</b> Qin Wu; avt@ietf.org<br>
<b>Subject:</b> Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></span></p>

</div>

</div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi Qin,</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Thank you for your reply.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Reading your responses,you seem to focus really on the SSM&nbsp;
case with an architecture where the single DS (there can be only one DS per
SSM) hosts the sole feedback target, which also hosts the loss detector function.
My interpretation is also that in your draft the loss detector sends the FIR or
NACK supression message to the DS, which then forwards/relays this&nbsp;message
over the SSM to all receivers, right?&nbsp; </span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>My main comment remains the same: you should at least incorporate
the architecture of RAMS (where&nbsp;there&nbsp;can be several FT&nbsp;
colocated with the BRS, which are disjoint from the DS). In this case the
FT/BRS should be able to provide a suppression indication to all receivers that
report to this FT (which is a subset of all receivers that connect to the SSM),
which means one should not make use of the (original) SSM for providing the
suppression indication.&nbsp; Your draft provides a possible solution for only
a limited use case, and not the most interesting one, IMO.&nbsp;&nbsp;</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Further down&nbsp;some&nbsp;responses to your answers below.</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Tom</span><o:p></o:p></p>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<div class=MsoNormal align=center style='text-align:center'>

<hr size=2 width="100%" align=center>

</div>

<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Qin Wu [mailto:sunseawq@huawei.com] <br>
<b>Sent:</b> dinsdag 19 oktober 2010 5:04<br>
<b>To:</b> Van Caenegem, Tom (Tom); avt@ietf.org<br>
<b>Subject:</b> Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04</span><o:p></o:p></p>

<div>

<p class=MsoNormal>Hi, Tom:<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Thank for your comments, please see my reply inline.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Regards!<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>-Qin<o:p></o:p></p>

</div>

<blockquote style='border:none;border-left:solid black 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

<div>

<p class=MsoNormal>----- Original Message ----- <span style='font-size:9.0pt;
font-family:"SimSun","serif"'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal style='background:#E4E4E4'><b>From:</b> <span
style='font-size:9.0pt;font-family:"SimSun","serif"'><a
href="mailto:tom.van_caenegem@alcatel-lucent.com"
title="tom.van_caenegem@alcatel-lucent.com"><span style='font-size:12.0pt;
font-family:"Times New Roman","serif"'>Van Caenegem, Tom (Tom)</span></a></span>
<span style='font-size:9.0pt;font-family:"SimSun","serif"'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>To:</b> <span style='font-size:9.0pt;font-family:"SimSun","serif"'><a
href="mailto:avt@ietf.org" title="avt@ietf.org"><span style='font-size:12.0pt;
font-family:"Times New Roman","serif"'>avt@ietf.org</span></a></span> ; <span
style='font-size:9.0pt;font-family:"SimSun","serif"'><a
href="mailto:sunseawq@huawei.com" title="sunseawq@huawei.com"><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'>Qin Wu</span></a></span>
<span style='font-size:9.0pt;font-family:"SimSun","serif"'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>Sent:</b> Tuesday, October 19, 2010 12:08 AM<span
style='font-size:9.0pt;font-family:"SimSun","serif"'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><b>Subject:</b> Re: [AVT] New Version Notification for
draft-wu-avt-retransmission-supression-rtp-04<span style='font-size:9.0pt;
font-family:"SimSun","serif"'><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=MsoNormal>Hi Qin,<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>I have some comments (look for TOM below) on the new draft
version 4.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Thx for addressing them,<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 2 Terminology<o:p></o:p></p>

</div>

<div><pre><span style='font-family:"Times New Roman","serif"'>&nbsp;Loss Detector:<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'><o:p>&nbsp;</o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector is one logical function which is used to detect<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the packet loss at the RTP layer and report it to the distribution<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; source.&nbsp; The function of the loss reporter may be collocated with<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into the same entity.&nbsp; In this case, for a session<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; defined as having a distribution source A, on ports n for the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channel and k for the RTCP channel, the Loss Detector are<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; identified by an IP address of distribution source A on port k.<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector MAY also be implemented in one or more entities<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; different from the distribution source.<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'> The function of the loss reporter may be collocated with<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into the same entity.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (1): ..same entity as what? i guess you mean the DS?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: Your are right.&nbsp; In this draft, the loss detector is just functionality of DS.&nbsp; Furthermore we only consider the loss detector as part of feedback target within the distribution source.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>Also this is just one example use case to explain how feedback suppression works in SSM use case.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (2): I do not understand how you associate the port k to a loss detector.. what do you mean with that?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]:&nbsp; Sorry for ambiguity of the description here. It actually means that the loss reporter is part of feedback target and share the same address and port.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>section 3</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>(..)</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>In order to detect packet loss before the receivers perceive it, one<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; or more intermediate nodes are placed between the media source and<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receiver (e.g., Distribution server, MCU, RTP translator).&nbsp; These<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; intermediaries monitor for packet loss upstream of themselves by<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; checking RTP sequence numbers, just as receivers do.&nbsp; Upon detecting<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; (or suspecting) an upstream loss, the intermediary may send Feedback<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; Suppression message towards the receivers as defined in this<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; specification.<o:p></o:p></span></pre></div>

<div>

<p class=MsoNormal>TOM (3) :&nbsp;...based on the two above definitions
...is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;hence&nbsp;equal&nbsp;to&nbsp;a&nbsp;&quot;Loss&nbsp;detector&quot;?&nbsp;
<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>+ How does a Loss detector differ from an RTP receiver?<o:p></o:p></p>

</div>

</blockquote>

<blockquote style='border:none;border-left:solid black 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

<div>

<p class=MsoNormal>[Qin]: The loss detector is just one functionality of
Distribution Source in the SSM use case.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Maybe we should&nbsp;interpret it as &quot;Loss detection
functionality&quot;. How the packet loss is detected is out of scope of this
document.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Different from dedicated RTP receiver, the intermediate node
does not depend on existing NACK message for Retransmission request for packet
loss.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>the intermediate node can create message by itself.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 6.1<o:p></o:p></p>

</div>

<pre><span style='font-family:"Times New Roman","serif"'>In order to avoid the forms of NACK implosion described in section 1,<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; the Loss Detector is introduced.&nbsp; The Loss Detector detects and<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; reports packet loss, on both the upstream link and the downstream<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; aggregate link.&nbsp; How the loss detector SHOULD detect the packet loss<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; is beyond of scope of this document.&nbsp; When upstream link or<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; downstream aggregate link packet loss occurs, the Loss Detector MAY<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; inform distribution source of the detected packet loss using Feedback<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; Suppression messages.&nbsp; In response, the distribution source either<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; forwards packet loss suppression report received from Loss Detector<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; or creates a Feedback Suppression report and sends it to all the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receivers, over the multicast channel.&nbsp; This loss suppression report<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; tells the receivers that the lost packet will either be forthcoming<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; from distribution source, or it irretrievably lost such that there is<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; nothing to be gained by the receiver sending a NACK to the media<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; source.&nbsp; The distribution source then can (optionally) ask for the<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; lost packets from the media source on behalf of all the RTP<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; receivers.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (4) : You probably do not want to have normative language in section 6.1, like &quot;How the loss detector SHOULD (..)&quot;. </span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]:Sorry, I can't get it.</span><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>TOM: this is really a detail. If you want to keep the sentence, you should e.g. say: &quot;How the loss detector can detect the packet loss...</span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (5) : It is pretty clear how a Loss detector will detect packet loss on its upstream link, I guess.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: As I said above,&nbsp; Loss detector is just functionality of some Distribution sources.&nbsp; Also what it said here in section 6.1 </span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>is just one example use case.</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>TOM (6):&nbsp; What is the rationale for a loss detector to send a Feedback suppression message to the DS.. why NOT a RTCP NACK FB?</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>[Qin]: As I explained in the last IETF presentaion,&nbsp; this cause NACK scematics mismatch. That's why many people in the last IETF meeeting suggest to create new RTCP message during presentation.</span><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>TOM: As your assumption is that the loss reporter is part of the FT that is part of the DS, then there is even no need to specify how the loss detector </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>reports the packet loss.&nbsp; In the more generic case, it makes more sense to me that the loss reporter just uses a NACK (or a FIR) if it detects packet loss, sent towards its FT or to the DS if this one hosts the FT that the loss detector reports to . It is then up to the FT or the DS to decide whether this message triggers a NACK or FIR suppression </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>indication to (s subset of) the receivers.&nbsp; This suppression indication does not necessarly require a new message format. A NACK/FIR&nbsp; that is sent to a RTP receiver can be defined as a feedback suppression indication. I do not see a good use case why NACKs from&nbsp; receivers&nbsp; would be reflected back in e.g. SSM use case by DS or FT&nbsp; to all the connected receivers. </span><o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>If we define that a NACK/FIR&nbsp; for a SSM may only be sent TOWARDS a RTP receiver, if this is for suppression indication, there is no ambiguity possibly.&nbsp;&nbsp; </span><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre><span
style='font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>Section 6.1.1</span><o:p></o:p></pre><pre><span
style='font-family:"Times New Roman","serif"'>If there are multiple Loss Detectors looking at the same RTP stream,<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; then the loss may be identified by more than one and those detecting<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; the loss will all send requests for the same packet loss.&nbsp; In this<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; case, the distribution source MUST filter the duplicated packet loss<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; request out and only forward one copy of the RTCP Feedback report<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; packet from the first Loss Detector to the group impacted by packet<o:p></o:p></span></pre><pre><span
style='font-family:"Times New Roman","serif"'>&nbsp;&nbsp; loss.<o:p></o:p></span></pre>

<div>

<p class=MsoNormal>TOM (7) : how will the DS send only a copy of the RTCP FB
report packet to the group impacted by packet loss. Is this done over the SSM? <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Sorry, maybe I sould say <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&quot;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Section 6.1.1<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>If there are multiple Distrbitution Source with the loss
detection support looking at the same RTP stream.....<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&quot;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Yes,&nbsp; in the SSM use case, DS send only a copy of the
RTCP Feedback Suppression Report packet the group via SSM.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>In your example you seem to focus only on architectures with
a DS which also hosts the FT and the loss reporter.&nbsp; There should be
sections covering cases where the DS is disjoint from the FT, with several FTs
and also several loss reporters disjoint from the FT and the DS.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin] I am not sure we need to address all the possible SSM
use cases. In this draft, we just give&nbsp;some examples of SSM use case.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>TOM (8) : what do you mean in above text with &quot;RTCP FB
report&quot;.. is that the suppression message you define in this draft?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Yes,&nbsp; RTCP Feedback Suppression Report. <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>TOM (9) : In section&nbsp; &quot;6.1.2.&nbsp; Distribution Source
Feedback Summary Model&quot; it is not clear what your architecture looks like.
You talk about multiple distribution sources.. whcih implies multiple SSMs..
Can you elaborate on that?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]:&nbsp; The scenario we are talking about in&nbsp; the
section 6.1.2, is&nbsp;to &nbsp;allow multiple Distrbituion source between
media source and the receiver. I will look at the this case as one special SSM
case. because&nbsp; each of distrbituion source resides at the upstream
direction or downstream direction of other distributions souces. Does it make
sense?<span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>TOM: RFC 5760 only defines one DS for a given SSM. If you depart
from that, you probably need to address and define&nbsp;this architecture in a
better way in your draft.</span>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom (10) :&nbsp; In section&nbsp; 6.3.&nbsp; Multipoint
Control Unit (MCU) use case,&nbsp; typically an overlay of unicast is used...
will the Feedback suppression message be sent over unicast as well? If so, what
is gain compared to just having each RTCP receiver providing each their NACK to
the MCU, even in case of a packet loss impacting all receivers?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: First The MCU case will not cause NACK Implosion,
instead, it may cause FIR implosion.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Second, using Feedback suppression message can restrict the FIR message sent
from receivers&nbsp;behind MCU and liberate the media source from&nbsp; FIR
implosion.<span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>TOM: it is not important whether this is about FIR or NACK. My
point is that iso having each receiver sending a NACK/FIR, you will now have to
send to each receiver a NACK/FIR&nbsp; suppression in the given scenario. There
are no differences in terms of Bandwdith overhead or server load in the two
cases, so why bother to send suppression messages?</span><o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom (11) : coming back to my previous comment on SDP
description in this draft&nbsp;:&nbsp;if this draft defines a RTCP feedback
suppression message, why is there need for a &quot;announcement&quot; in the
SDP for the receivers, that indicates they may receive such feedback
suppression messages?&nbsp; If a receiver supports the message, it will&nbsp;
simply&nbsp;act accordingly as described in the draft, no need to announce
that!.. <o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>[Qin]: Okay, I get your point. I would like to put this as
open issue, discuss it in the face to face IETF meeting.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Regards<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Tom<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

</blockquote>

</div>

</div>

</div>

</body>

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Roni, Tom,

I'd recommend moving the discussion away from specific scenarios to the =
extent possible, and focussing on the general mechanism. Yes, the draft =
should give examples of how retransmission suppression is used with the =
RAMS scenario, and probably with other scenarios, but those are just =
examples of how to use the general mechanism. If the semantics of that =
general mechanism are well-defined, then how it's used for the specific =
scenarios should become clearer.

Colin



On 19 Oct 2010, at 17:00, Roni Even wrote:
> Hi Tom,
> In the current RAMS there is no way for the BRS to send a suppression =
indication , the only available way is in the main SSM stream. I think =
that such BRS will be reporting nodes to the DS in the simple case. Are =
you suggesting that we define a new SSM session that will be initiated =
by the BRS. I think this will make more sense if we also address the =
retransmission support you described since the SSM session will serve =
for the retransmission.
> Currently this part is not in scope but we may ask the WG if they want =
to add this specific scenario
> =20
> Roni Even
> =20
> From: Van Caenegem, Tom (Tom) =
[mailto:tom.van_caenegem@alcatel-lucent.com]=20
> Sent: Tuesday, October 19, 2010 5:31 PM
> To: Roni Even; 'Qin Wu'; avt@ietf.org
> Subject: RE: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-04
> =20
> Hi Roni,
> =20
> In the typical RAMS architecture, the BRS will receive the SSM stream =
from the DS. If there are several BRS-es, there may be just one BRS that =
detects packet loss on its upstream link, but the others will perhaps =
not, as the packet loss took place on SSM tree branch that does not =
impact the other BRSes. In that case those RTP receivers associated to =
the other BRS-es in general will not be impacted and do not need to =
receive the suppression indication, while the one BRS that did sense the =
packet loss should probably best send a suppression indication (although =
there is a chance that only this BRS is impacted by packet loss, and not =
a single RTP receiver in the field reporting to this BRS/FT... this all =
depends on the considered topologies).
> =20
> As a BRS is 1-to-1 associated with a FT, this calls for consideration =
of architectures with (multiple) disjoint FT (and associated BRS in RAMS =
applications)  in the suppression draft. Such disjoint FT/BRS in general =
cannot act as so-called "intermediate nodes" or "translators" because =
they are not on the direct SSM path between the DS and the RTP =
receivers.
> =20
> Another aspect I want to highlight is that feedback suppression is =
closely aligned with retransmission. In the case described above, the =
one BRS that detected packet loss on its upstream link, may try to =
recover the missing packet (outside the scope of the draft) and =
retransmit over multicast to the receivers that report to the FT. This =
retransmission towards the receivers is also outside the current scope =
of the current draft, but the two are somehow linked.  In DVB IPTV, we =
introduced dedicated SSMs sourced by the BRS for sending NACK =
suppression and the (optional) retransmission. In the considered =
scenario, having a DS sending a missing packet -reported by a FT =
downstream in the SSM tree- over the original SSM is not a good approach =
IMO, as it results in general in unnecessary retransmissions for most =
receivers.
> =20
> Hope this clarifies.
> =20
> Tom
> =20
> From: Roni Even [mailto:Even.roni@huawei.com]=20
> Sent: dinsdag 19 oktober 2010 16:46
> To: Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org
> Subject: RE: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-04
>=20
> Hi Tom,
> I want to clarify your example since I am not sure what is special =
here. The BRS sends separate unicast streams to each participant so =
there is reason for suppression on these unicast streams. As for the =
primary multicast stream from the DS it goes to all receivers. Now if =
the BRS identify packet loss on the primary stream it should send =
suppression indication to all members of the multicast group regardless =
to which FT they report.
> Can you clarify
> =20
> Roni Even
> =20
> =20
> =20
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Van Caenegem, Tom (Tom)
> Sent: Tuesday, October 19, 2010 10:27 AM
> To: Qin Wu; avt@ietf.org
> Subject: Re: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-04
> =20
> Hi Qin,
> =20
> Thank you for your reply.
> =20
> Reading your responses,you seem to focus really on the SSM  case with =
an architecture where the single DS (there can be only one DS per SSM) =
hosts the sole feedback target, which also hosts the loss detector =
function. My interpretation is also that in your draft the loss detector =
sends the FIR or NACK supression message to the DS, which then =
forwards/relays this message over the SSM to all receivers, right?=20
> =20
> My main comment remains the same: you should at least incorporate the =
architecture of RAMS (where there can be several FT  colocated with the =
BRS, which are disjoint from the DS). In this case the FT/BRS should be =
able to provide a suppression indication to all receivers that report to =
this FT (which is a subset of all receivers that connect to the SSM), =
which means one should not make use of the (original) SSM for providing =
the suppression indication.  Your draft provides a possible solution for =
only a limited use case, and not the most interesting one, IMO. =20
> =20
> Further down some responses to your answers below.
> =20
> Tom
> =20
> =20
> From: Qin Wu [mailto:sunseawq@huawei.com]=20
> Sent: dinsdag 19 oktober 2010 5:04
> To: Van Caenegem, Tom (Tom); avt@ietf.org
> Subject: Re: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-04
>=20
> Hi, Tom:
> Thank for your comments, please see my reply inline.
> =20
> Regards!
> -Qin
> ----- Original Message -----
> From: Van Caenegem, Tom (Tom)
> To: avt@ietf.org ; Qin Wu
> Sent: Tuesday, October 19, 2010 12:08 AM
> Subject: Re: [AVT] New Version Notification for =
draft-wu-avt-retransmission-supression-rtp-04
> =20
> Hi Qin,
> =20
> I have some comments (look for TOM below) on the new draft version 4.
> Thx for addressing them,
> Tom
> =20
> =20
> Section 2 Terminology
>  Loss Detector:
> =20
>       The Loss Detector is one logical function which is used to =
detect
>       the packet loss at the RTP layer and report it to the =
distribution
>       source.  The function of the loss reporter may be collocated =
with
>       or integrated into the same entity.  In this case, for a session
>       defined as having a distribution source A, on ports n for the =
RTP
>       channel and k for the RTCP channel, the Loss Detector are
>       identified by an IP address of distribution source A on port k.
>       The Loss Detector MAY also be implemented in one or more =
entities
>       different from the distribution source.
>  The function of the loss reporter may be collocated with
>       or integrated into the same entity.
> TOM (1): ..same entity as what? i guess you mean the DS?
> [Qin]: Your are right.  In this draft, the loss detector is just =
functionality of DS.  Furthermore we only consider the loss detector as =
part of feedback target within the distribution source.
> Also this is just one example use case to explain how feedback =
suppression works in SSM use case.
> TOM (2): I do not understand how you associate the port k to a loss =
detector.. what do you mean with that?
> [Qin]:  Sorry for ambiguity of the description here. It actually means =
that the loss reporter is part of feedback target and share the same =
address and port.
> section 3
> =20
> (..)
> In order to detect packet loss before the receivers perceive it, one
>    or more intermediate nodes are placed between the media source and
>    receiver (e.g., Distribution server, MCU, RTP translator).  These
>    intermediaries monitor for packet loss upstream of themselves by
>    checking RTP sequence numbers, just as receivers do.  Upon =
detecting
>    (or suspecting) an upstream loss, the intermediary may send =
Feedback
>    Suppression message towards the receivers as defined in this
>    specification.
> TOM (3) : ...based on the two above definitions ...is an intermediate =
node hence equal to a "Loss detector"?=20
> + How does a Loss detector differ from an RTP receiver?
> [Qin]: The loss detector is just one functionality of Distribution =
Source in the SSM use case.
> Maybe we should interpret it as "Loss detection functionality". How =
the packet loss is detected is out of scope of this document.
> Different from dedicated RTP receiver, the intermediate node does not =
depend on existing NACK message for Retransmission request for packet =
loss.
> the intermediate node can create message by itself.
> =20
> Section 6.1
> In order to avoid the forms of NACK implosion described in section 1,
>    the Loss Detector is introduced.  The Loss Detector detects and
>    reports packet loss, on both the upstream link and the downstream
>    aggregate link.  How the loss detector SHOULD detect the packet =
loss
>    is beyond of scope of this document.  When upstream link or
>    downstream aggregate link packet loss occurs, the Loss Detector MAY
>    inform distribution source of the detected packet loss using =
Feedback
>    Suppression messages.  In response, the distribution source either
>    forwards packet loss suppression report received from Loss Detector
>    or creates a Feedback Suppression report and sends it to all the =
RTP
>    receivers, over the multicast channel.  This loss suppression =
report
>    tells the receivers that the lost packet will either be forthcoming
>    from distribution source, or it irretrievably lost such that there =
is
>    nothing to be gained by the receiver sending a NACK to the media
>    source.  The distribution source then can (optionally) ask for the
>    lost packets from the media source on behalf of all the RTP
>    receivers.
> TOM (4) : You probably do not want to have normative language in =
section 6.1, like "How the loss detector SHOULD (..)".=20
> [Qin]:Sorry, I can't get it.=20
> TOM: this is really a detail. If you want to keep the sentence, you =
should e.g. say: "How the loss detector can detect the packet loss...
> =20
> =20
> TOM (5) : It is pretty clear how a Loss detector will detect packet =
loss on its upstream link, I guess.
> [Qin]: As I said above,  Loss detector is just functionality of some =
Distribution sources.  Also what it said here in section 6.1=20
> is just one example use case.
> TOM (6):  What is the rationale for a loss detector to send a Feedback =
suppression message to the DS.. why NOT a RTCP NACK FB?
> [Qin]: As I explained in the last IETF presentaion,  this cause NACK =
scematics mismatch. That's why many people in the last IETF meeeting =
suggest to create new RTCP message during presentation.=20
> =20
> TOM: As your assumption is that the loss reporter is part of the FT =
that is part of the DS, then there is even no need to specify how the =
loss detector=20
> reports the packet loss.  In the more generic case, it makes more =
sense to me that the loss reporter just uses a NACK (or a FIR) if it =
detects packet loss, sent towards its FT or to the DS if this one hosts =
the FT that the loss detector reports to . It is then up to the FT or =
the DS to decide whether this message triggers a NACK or FIR suppression=20=

> indication to (s subset of) the receivers.  This suppression =
indication does not necessarly require a new message format. A NACK/FIR  =
that is sent to a RTP receiver can be defined as a feedback suppression =
indication. I do not see a good use case why NACKs from  receivers  =
would be reflected back in e.g. SSM use case by DS or FT  to all the =
connected receivers.=20
> If we define that a NACK/FIR  for a SSM may only be sent TOWARDS a RTP =
receiver, if this is for suppression indication, there is no ambiguity =
possibly.  =20
> =20
> =20
> Section 6.1.1
> If there are multiple Loss Detectors looking at the same RTP stream,
>    then the loss may be identified by more than one and those =
detecting
>    the loss will all send requests for the same packet loss.  In this
>    case, the distribution source MUST filter the duplicated packet =
loss
>    request out and only forward one copy of the RTCP Feedback report
>    packet from the first Loss Detector to the group impacted by packet
>    loss.
> TOM (7) : how will the DS send only a copy of the RTCP FB report =
packet to the group impacted by packet loss. Is this done over the SSM?
> =20
> [Qin]: Sorry, maybe I sould say
> "
> Section 6.1.1
> If there are multiple Distrbitution Source with the loss detection =
support looking at the same RTP stream.....
> "
> Yes,  in the SSM use case, DS send only a copy of the RTCP Feedback =
Suppression Report packet the group via SSM.
> =20
> In your example you seem to focus only on architectures with a DS =
which also hosts the FT and the loss reporter.  There should be sections =
covering cases where the DS is disjoint from the FT, with several FTs =
and also several loss reporters disjoint from the FT and the DS.
> =20
> [Qin] I am not sure we need to address all the possible SSM use cases. =
In this draft, we just give some examples of SSM use case.
> =20
> TOM (8) : what do you mean in above text with "RTCP FB report".. is =
that the suppression message you define in this draft?
> =20
> [Qin]: Yes,  RTCP Feedback Suppression Report.
> =20
> TOM (9) : In section  "6.1.2.  Distribution Source Feedback Summary =
Model" it is not clear what your architecture looks like. You talk about =
multiple distribution sources.. whcih implies multiple SSMs.. Can you =
elaborate on that?
> =20
> [Qin]:  The scenario we are talking about in  the section 6.1.2, is to =
 allow multiple Distrbituion source between media source and the =
receiver. I will look at the this case as one special SSM case. because  =
each of distrbituion source resides at the upstream direction or =
downstream direction of other distributions souces. Does it make sense?=20=

> =20
> TOM: RFC 5760 only defines one DS for a given SSM. If you depart from =
that, you probably need to address and define this architecture in a =
better way in your draft.=20
> =20
> Tom (10) :  In section  6.3.  Multipoint Control Unit (MCU) use case,  =
typically an overlay of unicast is used... will the Feedback suppression =
message be sent over unicast as well? If so, what is gain compared to =
just having each RTCP receiver providing each their NACK to the MCU, =
even in case of a packet loss impacting all receivers?
> =20
> [Qin]: First The MCU case will not cause NACK Implosion, instead, it =
may cause FIR implosion.
>            Second, using Feedback suppression message can restrict the =
FIR message sent from receivers behind MCU and liberate the media source =
from  FIR implosion.=20
> =20
> TOM: it is not important whether this is about FIR or NACK. My point =
is that iso having each receiver sending a NACK/FIR, you will now have =
to send to each receiver a NACK/FIR  suppression in the given scenario. =
There are no differences in terms of Bandwdith overhead or server load =
in the two cases, so why bother to send suppression messages?
> =20
> =20
> Tom (11) : coming back to my previous comment on SDP description in =
this draft : if this draft defines a RTCP feedback suppression message, =
why is there need for a "announcement" in the SDP for the receivers, =
that indicates they may receive such feedback suppression messages?  If =
a receiver supports the message, it will  simply act accordingly as =
described in the draft, no need to announce that!..
> =20
> [Qin]: Okay, I get your point. I would like to put this as open issue, =
discuss it in the face to face IETF meeting.
> =20
> Regards
> Tom
> =20
> =20
> =20
> =20
> =20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>Roni, Tom,</div><div><br></div><div>I'd recommend moving the =
discussion away from specific scenarios to the extent possible, and =
focussing on the general mechanism.&nbsp;Yes, the draft should give =
examples of how retransmission suppression is used with the RAMS =
scenario, and probably with other scenarios, but those are just examples =
of how to use the general mechanism. If the semantics of that general =
mechanism are well-defined, then how it's used for the specific =
scenarios should become =
clearer.</div><div><br></div><div>Colin</div><div><br></div><div><br></div=
><br><div><div>On 19 Oct 2010, at 17:00, Roni Even =
wrote:</div><blockquote type=3D"cite"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; font-family: 'Lucida Sans =
Typewriter'; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
bgcolor=3D"#CCE8CF" lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Hi =
Tom,<o:p></o:p></span></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">In the =
current RAMS there is no way for the BRS to send a suppression =
indication , the only available way is in the main SSM stream. I think =
that such BRS will be reporting nodes to the DS in the simple case. Are =
you suggesting that we define a new SSM session that will be initiated =
by the BRS. I think this will make more sense if we also address the =
retransmission support you described since the SSM session will serve =
for the retransmission.<o:p></o:p></span></div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">Currently this part is not in scope but we may ask =
the WG if they want to add this specific =
scenario<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Roni =
Even<o:p></o:p></span></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"border-top-style: none; =
border-right-style: none; border-bottom-style: none; border-width: =
initial; border-color: initial; border-left-style: solid; =
border-left-color: blue; border-left-width: 1.5pt; padding-top: 0in; =
padding-right: 0in; padding-bottom: 0in; padding-left: 4pt; "><div><div =
style=3D"border-right-style: none; border-bottom-style: none; =
border-left-style: none; border-width: initial; border-color: initial; =
border-top-style: solid; border-top-color: rgb(181, 196, 223); =
border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; =
padding-bottom: 0in; padding-left: 0in; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">From:</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; "><span class=3D"Apple-converted-space">&nbsp;</span>Van =
Caenegem, Tom (Tom) [mailto:tom.van_caenegem@alcatel-lucent.com]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Tuesday, October 19, 2010 =
5:31 PM<br><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Roni Even; 'Qin Wu';<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">avt@ietf.org</a><br><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>RE: [AVT] New Version =
Notification for =
draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></span></div></div=
></div><div style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: =
0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; color: blue; ">Hi =
Roni,</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; ">In the typical RAMS =
architecture, the BRS will receive the SSM stream from the DS. If there =
are several BRS-es, there may be just one BRS that detects packet loss =
on its upstream link, but the others will perhaps not, as the packet =
loss took place on SSM tree branch&nbsp;that does not impact the other =
BRSes. In that case those RTP receivers associated to the other BRS-es =
in general will not be impacted and do not need to receive the =
suppression indication, while the one BRS that did sense the packet loss =
should probably best send a suppression indication (although there is a =
chance that only&nbsp;this BRS is impacted by packet loss, and not a =
single RTP receiver in the field reporting to this BRS/FT... this all =
depends on the considered topologies).</span><o:p></o:p></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; color: blue; ">As a BRS is 1-to-1 =
associated with a FT, this calls for consideration of architectures with =
(multiple) disjoint FT (and associated BRS in RAMS applications)&nbsp; =
in the suppression draft. Such disjoint FT/BRS in general cannot act as =
so-called "intermediate nodes" or "translators" because they are not on =
the direct SSM path between the DS and the RTP =
receivers.</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; ">Another aspect I want to =
highlight is that feedback suppression is closely aligned with =
retransmission. In the case described above, the one BRS that detected =
packet loss on its upstream link, may try to recover the missing packet =
(outside the scope of the draft) and retransmit over multicast to the =
receivers that report to the FT. This retransmission towards the =
receivers is also outside the current scope of the current draft, but =
the two are somehow linked.&nbsp; In DVB IPTV, we introduced dedicated =
SSMs sourced by the BRS for&nbsp;sending NACK suppression and the =
(optional) retransmission.&nbsp;In the considered scenario, having a DS =
sending a missing packet -reported by a FT downstream in the SSM tree- =
over the original SSM&nbsp;is not a good approach IMO, as it results in =
general in unnecessary retransmissions for most =
receivers.</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; ">Hope this =
clarifies.</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; =
">Tom</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div class=3D"MsoNormal" align=3D"center" =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; text-align: center; "><hr size=3D"2" width=3D"100%" =
align=3D"center"></div><p class=3D"MsoNormal" style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 12pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">From:</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; "><span class=3D"Apple-converted-space">&nbsp;</span>Roni =
Even [mailto:Even.roni@huawei.com]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>dinsdag 19 oktober 2010 =
16:46<br><b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span>Van =
Caenegem, Tom (Tom); 'Qin Wu';<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">avt@ietf.org</a><br><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>RE: [AVT] New Version =
Notification for =
draft-wu-avt-retransmission-supression-rtp-04</span><o:p></o:p></p><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Hi =
Tom,<o:p></o:p></span></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">I want to =
clarify your example since I am not sure what is special here. The BRS =
sends separate unicast streams to each participant so there is reason =
for suppression on these unicast streams. As for the primary multicast =
stream from the DS it goes to all receivers. Now if the BRS identify =
packet loss on the primary stream it should send suppression indication =
to all members of the multicast group regardless to which FT they =
report.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Can =
you clarify<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Roni =
Even<o:p></o:p></span></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"border-top-style: none; =
border-right-style: none; border-bottom-style: none; border-width: =
initial; border-color: initial; border-left-style: solid; =
border-left-color: blue; border-left-width: 1.5pt; padding-top: 0in; =
padding-right: 0in; padding-bottom: 0in; padding-left: 4pt; "><div><div =
style=3D"border-right-style: none; border-bottom-style: none; =
border-left-style: none; border-width: initial; border-color: initial; =
border-top-style: solid; border-top-color: rgb(181, 196, 223); =
border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; =
padding-bottom: 0in; padding-left: 0in; "><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">From:</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; "><span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">avt-bounces@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>[mailto:avt-bounces@ietf.org]=
<span class=3D"Apple-converted-space">&nbsp;</span><b>On Behalf Of<span =
class=3D"Apple-converted-space">&nbsp;</span></b>Van Caenegem, Tom =
(Tom)<br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Tuesday, October 19, 2010 =
10:27 AM<br><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Qin Wu;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">avt@ietf.org</a><br><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [AVT] New Version =
Notification for =
draft-wu-avt-retransmission-supression-rtp-04<o:p></o:p></span></div></div=
></div><div style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: =
0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; color: blue; ">Hi =
Qin,</span><o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; ">&nbsp;<o:p></o:p></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
color: blue; ">Thank you for your reply.</span><o:p></o:p></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; color: blue; ">Reading your =
responses,you seem to focus really on the SSM&nbsp; case with an =
architecture where the single DS (there can be only one DS per SSM) =
hosts the sole feedback target, which also hosts the loss detector =
function. My interpretation is also that in your draft the loss detector =
sends the FIR or NACK supression message to the DS, which then =
forwards/relays this&nbsp;message over the SSM to all receivers, =
right?&nbsp;</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; ">My main comment remains =
the same: you should at least incorporate the architecture of RAMS =
(where&nbsp;there&nbsp;can be several FT&nbsp; colocated with the BRS, =
which are disjoint from the DS). In this case the FT/BRS should be able =
to provide a suppression indication to all receivers that report to this =
FT (which is a subset of all receivers that connect to the SSM), which =
means one should not make use of the (original) SSM for providing the =
suppression indication.&nbsp; Your draft provides a possible solution =
for only a limited use case, and not the most interesting one, =
IMO.&nbsp;&nbsp;</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; ">Further =
down&nbsp;some&nbsp;responses to your answers =
below.</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; color: blue; =
">Tom</span><o:p></o:p></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><o:p>&nbsp;</o:p></div><div =
class=3D"MsoNormal" align=3D"center" style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; text-align: center; "><hr =
size=3D"2" width=3D"100%" align=3D"center"></div><p class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 12pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; ">From:</span></b><span style=3D"font-size: 10pt; =
font-family: Tahoma, sans-serif; "><span =
class=3D"Apple-converted-space">&nbsp;</span>Qin Wu =
[mailto:sunseawq@huawei.com]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>dinsdag 19 oktober 2010 =
5:04<br><b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span>Van =
Caenegem, Tom (Tom);<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">avt@ietf.org</a><br><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [AVT] New Version =
Notification for =
draft-wu-avt-retransmission-supression-rtp-04</span><o:p></o:p></p><div><d=
iv style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">Hi, Tom:<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">Thank for your =
comments, please see my reply inline.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">Regards!<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">-Qin<o:p></o:p></div></div><blockquote style=3D"border-top-style: =
none; border-right-style: none; border-bottom-style: none; border-width: =
initial; border-color: initial; border-left-style: solid; =
border-left-color: black; border-left-width: 1.5pt; padding-top: 0in; =
padding-right: 0in; padding-bottom: 0in; padding-left: 4pt; margin-left: =
3.75pt; margin-top: 5pt; margin-right: 0in; margin-bottom: 5pt; =
"><div><div style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: =
0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">----- Original Message -----<span style=3D"font-size: =
9pt; font-family: SimSun, serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; background-image: initial; =
background-attachment: initial; background-origin: initial; =
background-clip: initial; background-color: rgb(228, 228, 228); =
"><b>From:</b><span class=3D"Apple-converted-space">&nbsp;</span><span =
style=3D"font-size: 9pt; font-family: SimSun, serif; "><a =
href=3D"mailto:tom.van_caenegem@alcatel-lucent.com" =
title=3D"tom.van_caenegem@alcatel-lucent.com" style=3D"color: blue; =
text-decoration: underline; "><span style=3D"font-size: 12pt; =
font-family: 'Times New Roman', serif; ">Van Caenegem, Tom =
(Tom)</span></a></span><span style=3D"font-size: 9pt; font-family: =
SimSun, serif; "><o:p></o:p></span></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><span style=3D"font-size: =
9pt; font-family: SimSun, serif; "><a href=3D"mailto:avt@ietf.org" =
title=3D"avt@ietf.org" style=3D"color: blue; text-decoration: underline; =
"><span style=3D"font-size: 12pt; font-family: 'Times New Roman', serif; =
">avt@ietf.org</span></a></span><span =
class=3D"Apple-converted-space">&nbsp;</span>;<span =
class=3D"Apple-converted-space">&nbsp;</span><span style=3D"font-size: =
9pt; font-family: SimSun, serif; "><a href=3D"mailto:sunseawq@huawei.com" =
title=3D"sunseawq@huawei.com" style=3D"color: blue; text-decoration: =
underline; "><span style=3D"font-size: 12pt; font-family: 'Times New =
Roman', serif; ">Qin Wu</span></a></span><span style=3D"font-size: 9pt; =
font-family: SimSun, serif; "><o:p></o:p></span></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Tuesday, October 19, 2010 =
12:08 AM<span style=3D"font-size: 9pt; font-family: SimSun, serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [AVT] New Version =
Notification for draft-wu-avt-retransmission-supression-rtp-04<span =
style=3D"font-size: 9pt; font-family: SimSun, serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Hi =
Qin,<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">I have some comments =
(look for TOM below) on the new draft version =
4.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Thx for addressing =
them,<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">Tom<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Section 2 =
Terminology<o:p></o:p></div></div><div><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;Loss Detector:<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector is =
one logical function which is used to detect<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the packet loss at the RTP layer and =
report it to the distribution<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; source.&nbsp; The function of the loss =
reporter may be collocated with<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into the same =
entity.&nbsp; In this case, for a session<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; defined as having a distribution source =
A, on ports n for the RTP<o:p></o:p></span></pre><pre style=3D"margin-top:=
 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; "><span style=3D"font-family:=
 'Times New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; channel and =
k for the RTCP channel, the Loss Detector =
are<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; identified by an IP =
address of distribution source A on port k.<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The Loss Detector MAY also be =
implemented in one or more entities<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; different from the distribution =
source.<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; "> The function of the loss reporter may be =
collocated with<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; or integrated into =
the same entity.</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">TOM (1): ..same entity as what? i guess you mean =
the DS?</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">[Qin]: Your are right.&nbsp; In this draft, the =
loss detector is just functionality of DS.&nbsp; Furthermore we only =
consider the loss detector as part of feedback target within the =
distribution source.</span><o:p></o:p></pre><pre style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; "><span style=3D"font-family:=
 'Times New Roman', serif; ">Also this is just one example use case to =
explain how feedback suppression works in SSM use =
case.</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">TOM (2): I do not understand how you associate the =
port k to a loss detector.. what do you mean with =
that?</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">[Qin]:&nbsp; Sorry for ambiguity of the description =
here. It actually means that the loss reporter is part of feedback =
target and share the same address and port.</span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">section =
3</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;</span><o:p></o:p></pre><pre style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; "><span style=3D"font-family:=
 'Times New Roman', serif; ">(..)</span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">In order to detect =
packet loss before the receivers perceive it, =
one<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp; or more intermediate nodes are placed =
between the media source and<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; receiver =
(e.g., Distribution server, MCU, RTP translator).&nbsp; =
These<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; intermediaries monitor for packet loss =
upstream of themselves by<o:p></o:p></span></pre><pre style=3D"margin-top:=
 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; "><span style=3D"font-family:=
 'Times New Roman', serif; ">&nbsp;&nbsp; checking RTP sequence numbers, =
just as receivers do.&nbsp; Upon detecting<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; (or =
suspecting) an upstream loss, the intermediary may send =
Feedback<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; Suppression message towards the =
receivers as defined in this<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; =
specification.<o:p></o:p></span></pre></div><div><div style=3D"margin-top:=
 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">TOM (3) =
:&nbsp;...based on the two above definitions =
...is&nbsp;an&nbsp;intermediate&nbsp;node&nbsp;hence&nbsp;equal&nbsp;to&nb=
sp;a&nbsp;"Loss&nbsp;detector"?&nbsp;<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">+ How does a Loss detector differ from an RTP =
receiver?<o:p></o:p></div></div></blockquote><blockquote =
style=3D"border-top-style: none; border-right-style: none; =
border-bottom-style: none; border-width: initial; border-color: initial; =
border-left-style: solid; border-left-color: black; border-left-width: =
1.5pt; padding-top: 0in; padding-right: 0in; padding-bottom: 0in; =
padding-left: 4pt; margin-left: 3.75pt; margin-top: 5pt; margin-right: =
0in; margin-bottom: 5pt; "><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">[Qin]: The loss detector =
is just one functionality of Distribution Source in the SSM use =
case.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Maybe we =
should&nbsp;interpret it as "Loss detection functionality". How the =
packet loss is detected is out of scope of this =
document.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Different from dedicated =
RTP receiver, the intermediate node does not depend on existing NACK =
message for Retransmission request for packet =
loss.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">the intermediate node can =
create message by itself.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">Section =
6.1<o:p></o:p></div></div><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">In order to avoid the forms of NACK implosion described =
in section 1,<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; the Loss Detector is introduced.&nbsp; =
The Loss Detector detects and<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; reports =
packet loss, on both the upstream link and the =
downstream<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; aggregate link.&nbsp; How the loss =
detector SHOULD detect the packet loss<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; is beyond =
of scope of this document.&nbsp; When upstream link =
or<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp; downstream aggregate link packet loss =
occurs, the Loss Detector MAY<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; inform =
distribution source of the detected packet loss using =
Feedback<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; Suppression messages.&nbsp; In =
response, the distribution source either<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; forwards =
packet loss suppression report received from Loss =
Detector<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; or creates a Feedback Suppression =
report and sends it to all the RTP<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; =
receivers, over the multicast channel.&nbsp; This loss suppression =
report<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; tells the receivers that the lost =
packet will either be forthcoming<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; from =
distribution source, or it irretrievably lost such that there =
is<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp; nothing to be gained by the receiver =
sending a NACK to the media<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; =
source.&nbsp; The distribution source then can (optionally) ask for =
the<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp; lost packets from the media source on =
behalf of all the RTP<o:p></o:p></span></pre><pre style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; "><span style=3D"font-family:=
 'Times New Roman', serif; ">&nbsp;&nbsp; =
receivers.</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">TOM (4) : You probably do not want to have =
normative language in section 6.1, like "How the loss detector SHOULD =
(..)". </span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">[Qin]:Sorry, I can't get it.</span><span =
style=3D"font-family: Arial, sans-serif; color: blue; =
">&nbsp;</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">TOM: this is really a detail. If you want to =
keep the sentence, you should e.g. say: "How the loss detector can =
detect the packet loss...</span><o:p></o:p></pre><pre style=3D"margin-top:=
 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 10pt; font-family: 'Courier New'; =
">&nbsp;<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">&nbsp;</span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">TOM (5) : It is pretty =
clear how a Loss detector will detect packet loss on its upstream link, =
I guess.</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">[Qin]: As I said above,&nbsp; Loss detector is just =
functionality of some Distribution sources.&nbsp; Also what it said here =
in section 6.1 </span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">is just one example use =
case.</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">TOM (6):&nbsp; What is the rationale for a loss =
detector to send a Feedback suppression message to the DS.. why NOT a =
RTCP NACK FB?</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">[Qin]: As I explained in the last IETF =
presentaion,&nbsp; this cause NACK scematics mismatch. That's why many =
people in the last IETF meeeting suggest to create new RTCP message =
during presentation.</span><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">&nbsp;</span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; =
">&nbsp;<o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">TOM: As your assumption is that the loss =
reporter is part of the FT that is part of the DS, then there is even no =
need to specify how the loss detector </span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: Arial, sans-serif; color: blue; ">reports the =
packet loss.&nbsp; In the more generic case, it makes more sense to me =
that the loss reporter just uses a NACK (or a FIR) if it detects packet =
loss, sent towards its FT or to the DS if this one hosts the FT that the =
loss detector reports to . It is then up to the FT or the DS to decide =
whether this message triggers a NACK or FIR suppression =
</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">indication to (s subset of) the =
receivers.&nbsp; This suppression indication does not necessarly require =
a new message format. A NACK/FIR&nbsp; that is sent to a RTP receiver =
can be defined as a feedback suppression indication. I do not see a good =
use case why NACKs from&nbsp; receivers&nbsp; would be reflected back in =
e.g. SSM use case by DS or FT&nbsp; to all the connected receivers. =
</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: Arial, =
sans-serif; color: blue; ">If we define that a NACK/FIR&nbsp; for a SSM =
may only be sent TOWARDS a RTP receiver, if this is for suppression =
indication, there is no ambiguity possibly.&nbsp;&nbsp; =
</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; ">&nbsp;<o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: Arial, sans-serif; color: blue; =
">&nbsp;</span><o:p></o:p></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">Section 6.1.1</span><o:p></o:p></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">If there are multiple =
Loss Detectors looking at the same RTP =
stream,<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; then the loss may be identified by =
more than one and those detecting<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; the loss =
will all send requests for the same packet loss.&nbsp; In =
this<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10pt; =
font-family: 'Courier New'; "><span style=3D"font-family: 'Times New =
Roman', serif; ">&nbsp;&nbsp; case, the distribution source MUST filter =
the duplicated packet loss<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; request =
out and only forward one copy of the RTCP Feedback =
report<o:p></o:p></span></pre><pre style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
10pt; font-family: 'Courier New'; "><span style=3D"font-family: 'Times =
New Roman', serif; ">&nbsp;&nbsp; packet from the first Loss Detector to =
the group impacted by packet<o:p></o:p></span></pre><pre =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10pt; font-family: 'Courier New'; "><span =
style=3D"font-family: 'Times New Roman', serif; ">&nbsp;&nbsp; =
loss.<o:p></o:p></span></pre><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">TOM (7) : how will the DS =
send only a copy of the RTCP FB report packet to the group impacted by =
packet loss. Is this done over the SSM?<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">[Qin]: Sorry, =
maybe I sould say<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">"<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Section =
6.1.1<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">If there are multiple =
Distrbitution Source with the loss detection support looking at the same =
RTP stream.....<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">"<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Yes,&nbsp; in the SSM use =
case, DS send only a copy of the RTCP Feedback Suppression Report packet =
the group via SSM.<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">In your example you seem =
to focus only on architectures with a DS which also hosts the FT and the =
loss reporter.&nbsp; There should be sections covering cases where the =
DS is disjoint from the FT, with several FTs and also several loss =
reporters disjoint from the FT and the =
DS.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">[Qin] I am not sure we =
need to address all the possible SSM use cases. In this draft, we just =
give&nbsp;some examples of SSM use case.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">TOM (8) : what =
do you mean in above text with "RTCP FB report".. is that the =
suppression message you define in this =
draft?<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">[Qin]: Yes,&nbsp; RTCP =
Feedback Suppression Report.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">TOM (9) : In =
section&nbsp; "6.1.2.&nbsp; Distribution Source Feedback Summary Model" =
it is not clear what your architecture looks like. You talk about =
multiple distribution sources.. whcih implies multiple SSMs.. Can you =
elaborate on that?<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">[Qin]:&nbsp; The scenario =
we are talking about in&nbsp; the section 6.1.2, is&nbsp;to &nbsp;allow =
multiple Distrbituion source between media source and the receiver. I =
will look at the this case as one special SSM case. because&nbsp; each =
of distrbituion source resides at the upstream direction or downstream =
direction of other distributions souces. Does it make sense?<span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; color: blue; =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; color: blue; ">TOM: RFC 5760 only =
defines one DS for a given SSM. If you depart from that, you probably =
need to address and define&nbsp;this architecture in a better way in =
your draft.</span>&nbsp;<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">Tom (10) =
:&nbsp; In section&nbsp; 6.3.&nbsp; Multipoint Control Unit (MCU) use =
case,&nbsp; typically an overlay of unicast is used... will the Feedback =
suppression message be sent over unicast as well? If so, what is gain =
compared to just having each RTCP receiver providing each their NACK to =
the MCU, even in case of a packet loss impacting all =
receivers?<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">[Qin]: First The MCU case =
will not cause NACK Implosion, instead, it may cause FIR =
implosion.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Second, =
using Feedback suppression message can restrict the FIR message sent =
from receivers&nbsp;behind MCU and liberate the media source from&nbsp; =
FIR implosion.<span style=3D"font-size: 10pt; font-family: Arial, =
sans-serif; color: blue; ">&nbsp;</span><o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; color: blue; =
">TOM: it is not important whether this is about FIR or NACK. My point =
is that iso having each receiver sending a NACK/FIR, you will now have =
to send to each receiver a NACK/FIR&nbsp; suppression in the given =
scenario. There are no differences in terms of Bandwdith overhead or =
server load in the two cases, so why bother to send suppression =
messages?</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Tom (11) : coming back to =
my previous comment on SDP description in this draft&nbsp;:&nbsp;if this =
draft defines a RTCP feedback suppression message, why is there need for =
a "announcement" in the SDP for the receivers, that indicates they may =
receive such feedback suppression messages?&nbsp; If a receiver supports =
the message, it will&nbsp; simply&nbsp;act accordingly as described in =
the draft, no need to announce that!..<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">[Qin]: Okay, I =
get your point. I would like to put this as open issue, discuss it in =
the face to face IETF meeting.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">Regards<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">Tom<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div></blockquote></div></div></div>____________=
___________________________________<br>Audio/Video Transport Working =
Group<br><a href=3D"mailto:avt@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">avt@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/avt" style=3D"color: blue; =
text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/avt</a><br></div></span></blockquo=
te></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></body></html>=

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Date: Wed, 20 Oct 2010 10:27:03 +0800
From: Qin Wu <sunseawq@huawei.com>
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Tom,
 Look at the new version of draft, you will find we focus on defining more general mechanism of "feedback suppression" to address different use cases of "feedback implosion".

SSM case is just one example use case we look at. We can address the case you suggested below(which assumes a single DS, one or multiple disjoint FTs, where a FT is co-positioned with a BRS (same IP address))
But I am not sure this case is referred to as "RAMS case". And also what confuse to me in the RAMS case is Retransmission Server is usually used for channel switching in the RAMS case and just used to send unicast RTP stream to one receiver when this receiver requests a unicast burst from Retransmission server. Suppose there is two Retransmission server (i.e., RS1 and RS2 )behind the DS,if the packet loss happens between the DS and  RS1 but there is no packet loss happened between DS and RS2, do you think RS1 should report the packet loss to its upstream DS.
This is not allowed in our draft.  In my understand, maybe there is only one receiver requesting unicast burst, it is not necessary to suppress such RAMS message for one receiver.

In our draft, one similar case is we allow multiple DSs in the SSM case, in such case, packet loss always happens at the upsteam of each DS, each DS should not report packet loss to the upstream DS, in this way, even DS is connected to a different branch of the SSM tree, it is not necessary to send feedback suppresion to all the receiver if some didn't suffer from packet loss.

If there is one downstream DS or intermediate node placed at the downstream of its upstream DS, the downstream may either simply forward the Feedback Suppression message received from upstream, or augment (or replace) it with a  feedback suppression message that reflects the loss pattern they have themselves seen. they should not initiate their own  additional feedback suppression messages for the same packet sequence numbers.  The role of such downstream DS has been clearly defined in the section 3 of draft-wu-avt-retransmission-suppression-rtp-04 as follows:
"
   RTCP Feedback Suppression follows the same semantic model as RTCP
   NACK - it conveys the packet receipt/loss events at the sequence
   number level and considers missing packets as unrepaired.  But unlike
   RTCP NACK, the Feedback Suppression messages can be generated at
   intermediate nodes who are not RTP receivers and sent to the
   corresponding receivers.  Intermediaries downstream of an
   intermediary detecting loss obviously SHOULD NOT initiate their own
   additional feedback suppression messages for the same packet sequence
   numbers.  They may either simply forward the Feedback Suppression
   message received from upstream, or augment (or replace) it with a
   feedback suppression message that reflects the loss pattern they have
   themselves seen.
"

As for why we creat new message instead of using existing NACK, 
NACK lacks scemantics for suppression and NACK as one feedback message terminate at the server, the intermediate node e.g., distribution
source as the server receiving NACK will confuse whether the NACK should be forwarded to the server or the receiver. And the distribution source
has no behavior to be defined for dealing with NACKs.
Also Feedback Suppression message can work without receiving NACKs sent from receiver. Using NACK for packet loss report from some dedicated receiver is
just a way for the distribution source or intermediate node to detect packet loss. But it is not the only way.


Regards!
-Qin
  ----- Original Message ----- 
  From: Van Caenegem, Tom (Tom) 
  To: Qin Wu ; avt@ietf.org 
  Sent: Tuesday, October 19, 2010 9:26 PM
  Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04


  Qin,

  focusing on just SSM arch: I quote from your earlier response:

  In this draft, the loss detector is just functionality of DS.  Furthermore we only consider the loss detector as part of feedback target within the distribution source. Also this is just one example use case to explain how feedback suppression works in SSM use case.

  Again, your draft does explain in detail how feedback suppression works for a SSM as described here above, but does not address the RAMS case (which assumes a single DS, one or multiple disjoint FTs, where a FT is co-positioned with a BRS (same IP address). 

  Just consider a DS with multiple disjoint FTs (as in RAMS, and also permitted by RFC 5760). Typically each FT is connected to a different branch of the SSM tree. 

  Currently your draft imposes that the feedback suppression message is sent by the DS over the SSM. This is not a good solution as you either restrict the solution to packet losses that are common to all SSM RTP receivers (covering packet loss between media source and DS ) or you will send suppression indications (and optionally forthcoming retransmission) to all receivers even if some did not suffer from packet loss. Therefore the draft should also define how suppression information is sent to (a subset of) the receivers. It does not make sense to me that IETF would define FB suppression for a architecture that does not take into account deployed SSM architectures for IPTV.

  Even within the SSM architecture you are focusing on, I am very much confused: you say there is one DS that hosts the FT and loss reporting function. What is then the role of the intermediate sources downstream of the DS? I really would appreciate you illustrate with some more architecture drawings.

  The other point I still am not convinced of, is  the use of NACK versus dedicated suppression message. There should be more clarity in the different architectures you consider before we can make any statements on whether having a NACK or FIR  that is used as suppression message towards the SSM receivers, is confusing or not. I do not disagree that NACK as defined in RFC 4588 cannot be used as suppression message. But if we define it this way in a this "suppression"  draft/RFC, and no confusions can occur, I do not see the issue, especially as the syntax of the suppression message is completely the same as a NACK/FIR. 

  Regards
  Tom







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<DIV><STRONG>Tom,</STRONG></DIV>
<DIV><STRONG>&nbsp;Look at the new version of draft, you will find we focus on 
defining more general mechanism of "feedback suppression"&nbsp;to address 
different use cases of "feedback implosion".</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>SSM case is just one example use case we look at. We can 
address&nbsp;the case you suggested below(</STRONG><FONT 
color=#0000ff><STRONG>which assumes a single DS, one or multiple disjoint FTs, 
where&nbsp;a FT is co-positioned with a BRS (same IP 
address</STRONG>)</FONT><STRONG>)</STRONG></DIV>
<DIV><STRONG>But I am not sure this case is referred to as "RAMS case".&nbsp;And 
also what confuse to me in the RAMS case is Retransmission Server is usually 
used for channel switching in the RAMS case and just used to send unicast 
</STRONG><STRONG>RTP stream to one receiver when&nbsp;this receiver requests a 
unicast burst from Retransmission server. Suppose there is two Retransmission 
server (i.e., RS1 and RS2 )behind the DS,if the packet loss happens&nbsp;between 
the DS and&nbsp;&nbsp;RS1&nbsp;but there is no packet loss happened between DS 
and RS2, do you think RS1 should report the packet loss to its upstream 
DS.</STRONG></DIV>
<DIV><STRONG>This is not allowed in our draft.&nbsp; </STRONG><STRONG>In my 
understand, maybe there is only one receiver requesting unicast burst, it is not 
necessary to suppress such RAMS message for one receiver.</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>In our draft,&nbsp;one similar case is we allow multiple DSs in the 
SSM case, in such case, packet loss always happens at the upsteam of each DS, 
each DS should not report packet loss to the upstream DS, in this way, even 
</STRONG><STRONG>DS is connected to a different branch of the SSM tree, it is 
not necessary to send feedback suppresion to all the receiver if some didn't 
suffer from packet loss.</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>If there is one downstream DS or intermediate node placed at the 
downstream of&nbsp;its upstream DS, the downstream may either simply forward the 
Feedback Suppression&nbsp;message received from upstream, or augment (or 
replace) it with a&nbsp; feedback suppression message that reflects the loss 
pattern they have&nbsp;themselves seen. they should not initiate their own&nbsp; 
additional feedback suppression messages for the same packet 
sequence&nbsp;numbers.&nbsp; The role of such downstream DS has been clearly 
defined in the section 3 of draft-wu-avt-retransmission-suppression-rtp-04 as 
follows:</STRONG></DIV>
<DIV><STRONG>"</STRONG></DIV>
<DIV>&nbsp;&nbsp; RTCP Feedback Suppression follows the same semantic model as 
RTCP<BR>&nbsp;&nbsp; NACK - it conveys the packet receipt/loss events at the 
sequence<BR>&nbsp;&nbsp; number level and considers missing packets as 
unrepaired.&nbsp; But unlike<BR>&nbsp;&nbsp; RTCP NACK, the Feedback Suppression 
messages can be generated at<BR>&nbsp;&nbsp; intermediate nodes who are not RTP 
receivers and sent to the<BR>&nbsp;&nbsp; corresponding receivers.&nbsp; 
Intermediaries downstream of an<BR>&nbsp;&nbsp; intermediary detecting loss 
obviously SHOULD NOT initiate their own<BR>&nbsp;&nbsp; additional feedback 
suppression messages for the same packet sequence<BR>&nbsp;&nbsp; numbers.&nbsp; 
They may either simply forward the Feedback Suppression<BR>&nbsp;&nbsp; message 
received from upstream, or augment (or replace) it with a<BR>&nbsp;&nbsp; 
feedback suppression message that reflects the loss pattern they 
have<BR>&nbsp;&nbsp; themselves seen.</DIV>
<DIV><STRONG>"</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>As for why we creat new message instead of using existing NACK, 
</STRONG></DIV>
<DIV><STRONG>NACK lacks scemantics for suppression and NACK as one feedback 
message terminate at the server, the intermediate node e.g., 
distribution<BR>source as the server receiving NACK will confuse whether the 
NACK should be forwarded to the server or the receiver. And the distribution 
source</STRONG></DIV>
<DIV><STRONG>has no behavior to be defined for dealing with 
NACKs.</STRONG></DIV>
<DIV><STRONG>Also Feedback Suppression message can work without receiving NACKs 
sent from receiver. Using NACK for packet loss report from some dedicated 
receiver is</STRONG></DIV>
<DIV><STRONG>just a way for the distribution source or intermediate node to 
detect packet loss. But it is not the only way.</STRONG><BR></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>Regards!</STRONG></DIV>
<DIV><STRONG>-Qin</STRONG></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px" 
dir=ltr>
  <DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B> 
  <A title=tom.van_caenegem@alcatel-lucent.com 
  href="mailto:tom.van_caenegem@alcatel-lucent.com">Van Caenegem, Tom (Tom)</A> 
  </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>To:</B> <A title=sunseawq@huawei.com 
  href="mailto:sunseawq@huawei.com">Qin Wu</A> ; <A title=avt@ietf.org 
  href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Sent:</B> Tuesday, October 19, 2010 9:26 PM</DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Subject:</B> RE: [AVT] New Version Notification 
  for draft-wu-avt-retransmission-supression-rtp-04</DIV>
  <DIV><BR></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><FONT color=#0000ff 
  size=2 face=Arial>Qin,</FONT></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><FONT color=#0000ff 
  size=2 face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><FONT color=#0000ff 
  size=2 face=Arial>focusing on just SSM arch: I quote from your earlier 
  response:</FONT></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><FONT color=#0000ff 
  size=2 face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010>In this draft, the loss 
  detector is just functionality of DS.&nbsp; Furthermore we only consider the 
  loss detector as part of feedback target within the distribution source. <SPAN 
  class=237051013-18102010><FONT face="Times New Roman">Also this is just one 
  example use case to explain how feedback suppression works in SSM use 
  case.</FONT></SPAN></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 face=Arial>Again, your 
  draft does explain in detail how feedback suppression&nbsp;works for a SSM as 
  described here above, but does not address the RAMS case (which assumes a 
  single DS, one or multiple disjoint FTs, where&nbsp;a FT is co-positioned with 
  a BRS (same IP address). </FONT></SPAN></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 face=Arial>Just consider a 
  DS with multiple disjoint&nbsp;FTs (as in RAMS, and also permitted by RFC 
  5760). Typically each&nbsp;FT is connected to a different branch of the SSM 
  tree. </FONT></SPAN></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><SPAN class=764462011-19102010><SPAN 
  class=237051013-18102010><FONT color=#0000ff size=2 
  face=Arial>Currently&nbsp;your draft imposes that the feedback suppression 
  message is sent by the DS over the SSM. This is not a good solution as you 
  either restrict the solution to packet losses that are common to all SSM RTP 
  receivers (covering packet loss between media source and DS ) or you will send 
  suppression indications (and optionally forthcoming retransmission) to all 
  receivers even if some did not suffer from packet 
  loss.&nbsp;Therefore&nbsp;the draft should also&nbsp;define how suppression 
  information is sent to (a subset of) the receivers. It does not make sense to 
  me that IETF would define FB suppression for a architecture that does not 
  take&nbsp;into account deployed SSM architectures for 
  IPTV.</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial>Even within the SSM architecture you are 
  focusing on, I am very much confused: you say there is one DS that hosts the 
  FT and loss reporting function. What is then the role of the intermediate 
  sources downstream of the DS? I really would appreciate you illustrate with 
  some more architecture drawings.</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial>The other point I still am not convinced of, 
  is &nbsp;the use of NACK versus dedicated suppression message.&nbsp;There 
  should be more clarity in the different architectures you consider before we 
  can make any statements on whether having a NACK or FIR&nbsp;&nbsp;that 
  is&nbsp;used as suppression message towards the SSM receivers, is confusing or 
  not. I do not disagree that NACK as defined in RFC 4588 cannot be used as 
  suppression message. But if we define it this way in a&nbsp;this "suppression" 
  &nbsp;draft/RFC, and no confusions can occur, I do not see the issue, 
  especially as the syntax of the suppression message is completely the same as 
  a NACK/FIR. </FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial>Regards</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial>Tom</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
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  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=764462011-19102010><SPAN class=237051013-18102010><FONT 
  color=#0000ff size=2 face=Arial></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV dir=ltr align=left><BR></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left>
  <HR tabIndex=-1>
  </DIV></BLOCKQUOTE></BODY></HTML>

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Date: Wed, 20 Oct 2010 11:46:08 +0800
From: Qin Wu <sunseawq@huawei.com>
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Cc: VAN CAENEGEM Tom <Tom.Van_Caenegem@alcatel-lucent.com>, 'Roni Even' <Even.roni@huawei.com>
Subject: [AVT] Fw: New Version Notification for draft-wu-avt-retransmission-supression-rtp-04
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Tom:
I get your concern now. The RAMS architecure in your mind is one media source-> one distribution source -> multiple BRS in parallel-> the recievers behind each BRS.
In this scenario one distribution source is placed between media source and BRS for relaying SSM stream from media source. The BRS will receive the SSM stream from the DS. Suppose there are several BRSes behind the distribution source or media source, there may be just one BRS that detects packet loss on its upstream link between the distribution source and BRS, but the others will perhaps not, as the packet loss took place on SSM tree branch that does not impact the other BRSes. 

In this case, event the distribution source with loss detection functionality support can not detect packet loss at the downstream of itself, therefore the distribution source SHOULD NOT create new Feedback Suppression message and send it to all the BRS. In order to address this case, If BRS has loss detection support, the BRS MAY choose to create new  Feedback Suppression message and send it to the receivers behind this BRS.

In this way, the packet loss took place on SSM tree branch will not impact other BRSes.
Hope this clarifies.

BTW: I can add this use case to address your concern in the new version.

Regards!
-Qin
  ----- Original Message ----- 
  From: Van Caenegem, Tom (Tom) 
  To: Roni Even ; 'Qin Wu' ; avt@ietf.org 
  Sent: Tuesday, October 19, 2010 11:30 PM
  Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04


  Hi Roni,

  In the typical RAMS architecture, the BRS will receive the SSM stream from the DS. If there are several BRS-es, there may be just one BRS that detects packet loss on its upstream link, but the others will perhaps not, as the packet loss took place on SSM tree branch that does not impact the other BRSes. In that case those RTP receivers associated to the other BRS-es in general will not be impacted and do not need to receive the suppression indication, while the one BRS that did sense the packet loss should probably best send a suppression indication (although there is a chance that only this BRS is impacted by packet loss, and not a single RTP receiver in the field reporting to this BRS/FT... this all depends on the considered topologies). 

  As a BRS is 1-to-1 associated with a FT, this calls for consideration of architectures with (multiple) disjoint FT (and associated BRS in RAMS applications)  in the suppression draft. Such disjoint FT/BRS in general cannot act as so-called "intermediate nodes" or "translators" because they are not on the direct SSM path between the DS and the RTP receivers. 

  Another aspect I want to highlight is that feedback suppression is closely aligned with retransmission. In the case described above, the one BRS that detected packet loss on its upstream link, may try to recover the missing packet (outside the scope of the draft) and retransmit over multicast to the receivers that report to the FT. This retransmission towards the receivers is also outside the current scope of the current draft, but the two are somehow linked.  In DVB IPTV, we introduced dedicated SSMs sourced by the BRS for sending NACK suppression and the (optional) retransmission. In the considered scenario, having a DS sending a missing packet -reported by a FT downstream in the SSM tree- over the original SSM is not a good approach IMO, as it results in general in unnecessary retransmissions for most receivers.

  Hope this clarifies.

  Tom


------------------------------------------------------------------------------

  From: Roni Even [mailto:Even.roni@huawei.com] 
  Sent: dinsdag 19 oktober 2010 16:46
  To: Van Caenegem, Tom (Tom); 'Qin Wu'; avt@ietf.org
  Subject: RE: [AVT] New Version Notification for draft-wu-avt-retransmission-supression-rtp-04


  Hi Tom,

  I want to clarify your example since I am not sure what is special here. The BRS sends separate unicast streams to each participant so there is reason for suppression on these unicast streams. As for the primary multicast stream from the DS it goes to all receivers. Now if the BRS identify packet loss on the primary stream it should send suppression indication to all members of the multicast group regardless to which FT they report.

  Can you clarify

   

  Roni Even

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<BODY lang=EN-US link=blue bgColor=#cce8cf vLink=purple>
<DIV>Tom:</DIV>
<DIV>I get your concern now. The RAMS architecure in your mind is one media 
source-&gt; one distribution source -&gt; multiple BRS in parallel-&gt; the 
recievers behind each BRS.</DIV>
<DIV>In this scenario one distribution source is placed between media source and 
BRS for relaying SSM stream from media source. The BRS will receive the SSM 
stream from the DS. Suppose there are several BRSes behind the distribution 
source or media source, there may be just one BRS that detects packet loss on 
its upstream link between the distribution source and BRS, but the others will 
perhaps not, as the packet loss took place on SSM tree branch that does not 
impact the other BRSes. </DIV>
<DIV>&nbsp;</DIV>
<DIV>In this case,&nbsp;event the distribution source with loss detection 
functionality support can not detect packet loss at the downstream of itself, 
therefore the distribution source SHOULD NOT create new Feedback Suppression 
message and send it to all the BRS. In order to address this case, If BRS has 
loss detection support, the BRS&nbsp;MAY choose to create 
new&nbsp;&nbsp;Feedback Suppression message and send it to the receivers behind 
this BRS.</DIV>
<DIV><FONT size=2 face=&#23435;&#20307;></FONT>&nbsp;</DIV>
<DIV>In this way, the packet loss took place on SSM tree branch will not impact 
other BRSes.</DIV>
<DIV>Hope this clarifies.</DIV>
<DIV><FONT size=2 face=&#23435;&#20307;></FONT>&nbsp;</DIV>
<DIV>BTW: I can add this use case to address your concern in the new 
version.</DIV>
<DIV><FONT size=2 face=&#23435;&#20307;></FONT>&nbsp;</DIV>
<DIV>Regards!</DIV>
<DIV>-Qin</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px" 
dir=ltr>
  <DIV style="FONT: 9pt &#23435;&#20307;">----- Original Message ----- </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B> 
  <A title=tom.van_caenegem@alcatel-lucent.com 
  href="mailto:tom.van_caenegem@alcatel-lucent.com">Van Caenegem, Tom (Tom)</A> 
  </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>To:</B> <A title=Even.roni@huawei.com 
  href="mailto:Even.roni@huawei.com">Roni Even</A> ; <A 
  title=sunseawq@huawei.com href="mailto:sunseawq@huawei.com">'Qin Wu'</A> ; <A 
  title=avt@ietf.org href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Sent:</B> Tuesday, October 19, 2010 11:30 
PM</DIV>
  <DIV style="FONT: 9pt &#23435;&#20307;"><B>Subject:</B> RE: [AVT] New Version Notification 
  for draft-wu-avt-retransmission-supression-rtp-04</DIV>
  <DIV><BR></DIV>
  <DIV dir=ltr align=left><SPAN class=783445214-19102010><FONT color=#0000ff 
  size=2 face=Arial>Hi Roni,</FONT></SPAN></DIV>
  <DIV dir=ltr align=left><SPAN class=783445214-19102010><FONT color=#0000ff 
  size=2 face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 face=Arial>In the typical 
  RAMS architecture, the BRS will receive the SSM stream from the DS. If there 
  are several BRS-es, there may be just one BRS that detects packet loss on its 
  upstream link, but the others will perhaps not, as the packet loss took place 
  on SSM tree branch&nbsp;that does not impact the other BRSes. In that case 
  those RTP receivers associated to the other BRS-es in general will not be 
  impacted and do not need to receive the suppression indication, while the one 
  BRS that did sense the packet loss should probably best send a suppression 
  indication (although there is a chance that only&nbsp;this BRS is impacted by 
  packet loss, and not a single RTP receiver in the field reporting to this 
  BRS/FT... this all depends on the considered topologies). </FONT></SPAN></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 face=Arial>As a BRS is 
  1-to-1 associated with a FT, this calls for consideration of architectures 
  with (multiple) disjoint FT (and associated BRS in RAMS applications)&nbsp; in 
  the suppression draft. Such disjoint FT/BRS in general cannot act as so-called 
  "intermediate nodes" or "translators" because they are not on the direct SSM 
  path between the DS and the RTP receivers. </FONT></SPAN></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 face=Arial>Another aspect 
  I want to highlight is that feedback suppression is closely aligned with 
  retransmission. In the case described above, the one BRS that detected packet 
  loss on its upstream link, may try to recover the missing packet (outside the 
  scope of the draft) and retransmit over multicast to the receivers that report 
  to the FT. This retransmission towards the receivers is also outside the 
  current scope of the current draft, but the two are somehow linked.&nbsp; In 
  DVB IPTV, we introduced dedicated SSMs sourced by the BRS for&nbsp;sending 
  NACK suppression and the (optional) retransmission.&nbsp;In the considered 
  scenario, having a DS sending a missing packet -reported by a FT downstream in 
  the SSM tree- over the original SSM&nbsp;is not a good approach IMO, as it 
  results in general in unnecessary retransmissions for most 
  receivers.</FONT></SPAN></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 face=Arial>Hope this 
  clarifies.</FONT></SPAN></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 
  face=Arial></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010><FONT color=#0000ff size=2 
  face=Arial>Tom</FONT></SPAN></DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><SPAN 
  class=783445214-19102010></SPAN>&nbsp;</DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left>
  <HR tabIndex=-1>
  </DIV>
  <DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left><FONT size=2 
  face=Tahoma><B>From:</B> Roni Even [mailto:Even.roni@huawei.com] 
  <BR><B>Sent:</B> dinsdag 19 oktober 2010 16:46<BR><B>To:</B> Van Caenegem, Tom 
  (Tom); 'Qin Wu'; avt@ietf.org<BR><B>Subject:</B> RE: [AVT] New Version 
  Notification for 
  draft-wu-avt-retransmission-supression-rtp-04<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV class=WordSection1>
  <P class=MsoNormal><SPAN 
  style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt">Hi 
  Tom,<o:p></o:p></SPAN></P>
  <P class=MsoNormal><SPAN 
  style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt">I 
  want to clarify your example since I am not sure what is special here. The BRS 
  sends separate unicast streams to each participant so there is reason for 
  suppression on these unicast streams. As for the primary multicast stream from 
  the DS it goes to all receivers. Now if the BRS identify packet loss on the 
  primary stream it should send suppression indication to all members of the 
  multicast group regardless to which FT they report.<o:p></o:p></SPAN></P>
  <P class=MsoNormal><SPAN 
  style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt">Can 
  you clarify<o:p></o:p></SPAN></P>
  <P class=MsoNormal><SPAN 
  style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;</o:p></SPAN></P>
  <P class=MsoNormal><SPAN 
  style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt">Roni 
  Even</SPAN></P></DIV></BLOCKQUOTE></BODY></HTML>

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Hi, folks:
The new version of draft-wu-avt-retransmission-supression-rtp is available at:
 http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-05.txt
We tried to address the comments regarding the example on how Feedback Suppression mechanism is used 
with the RAMS scenario in the new section 6.2 "Unicast based Rapid Acquisition of Multicast Stream (RAMS) use case".

To be clear, this mechanism is used as a suppression indication from the server (or  intermediary) to the client. It tells RTP receivers 
explicitly that feedback for a particular packet loss is  not needed before the RTP receiver reacts to the loss and invokes 
its packet loss repair machinery. This mechanism can works for all the Feedback implosion use case.

Regards!
-Qin
----- Original Message ----- 
From: <Internet-Drafts@ietf.org>
To: <i-d-announce@ietf.org>
Sent: Wednesday, October 20, 2010 12:45 PM
Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-05.txt


>A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> Title           : RTCP Report Extension for Feedback Suppression
> Author(s)       : Q. Wu, et al.
> Filename        : draft-wu-avt-retransmission-supression-rtp-05.txt
> Pages           : 22
> Date            : 2010-10-19
> 
> When packet loss close to the media source or intermediary of the
> session is detected as a loss by a large number of receivers, large
> number of feedback requests used to ask for the lost RTP packets are
> all addressed to the same media source, or a designated feedback
> target.  This may result in a phenomenon known variously as a
> "feedback storm " or "feedback implosion ".
> 
> This document specifies an extension to the RTCP feedback messages
> defined in the Audio-Visual Profile with Feedback (AVPF).  This
> extension allows an intermediate node in the network (such as an RTP
> translator) inform downstream receivers that packet loss was detected
> and sending a feedback message concerning a specified set of RTP
> packets may be unnecessary, or even harmful.  Receivers respond to
> receipt of a feedback suppression message by not sending a feedback
> message (e.g. a NACK) that they otherwise would, This in turn reduces
> load on both the feedback target and on the network.  The proposed
> extension is useful for single-source multicast sessions.  In
> addition, it can be applied to any other types of RTP sessions and
> topologies that might benefit from feedback suppression mechanism.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-05.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
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From paromitaz@gmail.com  Wed Oct 20 01:24:26 2010
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From: paromita chowdhury <paromitaz@gmail.com>
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Subject: [AVT] SSRC change in a single RTP session
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Hi,

I have read the RTP rfc 3550 a couple of times, still have a doubt on the
following issue:

- If a RTP unicast session with no change in the source and destination
address, changes its SSRC during the call session is it permissible? In
other words, between the same pair of source and destination addresses, can
a change in the SSRC value acceptable at the destination?

I request the experts in this field to respond to this query?

Thanks
PC

--00163649a2f594f4de0493082884
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Hi,<br><br>I have read the RTP rfc 3550 a couple of times, still have a dou=
bt on the following issue:<br><br>- If a RTP unicast session with no change=
 in the source and destination address, changes its SSRC during the call se=
ssion is it permissible? In other words, between the same pair of source an=
d destination addresses, can a change in the SSRC value acceptable at the d=
estination?<br>
<br>I request the experts in this field to respond to this query?<br><br>Th=
anks<br>PC<br>

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To: paromita chowdhury <paromitaz@gmail.com>
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Subject: Re: [AVT] SSRC change in a single RTP session
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On 20 Oct 2010, at 09:25, paromita chowdhury wrote:
> I have read the RTP rfc 3550 a couple of times, still have a doubt on =
the following issue:
>=20
> - If a RTP unicast session with no change in the source and =
destination address, changes its SSRC during the call session is it =
permissible? In other words, between the same pair of source and =
destination addresses, can a change in the SSRC value acceptable at the =
destination?


Yes, the SSRC can change. In addition, you might see multiple SSRCs on =
the session (a unicast session is not restricted to one single SSRC from =
each participant, or even only two participants, since one may be an RTP =
translator or mixer).=20

--=20
Colin Perkins
http://csperkins.org/




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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On 20 Oct 2010, at 09:25, paromita chowdhury =
wrote:</div><blockquote type=3D"cite">I have read the RTP rfc 3550 a =
couple of times, still have a doubt on the following issue:<br><br>- If =
a RTP unicast session with no change in the source and destination =
address, changes its SSRC during the call session is it permissible? In =
other words, between the same pair of source and destination addresses, =
can a change in the SSRC value acceptable at the =
destination?<br></blockquote></div><div><br></div><div><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div>Yes, the SSRC =
can change. In addition, you might see multiple SSRCs on the session (a =
unicast session is not restricted to one single SSRC from each =
participant, or even only two participants, since one may be an RTP =
translator or mixer).&nbsp;<br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></body></html>=

--Apple-Mail-248-20003757--

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Message-ID: <AANLkTi=6jXtBpGcattXEzoCO-shLMUnZ-qRxd6Ux9Q3v@mail.gmail.com>
From: paromita chowdhury <paromitaz@gmail.com>
To: avt@ietf.org
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Cc: csp@csperkins.org
Subject: Re: [AVT] SSRC change in a single RTP session
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I noticed in a audio(VoIP) call with no change in the source or destination
IP addresses, the ssrc value of side of the communication changed while the
call was still going.

I find in the RFC 3550 that ssrc may change when one of the endpoint changes
but it mentions nothing about the scenario when endpoints do not change. Can
you please point/direct me to the section of the RFC stating it?

Thanks

On 20 October 2010 01:54, Colin Perkins <csp@csperkins.org> wrote:

> On 20 Oct 2010, at 09:25, paromita chowdhury wrote:
>
> I have read the RTP rfc 3550 a couple of times, still have a doubt on the
> following issue:
>
> - If a RTP unicast session with no change in the source and destination
> address, changes its SSRC during the call session is it permissible? In
> other words, between the same pair of source and destination addresses, can
> a change in the SSRC value acceptable at the destination?
>
>
> Yes, the SSRC can change. In addition, you might see multiple SSRCs on the
> session (a unicast session is not restricted to one single SSRC from each
> participant, or even only two participants, since one may be an RTP
> translator or mixer).
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
>

--0015174befdcb60b0d049308c609
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I noticed in a audio(VoIP) call with no change in the source or destination=
 IP addresses, the ssrc value of side of the communication changed while th=
e call was still going. <br><br>I find in the RFC 3550 that ssrc may change=
 when one of the endpoint changes but it mentions nothing about the scenari=
o when endpoints do not change. Can you please point/direct me to the secti=
on of the RFC stating it?<br>
<br>Thanks<br><br><div class=3D"gmail_quote">On 20 October 2010 01:54, Coli=
n Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org">csp@cs=
perkins.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; =
padding-left: 1ex;">
<div style=3D"word-wrap: break-word;"><div class=3D"im"><div><div>On 20 Oct=
 2010, at 09:25, paromita chowdhury wrote:</div><blockquote type=3D"cite">I=
 have read the RTP rfc 3550 a couple of times, still have a doubt on the fo=
llowing issue:<br>
<br>- If a RTP unicast session with no change in the source and destination=
 address, changes its SSRC during the call session is it permissible? In ot=
her words, between the same pair of source and destination addresses, can a=
 change in the SSRC value acceptable at the destination?<br>
</blockquote></div><div><br></div></div><div><span style=3D"border-collapse=
: separate; color: rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-s=
tyle: normal; font-variant: normal; font-weight: normal; letter-spacing: no=
rmal; line-height: normal; text-indent: 0px; text-transform: none; white-sp=
ace: normal; word-spacing: 0px;"><span style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: &#39;Lucida Sans Typewriter&#39;; font-si=
ze: 9px; font-style: normal; font-variant: normal; font-weight: normal; let=
ter-spacing: normal; line-height: normal; text-indent: 0px; text-transform:=
 none; white-space: normal; word-spacing: 0px;"><span style=3D"border-colla=
pse: separate; color: rgb(0, 0, 0); font-family: &#39;Lucida Sans Typewrite=
r&#39;; font-size: 9px; font-style: normal; font-variant: normal; font-weig=
ht: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px;"><span style=
=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: &#39;Lucid=
a Sans Typewriter&#39;; font-size: 9px; font-style: normal; font-variant: n=
ormal; font-weight: normal; letter-spacing: normal; line-height: normal; te=
xt-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0p=
x;"><span style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-fam=
ily: &#39;Lucida Sans Typewriter&#39;; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; line-hei=
ght: normal; text-indent: 0px; text-transform: none; white-space: normal; w=
ord-spacing: 0px;"><div>
Yes, the SSRC can change. In addition, you might see multiple SSRCs on the =
session (a unicast session is not restricted to one single SSRC from each p=
articipant, or even only two participants, since one may be an RTP translat=
or or mixer).=A0<br>
<br></div><div>--=A0</div><div></div><div>Colin Perkins</div><div><a href=
=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</a></div=
></span></span></span><br></span><br></span>
</div>
<br></div></blockquote></div><br>

--0015174befdcb60b0d049308c609--

From csp@csperkins.org  Wed Oct 20 02:42:25 2010
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From: Colin Perkins <csp@csperkins.org>
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Date: Wed, 20 Oct 2010 10:38:47 +0100
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References: <AANLkTikQ5ShYa-CR7-28Wvj8PxH+Gb6dM9rVUWP6ugLW@mail.gmail.com> <2F4E0AC9-6542-4153-A652-9FEF6CB870B0@csperkins.org> <AANLkTi=6jXtBpGcattXEzoCO-shLMUnZ-qRxd6Ux9Q3v@mail.gmail.com>
To: paromita chowdhury <paromitaz@gmail.com>
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Subject: Re: [AVT] SSRC change in a single RTP session
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An SSRC collision can happen at any time, and will force a change of =
SSRC. A new participant can join at any time, causing new SSRCs to =
appear. An existing participant may decide to send an additional media =
stream into a session, causing a new SSRC to appear. An application may =
restart, causing an SSRC change. A source may change its transport =
address requiring an SSRC change, but do so behind a NAT such that you =
don't see the change of transport address. An implementation may be =
broken, and change its SSRC for no reason.

Don't assume the SSRC is constant. Nothing in RFC 3550 requires it to =
be, and there are many things that may cause it to change.

Colin


On 20 Oct 2010, at 10:10, paromita chowdhury wrote:
> I noticed in a audio(VoIP) call with no change in the source or =
destination IP addresses, the ssrc value of side of the communication =
changed while the call was still going.=20
>=20
> I find in the RFC 3550 that ssrc may change when one of the endpoint =
changes but it mentions nothing about the scenario when endpoints do not =
change. Can you please point/direct me to the section of the RFC stating =
it?
>=20
> Thanks
>=20
> On 20 October 2010 01:54, Colin Perkins <csp@csperkins.org> wrote:
> On 20 Oct 2010, at 09:25, paromita chowdhury wrote:
>> I have read the RTP rfc 3550 a couple of times, still have a doubt on =
the following issue:
>>=20
>> - If a RTP unicast session with no change in the source and =
destination address, changes its SSRC during the call session is it =
permissible? In other words, between the same pair of source and =
destination addresses, can a change in the SSRC value acceptable at the =
destination?
>=20
>=20
> Yes, the SSRC can change. In addition, you might see multiple SSRCs on =
the session (a unicast session is not restricted to one single SSRC from =
each participant, or even only two participants, since one may be an RTP =
translator or mixer).=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/


--Apple-Mail-250-22682735
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	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">An =
SSRC collision can happen at any time, and will force a change of SSRC. =
A new participant can join at any time, causing new SSRCs to appear. An =
existing participant may decide to send an additional media stream into =
a session, causing a new SSRC to appear. An application may restart, =
causing an SSRC change. A source may change its transport address =
requiring an SSRC change, but do so behind a NAT such that you don't see =
the change of transport address. An implementation may be broken, and =
change its SSRC for no reason.<div><br></div><div>Don't assume the SSRC =
is constant. Nothing in RFC 3550 requires it to be, and there are many =
things that may cause it to =
change.</div><div><br></div><div>Colin<br><div><div><br></div><div><br><di=
v><div>On 20 Oct 2010, at 10:10, paromita chowdhury =
wrote:</div><blockquote type=3D"cite">I noticed in a audio(VoIP) call =
with no change in the source or destination IP addresses, the ssrc value =
of side of the communication changed while the call was still going. =
<br><br>I find in the RFC 3550 that ssrc may change when one of the =
endpoint changes but it mentions nothing about the scenario when =
endpoints do not change. Can you please point/direct me to the section =
of the RFC stating it?<br>
<br>Thanks<br><br><div class=3D"gmail_quote">On 20 October 2010 01:54, =
Colin Perkins <span dir=3D"ltr">&lt;<a =
href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"border-left-width: =
1px; border-left-style: solid; border-left-color: rgb(204, 204, 204); =
margin-top: 0pt; margin-right: 0pt; margin-bottom: 0pt; margin-left: =
0.8ex; padding-left: 1ex; position: static; z-index: auto; ">
<div style=3D"word-wrap: break-word;"><div class=3D"im"><div><div>On 20 =
Oct 2010, at 09:25, paromita chowdhury wrote:</div><blockquote =
type=3D"cite">I have read the RTP rfc 3550 a couple of times, still have =
a doubt on the following issue:<br>
<br>- If a RTP unicast session with no change in the source and =
destination address, changes its SSRC during the call session is it =
permissible? In other words, between the same pair of source and =
destination addresses, can a change in the SSRC value acceptable at the =
destination?<br>
</blockquote></div><div><br></div></div><div><span =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Arial; font-size: 9px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px;"><span style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; font-size: 9px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px;"><span =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; text-indent: 0px; text-transform: none; =
white-space: normal; word-spacing: 0px;"><span style=3D"border-collapse: =
separate; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px;"><span =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; text-indent: 0px; text-transform: none; =
white-space: normal; word-spacing: 0px;"><div>
Yes, the SSRC can change. In addition, you might see multiple SSRCs on =
the session (a unicast session is not restricted to one single SSRC from =
each participant, or even only two participants, since one may be an RTP =
translator or mixer).&nbsp;<br>
<br></div><div>--&nbsp;</div><div></div><div>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/" =
target=3D"_blank">http://csperkins.org/</a></div></span></span></span></sp=
an></span></div></div></blockquote></div></blockquote></div><br></div></di=
v></div></body></html>=

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From: paromita chowdhury <paromitaz@gmail.com>
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Subject: Re: [AVT] SSRC change in a single RTP session
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--0016e6dbdf47b1cbb70493094682
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Thanks for the response.. it clarified my doubt

On 20 October 2010 02:38, Colin Perkins <csp@csperkins.org> wrote:

> An SSRC collision can happen at any time, and will force a change of SSRC.
> A new participant can join at any time, causing new SSRCs to appear. An
> existing participant may decide to send an additional media stream into a
> session, causing a new SSRC to appear. An application may restart, causing
> an SSRC change. A source may change its transport address requiring an SSRC
> change, but do so behind a NAT such that you don't see the change of
> transport address. An implementation may be broken, and change its SSRC for
> no reason.
>
> Don't assume the SSRC is constant. Nothing in RFC 3550 requires it to be,
> and there are many things that may cause it to change.
>
> Colin
>
>
>
> On 20 Oct 2010, at 10:10, paromita chowdhury wrote:
>
> I noticed in a audio(VoIP) call with no change in the source or destination
> IP addresses, the ssrc value of side of the communication changed while the
> call was still going.
>
> I find in the RFC 3550 that ssrc may change when one of the endpoint
> changes but it mentions nothing about the scenario when endpoints do not
> change. Can you please point/direct me to the section of the RFC stating it?
>
> Thanks
>
> On 20 October 2010 01:54, Colin Perkins <csp@csperkins.org> wrote:
>
>> On 20 Oct 2010, at 09:25, paromita chowdhury wrote:
>>
>> I have read the RTP rfc 3550 a couple of times, still have a doubt on the
>> following issue:
>>
>> - If a RTP unicast session with no change in the source and destination
>> address, changes its SSRC during the call session is it permissible? In
>> other words, between the same pair of source and destination addresses, can
>> a change in the SSRC value acceptable at the destination?
>>
>>
>> Yes, the SSRC can change. In addition, you might see multiple SSRCs on the
>> session (a unicast session is not restricted to one single SSRC from each
>> participant, or even only two participants, since one may be an RTP
>> translator or mixer).
>>
>> --
>> Colin Perkins
>> http://csperkins.org/
>>
>
>

--0016e6dbdf47b1cbb70493094682
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thanks for the response.. it clarified my doubt<br><br><div class=3D"gmail_=
quote">On 20 October 2010 02:38, Colin Perkins <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, 2=
04, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style=3D"word-wrap: break-word;">An SSRC collision can happen at any t=
ime, and will force a change of SSRC. A new participant can join at any tim=
e, causing new SSRCs to appear. An existing participant may decide to send =
an additional media stream into a session, causing a new SSRC to appear. An=
 application may restart, causing an SSRC change. A source may change its t=
ransport address requiring an SSRC change, but do so behind a NAT such that=
 you don&#39;t see the change of transport address. An implementation may b=
e broken, and change its SSRC for no reason.<div>
<br></div><div>Don&#39;t assume the SSRC is constant. Nothing in RFC 3550 r=
equires it to be, and there are many things that may cause it to change.</d=
iv><div><br></div><div><font color=3D"#888888">Colin</font><div><div></div>
<div class=3D"h5"><br><div><div><br></div><div><br><div><div>On 20 Oct 2010=
, at 10:10, paromita chowdhury wrote:</div><blockquote type=3D"cite">I noti=
ced in a audio(VoIP) call with no change in the source or destination IP ad=
dresses, the ssrc value of side of the communication changed while the call=
 was still going. <br>
<br>I find in the RFC 3550 that ssrc may change when one of the endpoint ch=
anges but it mentions nothing about the scenario when endpoints do not chan=
ge. Can you please point/direct me to the section of the RFC stating it?<br=
>

<br>Thanks<br><br><div class=3D"gmail_quote">On 20 October 2010 01:54, Coli=
n Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org" target=
=3D"_blank">csp@csperkins.org</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0=
pt 0pt 0pt 0.8ex; padding-left: 1ex;">

<div style=3D"word-wrap: break-word;"><div><div><div>On 20 Oct 2010, at 09:=
25, paromita chowdhury wrote:</div><blockquote type=3D"cite">I have read th=
e RTP rfc 3550 a couple of times, still have a doubt on the following issue=
:<br>

<br>- If a RTP unicast session with no change in the source and destination=
 address, changes its SSRC during the call session is it permissible? In ot=
her words, between the same pair of source and destination addresses, can a=
 change in the SSRC value acceptable at the destination?<br>

</blockquote></div><div><br></div></div><div><span style=3D"border-collapse=
: separate; color: rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-s=
tyle: normal; font-variant: normal; font-weight: normal; letter-spacing: no=
rmal; line-height: normal; text-indent: 0px; text-transform: none; white-sp=
ace: normal; word-spacing: 0px;"><span style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: &#39;Lucida Sans Typewriter&#39;; font-si=
ze: 9px; font-style: normal; font-variant: normal; font-weight: normal; let=
ter-spacing: normal; line-height: normal; text-indent: 0px; text-transform:=
 none; white-space: normal; word-spacing: 0px;"><span style=3D"border-colla=
pse: separate; color: rgb(0, 0, 0); font-family: &#39;Lucida Sans Typewrite=
r&#39;; font-size: 9px; font-style: normal; font-variant: normal; font-weig=
ht: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px;"><span style=
=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: &#39;Lucid=
a Sans Typewriter&#39;; font-size: 9px; font-style: normal; font-variant: n=
ormal; font-weight: normal; letter-spacing: normal; line-height: normal; te=
xt-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0p=
x;"><span style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-fam=
ily: &#39;Lucida Sans Typewriter&#39;; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; line-hei=
ght: normal; text-indent: 0px; text-transform: none; white-space: normal; w=
ord-spacing: 0px;"><div>

Yes, the SSRC can change. In addition, you might see multiple SSRCs on the =
session (a unicast session is not restricted to one single SSRC from each p=
articipant, or even only two participants, since one may be an RTP translat=
or or mixer).=A0<br>

<br></div><div>--=A0</div><div></div><div>Colin Perkins</div><div><a href=
=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</a></div=
></span></span></span></span></span></div></div></blockquote></div></blockq=
uote>
</div><br></div></div></div></div></div></div></blockquote></div><br>

--0016e6dbdf47b1cbb70493094682--

From csp@csperkins.org  Wed Oct 20 06:50:19 2010
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From: Colin Perkins <csp@csperkins.org>
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To: Jinwei Xia <xiajinwei@huawei.com>
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Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-splicing-for-rtp-00
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Hi Jinwei,

My impression is that this draft confuses the issue by focussing too =
much on scenarios where the content to be inserted is also delivered =
over RTP. Whether the content to be inserted is delivered as a real-time =
RTP stream or fetched from local file storage does not affect the =
insertion process, and so doesn't need to be mentioned in the draft. I =
still believe that a standard RTP translator is sufficient to solve this =
problem, although there may be useful things to say about how to =
efficiently implement such a translator.=20

Colin


On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> Hi,
>=20
> =46rom the discussion on IETF 78th, there were some misunderstandings =
of the scope of splicing draft. So we discard previous problem statement =
draft while initiate new solution draft. In this new draft, We emphysize =
content insertion issue is indeed a RTP-generic rather than MPEG2 =
TS-specific.=20
>=20
> Hope the new draft can address most issues AVT concern, any comments =
are very appreciated!
>=20
>=20
> BR!
>=20
> Jinwei=20
>=20
>> -----Original Message-----
>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]=20
>> Sent: Saturday, October 16, 2010 9:58 AM
>> To: xiajinwei@huawei.com
>> Subject: New Version Notification for=20
>> draft-xia-avt-splicing-for-rtp-00
>>=20
>>=20
>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt=20
>> has been successfully submitted by Jinwei Xia and posted to=20
>> the IETF repository.
>>=20
>> Filename:	 draft-xia-avt-splicing-for-rtp
>> Revision:	 00
>> Title:		 Content Splicing for RTP Sessions
>> Creation_date:	 2010-10-16
>> WG ID:		 Independent Submission
>> Number_of_pages: 12
>>=20
>> Abstract:
>> This memo outlines RTP splicing.  Splicing is a process that=20
>> allows a new multimedia stream to be inserted into current=20
>> multimedia stream and to be conveyed to receiver for a period=20
>> of time.  This memo discusses the requirements of RTP=20
>> splicing.  In order to satisfy the requirements, this memo=20
>> lists several existing intermediary network elements as=20
>> alternatives and evaluates whether one of alternatives can be=20
>> used to perform RTP splicing.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat.
>>=20
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From csp@csperkins.org  Wed Oct 20 15:01:01 2010
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Qin,

Thanks for updating this. These changes address my concerns.

Colin


On 13 Oct 2010, at 13:32, Qin Wu wrote:
> Hi, folks:
> 04 version is posted which addresses the comments from Colin
> * Add some text to General Synchronisation Offset Metrics Block to =
allow
>  more than two stream in the RTP multimedia session.
> * Clarify the Picture Type in the terminology section
> * Separate TR 101 290 decodability metrics report from this draft
> Your comments and suggestions are welcome!
>=20
> Regards!
> -Qin
> ----- Original Message -----=20
> From: "IETF I-D Submission Tool" <idsubmission@ietf.org>
> To: <sunseawq@huawei.com>
> Cc: <gwz@net-zen.net>; <Roland.Schott@telekom.de>
> Sent: Wednesday, October 13, 2010 8:16 PM
> Subject: New Version Notification for =
draft-wu-avt-rtcp-xr-quality-monitoring-04
>=20
>=20
>>=20
>> A new version of I-D, draft-wu-avt-rtcp-xr-quality-monitoring-04.txt =
has been successfully submitted by Wenson Wu and posted to the IETF =
repository.
>>=20
>> Filename: draft-wu-avt-rtcp-xr-quality-monitoring
>> Revision: 04
>> Title: RTP Control Protocol Extended Reports (RTCP XR) Report Blocks =
for Real- time Video Quality Monitoring
>> Creation_date: 2010-10-13
>> WG ID: Independent Submission
>> Number_of_pages: 22
>>=20
>> Abstract:
>> This document defines a set of RTP Control Protocol Extended Reports
>> (RTCP XR) Report Blocks and associated SDP parameters allowing the
>> report of video quality metrics, primarily for video applications of
>> RTP.
>>=20
>>=20
>>=20
>> The IETF Secretariat.
>>=20
>>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From xiajinwei@huawei.com  Wed Oct 20 23:32:04 2010
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Date: Thu, 21 Oct 2010 14:33:25 +0800
From: Jinwei Xia <xiajinwei@huawei.com>
In-reply-to: <F9549DE5-7760-40A5-9690-AEDFE1CB6E80@csperkins.org>
To: 'Colin Perkins' <csp@csperkins.org>
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Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-splicing-for-rtp-00
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Hi Colin,

Thank your comments. The scenario description is given in problem statements
draft, it is unnecessary to repeat that in this draft. 

Indeed, a standard translator can handle splicing without any extension, and
it still needs instructions from media source to implement splicing. 

I will update the draft according to our maillist discussion. Any commets
from others are also very appreciated!


Thank you
Jinwei

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org] 
> Sent: Wednesday, October 20, 2010 9:52 PM
> To: Jinwei Xia
> Cc: 'IETF AVT WG'
> Subject: Re: [AVT] FW: New Version Notification for 
> draft-xia-avt-splicing-for-rtp-00
> 
> Hi Jinwei,
> 
> My impression is that this draft confuses the issue by 
> focussing too much on scenarios where the content to be 
> inserted is also delivered over RTP. Whether the content to 
> be inserted is delivered as a real-time RTP stream or fetched 
> from local file storage does not affect the insertion 
> process, and so doesn't need to be mentioned in the draft. I 
> still believe that a standard RTP translator is sufficient to 
> solve this problem, although there may be useful things to 
> say about how to efficiently implement such a translator. 
> 
> Colin
> 
> 
> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> > Hi,
> > 
> > From the discussion on IETF 78th, there were some 
> misunderstandings of the scope of splicing draft. So we 
> discard previous problem statement draft while initiate new 
> solution draft. In this new draft, We emphysize content 
> insertion issue is indeed a RTP-generic rather than MPEG2 
> TS-specific. 
> > 
> > Hope the new draft can address most issues AVT concern, any 
> comments are very appreciated!
> > 
> > 
> > BR!
> > 
> > Jinwei
> > 
> >> -----Original Message-----
> >> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> >> Sent: Saturday, October 16, 2010 9:58 AM
> >> To: xiajinwei@huawei.com
> >> Subject: New Version Notification for 
> >> draft-xia-avt-splicing-for-rtp-00
> >> 
> >> 
> >> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
> >> has been successfully submitted by Jinwei Xia and posted 
> to the IETF 
> >> repository.
> >> 
> >> Filename:	 draft-xia-avt-splicing-for-rtp
> >> Revision:	 00
> >> Title:		 Content Splicing for RTP Sessions
> >> Creation_date:	 2010-10-16
> >> WG ID:		 Independent Submission
> >> Number_of_pages: 12
> >> 
> >> Abstract:
> >> This memo outlines RTP splicing.  Splicing is a process 
> that allows a 
> >> new multimedia stream to be inserted into current 
> multimedia stream 
> >> and to be conveyed to receiver for a period of time.  This memo 
> >> discusses the requirements of RTP splicing.  In order to 
> satisfy the 
> >> requirements, this memo lists several existing 
> intermediary network 
> >> elements as alternatives and evaluates whether one of alternatives 
> >> can be used to perform RTP splicing.
> >> 
> >> 
> >> 
> >> 
> >> The IETF Secretariat.
> >> 
> >> 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> 
> 
> -- 
> Colin Perkins
> http://csperkins.org/
> 
> 
> 


From Even.roni@huawei.com  Wed Oct 20 23:58:40 2010
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Date: Thu, 21 Oct 2010 08:56:34 +0200
From: Roni Even <Even.roni@huawei.com>
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Cc: 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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Hi Colin,
I am not so sure about this being a regular RTP translator. According To RFC
5117 " A Translator always keeps the SSRC for a stream across the
translation".
Also in the splicing case it is not point to point or multipoint but more
like multipoint to point or multipoint. This looks more like a mixer that
does not provide the CSRC since one of the requirements is to make it look
to the receiver that all the media is coming from the same source.
One issue that will need to be addressed is how to report the RTCP RR from
the receivers back to the sender of the primary or insertion streams. This
may need some state keeping in the splicer.

Roni Even 

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Colin Perkins
> Sent: Wednesday, October 20, 2010 3:52 PM
> To: Jinwei Xia
> Cc: 'IETF AVT WG'
> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
> splicing-for-rtp-00
> 
> Hi Jinwei,
> 
> My impression is that this draft confuses the issue by focussing too
> much on scenarios where the content to be inserted is also delivered
> over RTP. Whether the content to be inserted is delivered as a real-
> time RTP stream or fetched from local file storage does not affect the
> insertion process, and so doesn't need to be mentioned in the draft. I
> still believe that a standard RTP translator is sufficient to solve
> this problem, although there may be useful things to say about how to
> efficiently implement such a translator.
> 
> Colin
> 
> 
> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> > Hi,
> >
> > From the discussion on IETF 78th, there were some misunderstandings
> of the scope of splicing draft. So we discard previous problem
> statement draft while initiate new solution draft. In this new draft,
> We emphysize content insertion issue is indeed a RTP-generic rather
> than MPEG2 TS-specific.
> >
> > Hope the new draft can address most issues AVT concern, any comments
> are very appreciated!
> >
> >
> > BR!
> >
> > Jinwei
> >
> >> -----Original Message-----
> >> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> >> Sent: Saturday, October 16, 2010 9:58 AM
> >> To: xiajinwei@huawei.com
> >> Subject: New Version Notification for
> >> draft-xia-avt-splicing-for-rtp-00
> >>
> >>
> >> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
> >> has been successfully submitted by Jinwei Xia and posted to
> >> the IETF repository.
> >>
> >> Filename:	 draft-xia-avt-splicing-for-rtp
> >> Revision:	 00
> >> Title:		 Content Splicing for RTP Sessions
> >> Creation_date:	 2010-10-16
> >> WG ID:		 Independent Submission
> >> Number_of_pages: 12
> >>
> >> Abstract:
> >> This memo outlines RTP splicing.  Splicing is a process that
> >> allows a new multimedia stream to be inserted into current
> >> multimedia stream and to be conveyed to receiver for a period
> >> of time.  This memo discusses the requirements of RTP
> >> splicing.  In order to satisfy the requirements, this memo
> >> lists several existing intermediary network elements as
> >> alternatives and evaluates whether one of alternatives can be
> >> used to perform RTP splicing.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat.
> >>
> >>
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> 
> 
> --
> Colin Perkins
> http://csperkins.org/
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From ingemar.s.johansson@ericsson.com  Thu Oct 21 00:22:27 2010
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Colin Perkins <csp@csperkins.org>, Bob Briscoe <rbriscoe@jungle.bt.co.uk>
Date: Thu, 21 Oct 2010 09:23:57 +0200
Thread-Topic: Removing nonce (WAS RE: [AVT] draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Subject: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Hi

I am working on the draft, removing the nonce stuff. As I understand it, se=
ction 5.3 can be removed entirely, but I guess some text is needed to indic=
ated that RFC3611 (4.1) can be used to indicate packets that are lost or EC=
N-CE marked.=20
ECN-CE is not mentioned in RFC3611, do you believe that it is valid to use =
RFC3611 for indication on ECN-CE. I guess it is if one stick to the princip=
le that ECN-CE should be reacted on in the same way as a packet loss. What =
is your opinion.

/Ingemar

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]=20
> Sent: den 19 oktober 2010 16:21
> To: Bob Briscoe
> Cc: ken carlberg; draft-ietf-avt-ecn-for-rtp@tools.ietf.org;=20
> avt@ietf.org
> Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating
>=20
> The ECN nonce support is pretty modular. I'm happy to rip it=20
> out of this draft, since we can always resurrect it in a=20
> separate draft if there's a need later.
>=20
> Colin
>=20
>=20
> On 26 Aug 2010, at 00:36, Bob Briscoe wrote:
> > Ken,
> >=20
> > We're violently agreeing I think, modulo just a little more=20
> optimism=20
> > on my part about ConEx ;)
> >=20
> > Colin already said (in a Maastricht corridor discussion)=20
> that he would be happy to defer receiver misbehaviour stuff.=20
> But I don't think I had put the specific proposal to him I=20
> put in my mail - modularising the possibility of two=20
> alternative future approaches. I've cut all that text from=20
> the thread now so, Colin, feel free to respond to Ken's=20
> request for your view in the previous mail instead.
> >=20
> > One response inline, altho I think that part of the=20
> conversation is getting pretty obscure...
> >=20
> > At 21:39 25/08/2010, ken carlberg wrote:
> >> Bob,
> >>=20
> >> FYI. its been a while since this email was sent, so I'm=20
> not snipping any of it, but my responses are few and=20
> scattered throughout.  And in this case, my opinions are not=20
> that strong, so I can probably be swayed by my co-authors if=20
> they feel differently.  That said...
> >>=20
> >> On Aug 5, 2010, at 6:41 PM, Bob Briscoe wrote:
> >> > Magnus, Ingemar, Colin, Piers, Ken,
> >=20
> > [snip]
> >=20
> >> > T3c/ Misbehaving end-points
> >> > -------------------
> >> > "
> >> > 1968                                                    =20
>  It is far from certain
> >> > 1969          that a receiver would be able to get a=20
> significant larger share of
> >> > 1970          the resources.  That assumes a high enough=20
> level of aggregation
> >> > 1971          that there are flows to acquire shares from.
> >> > "
> >> > Don't agree. It's completely certain that an=20
> unresponsive receiver (or sender) can get a significantly=20
> larger share of the resources, whether or not it uses ECN.=20
> Just try it. Send an unresponsive flow along with some=20
> responsive flows (e.g TCP) and the TCPs reduce their rates=20
> while the unresponsive flow gets whatever rate it chooses to send at.
> >> > "
> >>=20
> >> well, it depends on context.  if I have an aggregate of=20
> flows that are only periodically and in a bursty manner=20
> triggering a  congestive state, as opposed to a sustained=20
> level of congestion condition (though not collapse), then=20
> there are no guarantees that all the other flows will be=20
> marked or even suffer packet loss.  And...
> >=20
> >=20
> > [BB]: Going back to the para at issue in the I-D that I=20
> quoted: could I paraphrase it as, "If there are only two=20
> flows sharing a bottleneck, a cheating receiver cannot get=20
> more than twice as much as it would under equality, which=20
> isn't so bad."? If that's what it means, nah! Capacity=20
> sharing problems are surely better measured by how little=20
> others get, rather than how much more the cheater gets.
> >=20
> > Moving on to your periodic bursts scenario, I'm not quite=20
> sure I understand what you are getting at or how it's=20
> relevant to the para in the I-D.
> >=20
> > Whatever, surely fair shares aren't an issue during any=20
> period when a link isn't fully utilised. Becasue, if a link=20
> isn't fully utilised for some period, however brief, it is=20
> impossible to take more than your fair share over that period=20
> (for any definition of fairness) and it's impossible to cause=20
> others to get less than their fair share, because nothing you=20
> are doing stops others taking more than they are taking.
> >=20
> > Then,  during those bursts, it's only relevant that there=20
> would have marking or loss if someone else had taken a=20
> greater share than they were taking at that time. But while=20
> there's no marking or loss, it implies no-one else is wanting=20
> more than they are taking. So lack of marking or loss=20
> indicates there cannot have been any sharing problem.
> >=20
> > Or am I missing your point?
> >=20
> >=20
> > Bob
> >=20
> >=20
> >=20
> > ________________________________________________________________
> > Bob Briscoe,                                BT Innovate & Design=20
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> =

From magnus.westerlund@ericsson.com  Thu Oct 21 00:30:15 2010
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Subject: Re: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Ingemar Johansson S skrev 2010-10-21 09:23:
> Hi
> 
> I am working on the draft, removing the nonce stuff. As I understand it, section 5.3 can be removed entirely, but I guess some text is needed to indicated that RFC3611 (4.1) can be used to indicate packets that are lost or ECN-CE marked. 
> ECN-CE is not mentioned in RFC3611, do you believe that it is valid to use RFC3611 for indication on ECN-CE. I guess it is if one stick to the principle that ECN-CE should be reacted on in the same way as a packet loss. What is your opinion.

I at least are of the opinion that if a packet arrived with a CE mark,
it is still delivered. It shall not be marked as lost in the RTCP XR
packet loss vector. If we want/need to know exactly which packets where
CE marked, then we should define a CE marking XR vector for that.
Overloading of this would reduce the utility of the packet loss vector.

Did you have any specific need for explicit CE markings?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

From ingemar.s.johansson@ericsson.com  Thu Oct 21 00:49:49 2010
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 21 Oct 2010 09:51:14 +0200
Thread-Topic: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Subject: Re: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Hi=20

Sounds like we should define a CE report block similar to 4.1 in RFC3611 th=
en. If both CE and loss occur during a report interval one will of course g=
et both Loss and CE report blocks. A solution is of course a combined Loss/=
CE report block to reduce some of the overhead (I believe I can live with s=
eparate Loss and CE blocks and the extra overhead)

I can imagine that one can consider a more relaxed behavior to CE than Loss=
 although I have not seen much evidence of it sofar a least in wireless acc=
ess scenarios, one is there likely to see CE shortly before a loss or exces=
sive delay. Wireline may be a different story though..

/Ingemar=20

> -----Original Message-----
> From: Magnus Westerlund=20
> Sent: den 21 oktober 2010 09:32
> To: Ingemar Johansson S
> Cc: Colin Perkins; Bob Briscoe;=20
> draft-ietf-avt-ecn-for-rtp@tools.ietf.org; avt@ietf.org
> Subject: Re: [AVT] Removing nonce (WAS RE:=20
> draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
>=20
> Ingemar Johansson S skrev 2010-10-21 09:23:
> > Hi
> >=20
> > I am working on the draft, removing the nonce stuff. As I=20
> understand it, section 5.3 can be removed entirely, but I=20
> guess some text is needed to indicated that RFC3611 (4.1) can=20
> be used to indicate packets that are lost or ECN-CE marked.=20
> > ECN-CE is not mentioned in RFC3611, do you believe that it=20
> is valid to use RFC3611 for indication on ECN-CE. I guess it=20
> is if one stick to the principle that ECN-CE should be=20
> reacted on in the same way as a packet loss. What is your opinion.
>=20
> I at least are of the opinion that if a packet arrived with a=20
> CE mark, it is still delivered. It shall not be marked as=20
> lost in the RTCP XR packet loss vector. If we want/need to=20
> know exactly which packets where CE marked, then we should=20
> define a CE marking XR vector for that.
> Overloading of this would reduce the utility of the packet=20
> loss vector.
>=20
> Did you have any specific need for explicit CE markings?
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> =

From ingemar.s.johansson@ericsson.com  Thu Oct 21 02:17:55 2010
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Hi

After some consideration I believe that it is better to skip the RLE XR blo=
ck altogether. I don't see a clear need now to explicitly see if specific p=
ackets are CE marked. But should such a need occur later on, then we can ad=
d perhaps add this, this addition is also quite simple.

BTW.. It was not my intention to flood the AVT list with this thread, sorry=
 for the extra noise...

/Ingemar

> -----Original Message-----
> From: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]=20
> Sent: den 21 oktober 2010 09:51
> To: Magnus Westerlund
> Cc: Colin Perkins; Bob Briscoe;=20
> draft-ietf-avt-ecn-for-rtp@tools.ietf.org; avt@ietf.org
> Subject: RE: [AVT] Removing nonce (WAS RE:=20
> draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
>=20
> Hi=20
>=20
> Sounds like we should define a CE report block similar to 4.1=20
> in RFC3611 then. If both CE and loss occur during a report=20
> interval one will of course get both Loss and CE report=20
> blocks. A solution is of course a combined Loss/CE report=20
> block to reduce some of the overhead (I believe I can live=20
> with separate Loss and CE blocks and the extra overhead)
>=20
> I can imagine that one can consider a more relaxed behavior=20
> to CE than Loss although I have not seen much evidence of it=20
> sofar a least in wireless access scenarios, one is there=20
> likely to see CE shortly before a loss or excessive delay.=20
> Wireline may be a different story though..
>=20
> /Ingemar=20
>=20
> > -----Original Message-----
> > From: Magnus Westerlund
> > Sent: den 21 oktober 2010 09:32
> > To: Ingemar Johansson S
> > Cc: Colin Perkins; Bob Briscoe;
> > draft-ietf-avt-ecn-for-rtp@tools.ietf.org; avt@ietf.org
> > Subject: Re: [AVT] Removing nonce (WAS RE:=20
> > draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
> >=20
> > Ingemar Johansson S skrev 2010-10-21 09:23:
> > > Hi
> > >=20
> > > I am working on the draft, removing the nonce stuff. As I
> > understand it, section 5.3 can be removed entirely, but I=20
> guess some=20
> > text is needed to indicated that RFC3611 (4.1) can be used=20
> to indicate=20
> > packets that are lost or ECN-CE marked.
> > > ECN-CE is not mentioned in RFC3611, do you believe that it
> > is valid to use RFC3611 for indication on ECN-CE. I guess=20
> it is if one=20
> > stick to the principle that ECN-CE should be reacted on in the same=20
> > way as a packet loss. What is your opinion.
> >=20
> > I at least are of the opinion that if a packet arrived with=20
> a CE mark,=20
> > it is still delivered. It shall not be marked as lost in=20
> the RTCP XR=20
> > packet loss vector. If we want/need to know exactly which packets=20
> > where CE marked, then we should define a CE marking XR vector for=20
> > that.
> > Overloading of this would reduce the utility of the packet loss=20
> > vector.
> >=20
> > Did you have any specific need for explicit CE markings?
> >=20
> > Cheers
> >=20
> > Magnus Westerlund
> >=20
> >=20
> ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> >=20
> ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >=20
> ----------------------------------------------------------------------
> > =

From csp@csperkins.org  Thu Oct 21 10:21:18 2010
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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Hi Roni,

I think this problem becomes clearer if you consider a splicer that is =
reading the content to be inserted from a local file on disk. In that =
case, the splicer is unquestionably a standard RTP translator, in the =
middle of a one-to-one or one-to-many RTP session.=20

That there are also some cases where an entirely separate RTP session is =
used to deliver the content to be inserted doesn't (conceptually) affect =
the translation process at all.=20

Colin



On 21 Oct 2010, at 07:56, Roni Even wrote:
> Hi Colin,
> I am not so sure about this being a regular RTP translator. According =
To RFC 5117 " A Translator always keeps the SSRC for a stream across the =
translation". Also in the splicing case it is not point to point or =
multipoint but more like multipoint to point or multipoint. This looks =
more like a mixer that does not provide the CSRC since one of the =
requirements is to make it look to the receiver that all the media is =
coming from the same source. One issue that will need to be addressed is =
how to report the RTCP RR from the receivers back to the sender of the =
primary or insertion streams. This may need some state keeping in the =
splicer.
>=20
> Roni Even=20
>=20
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Colin Perkins
>> Sent: Wednesday, October 20, 2010 3:52 PM
>> To: Jinwei Xia
>> Cc: 'IETF AVT WG'
>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>> splicing-for-rtp-00
>>=20
>> Hi Jinwei,
>>=20
>> My impression is that this draft confuses the issue by focussing too =
much on scenarios where the content to be inserted is also delivered =
over RTP. Whether the content to be inserted is delivered as a real- =
time RTP stream or fetched from local file storage does not affect the =
insertion process, and so doesn't need to be mentioned in the draft. I =
still believe that a standard RTP translator is sufficient to solve this =
problem, although there may be useful things to say about how to =
efficiently implement such a translator.
>>=20
>> Colin
>>=20
>>=20
>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
>>> Hi,
>>>=20
>>> =46rom the discussion on IETF 78th, there were some =
misunderstandings
>> of the scope of splicing draft. So we discard previous problem
>> statement draft while initiate new solution draft. In this new draft,
>> We emphysize content insertion issue is indeed a RTP-generic rather
>> than MPEG2 TS-specific.
>>>=20
>>> Hope the new draft can address most issues AVT concern, any comments
>> are very appreciated!
>>>=20
>>>=20
>>> BR!
>>>=20
>>> Jinwei
>>>=20
>>>> -----Original Message-----
>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
>>>> Sent: Saturday, October 16, 2010 9:58 AM
>>>> To: xiajinwei@huawei.com
>>>> Subject: New Version Notification for
>>>> draft-xia-avt-splicing-for-rtp-00
>>>>=20
>>>>=20
>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
>>>> has been successfully submitted by Jinwei Xia and posted to
>>>> the IETF repository.
>>>>=20
>>>> Filename:	 draft-xia-avt-splicing-for-rtp
>>>> Revision:	 00
>>>> Title:		 Content Splicing for RTP Sessions
>>>> Creation_date:	 2010-10-16
>>>> WG ID:		 Independent Submission
>>>> Number_of_pages: 12
>>>>=20
>>>> Abstract:
>>>> This memo outlines RTP splicing.  Splicing is a process that
>>>> allows a new multimedia stream to be inserted into current
>>>> multimedia stream and to be conveyed to receiver for a period
>>>> of time.  This memo discusses the requirements of RTP
>>>> splicing.  In order to satisfy the requirements, this memo
>>>> lists several existing intermediary network elements as
>>>> alternatives and evaluates whether one of alternatives can be
>>>> used to perform RTP splicing.
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> The IETF Secretariat.
>>>>=20
>>>>=20
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>> --
>> Colin Perkins
>> http://csperkins.org/


From Even.roni@huawei.com  Thu Oct 21 11:28:10 2010
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Date: Thu, 21 Oct 2010 20:26:18 +0200
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <242A9169-3AE9-4C55-817A-B7B93F6005C9@csperkins.org>
To: 'Colin Perkins' <csp@csperkins.org>
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Cc: 'IETF AVT WG' <avt@ietf.org>
Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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Hi Colin,
In this use case the inserted content is not an RTP stream but comes from a
local storage. For this use case the splicer is a translator. It will still
need to handle the RTCP RR depending on who originated the data that was
sent by the translator.
My question will be if this is the typical use case or is there a use case
where the inserted data is not local and may arrive as RTP stream. In which
case there is more than one topology

Roni

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Thursday, October 21, 2010 7:23 PM
> To: Roni Even
> Cc: 'Jinwei Xia'; 'IETF AVT WG'
> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
> splicing-for-rtp-00
> 
> Hi Roni,
> 
> I think this problem becomes clearer if you consider a splicer that is
> reading the content to be inserted from a local file on disk. In that
> case, the splicer is unquestionably a standard RTP translator, in the
> middle of a one-to-one or one-to-many RTP session.
> 
> That there are also some cases where an entirely separate RTP session
> is used to deliver the content to be inserted doesn't (conceptually)
> affect the translation process at all.
> 
> Colin
> 
> 
> 
> On 21 Oct 2010, at 07:56, Roni Even wrote:
> > Hi Colin,
> > I am not so sure about this being a regular RTP translator. According
> To RFC 5117 " A Translator always keeps the SSRC for a stream across
> the translation". Also in the splicing case it is not point to point or
> multipoint but more like multipoint to point or multipoint. This looks
> more like a mixer that does not provide the CSRC since one of the
> requirements is to make it look to the receiver that all the media is
> coming from the same source. One issue that will need to be addressed
> is how to report the RTCP RR from the receivers back to the sender of
> the primary or insertion streams. This may need some state keeping in
> the splicer.
> >
> > Roni Even
> >
> >> -----Original Message-----
> >> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
> Of
> >> Colin Perkins
> >> Sent: Wednesday, October 20, 2010 3:52 PM
> >> To: Jinwei Xia
> >> Cc: 'IETF AVT WG'
> >> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
> >> splicing-for-rtp-00
> >>
> >> Hi Jinwei,
> >>
> >> My impression is that this draft confuses the issue by focussing too
> much on scenarios where the content to be inserted is also delivered
> over RTP. Whether the content to be inserted is delivered as a real-
> time RTP stream or fetched from local file storage does not affect the
> insertion process, and so doesn't need to be mentioned in the draft. I
> still believe that a standard RTP translator is sufficient to solve
> this problem, although there may be useful things to say about how to
> efficiently implement such a translator.
> >>
> >> Colin
> >>
> >>
> >> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> >>> Hi,
> >>>
> >>> From the discussion on IETF 78th, there were some misunderstandings
> >> of the scope of splicing draft. So we discard previous problem
> >> statement draft while initiate new solution draft. In this new
> draft,
> >> We emphysize content insertion issue is indeed a RTP-generic rather
> >> than MPEG2 TS-specific.
> >>>
> >>> Hope the new draft can address most issues AVT concern, any
> comments
> >> are very appreciated!
> >>>
> >>>
> >>> BR!
> >>>
> >>> Jinwei
> >>>
> >>>> -----Original Message-----
> >>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> >>>> Sent: Saturday, October 16, 2010 9:58 AM
> >>>> To: xiajinwei@huawei.com
> >>>> Subject: New Version Notification for
> >>>> draft-xia-avt-splicing-for-rtp-00
> >>>>
> >>>>
> >>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
> >>>> has been successfully submitted by Jinwei Xia and posted to
> >>>> the IETF repository.
> >>>>
> >>>> Filename:	 draft-xia-avt-splicing-for-rtp
> >>>> Revision:	 00
> >>>> Title:		 Content Splicing for RTP Sessions
> >>>> Creation_date:	 2010-10-16
> >>>> WG ID:		 Independent Submission
> >>>> Number_of_pages: 12
> >>>>
> >>>> Abstract:
> >>>> This memo outlines RTP splicing.  Splicing is a process that
> >>>> allows a new multimedia stream to be inserted into current
> >>>> multimedia stream and to be conveyed to receiver for a period
> >>>> of time.  This memo discusses the requirements of RTP
> >>>> splicing.  In order to satisfy the requirements, this memo
> >>>> lists several existing intermediary network elements as
> >>>> alternatives and evaluates whether one of alternatives can be
> >>>> used to perform RTP splicing.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> The IETF Secretariat.
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> Audio/Video Transport Working Group
> >>> avt@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/avt
> >>
> >> --
> >> Colin Perkins
> >> http://csperkins.org/


From csp@csperkins.org  Thu Oct 21 13:15:36 2010
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From: Colin Perkins <csp@csperkins.org>
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To: Roni Even <Even.roni@huawei.com>
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Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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Roni,

I don't think it matters that the content arrives as an RTP stream. It's =
an entirely separate stream, and doesn't affect the translation process. =
They're logically separate.

Colin



On 21 Oct 2010, at 19:26, Roni Even wrote:
> Hi Colin,
> In this use case the inserted content is not an RTP stream but comes =
from a local storage. For this use case the splicer is a translator. It =
will still need to handle the RTCP RR depending on who originated the =
data that was sent by the translator.
> My question will be if this is the typical use case or is there a use =
case where the inserted data is not local and may arrive as RTP stream. =
In which case there is more than one topology
>=20
> Roni
>=20
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: Thursday, October 21, 2010 7:23 PM
>> To: Roni Even
>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>> splicing-for-rtp-00
>>=20
>> Hi Roni,
>>=20
>> I think this problem becomes clearer if you consider a splicer that =
is
>> reading the content to be inserted from a local file on disk. In that
>> case, the splicer is unquestionably a standard RTP translator, in the
>> middle of a one-to-one or one-to-many RTP session.
>>=20
>> That there are also some cases where an entirely separate RTP session
>> is used to deliver the content to be inserted doesn't (conceptually)
>> affect the translation process at all.
>>=20
>> Colin
>>=20
>>=20
>>=20
>> On 21 Oct 2010, at 07:56, Roni Even wrote:
>>> Hi Colin,
>>> I am not so sure about this being a regular RTP translator. =
According
>> To RFC 5117 " A Translator always keeps the SSRC for a stream across
>> the translation". Also in the splicing case it is not point to point =
or
>> multipoint but more like multipoint to point or multipoint. This =
looks
>> more like a mixer that does not provide the CSRC since one of the
>> requirements is to make it look to the receiver that all the media is
>> coming from the same source. One issue that will need to be addressed
>> is how to report the RTCP RR from the receivers back to the sender of
>> the primary or insertion streams. This may need some state keeping in
>> the splicer.
>>>=20
>>> Roni Even
>>>=20
>>>> -----Original Message-----
>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
>> Of
>>>> Colin Perkins
>>>> Sent: Wednesday, October 20, 2010 3:52 PM
>>>> To: Jinwei Xia
>>>> Cc: 'IETF AVT WG'
>>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>>>> splicing-for-rtp-00
>>>>=20
>>>> Hi Jinwei,
>>>>=20
>>>> My impression is that this draft confuses the issue by focussing =
too
>> much on scenarios where the content to be inserted is also delivered
>> over RTP. Whether the content to be inserted is delivered as a real-
>> time RTP stream or fetched from local file storage does not affect =
the
>> insertion process, and so doesn't need to be mentioned in the draft. =
I
>> still believe that a standard RTP translator is sufficient to solve
>> this problem, although there may be useful things to say about how to
>> efficiently implement such a translator.
>>>>=20
>>>> Colin
>>>>=20
>>>>=20
>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
>>>>> Hi,
>>>>>=20
>>>>> =46rom the discussion on IETF 78th, there were some =
misunderstandings
>>>> of the scope of splicing draft. So we discard previous problem
>>>> statement draft while initiate new solution draft. In this new
>> draft,
>>>> We emphysize content insertion issue is indeed a RTP-generic rather
>>>> than MPEG2 TS-specific.
>>>>>=20
>>>>> Hope the new draft can address most issues AVT concern, any
>> comments
>>>> are very appreciated!
>>>>>=20
>>>>>=20
>>>>> BR!
>>>>>=20
>>>>> Jinwei
>>>>>=20
>>>>>> -----Original Message-----
>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
>>>>>> To: xiajinwei@huawei.com
>>>>>> Subject: New Version Notification for
>>>>>> draft-xia-avt-splicing-for-rtp-00
>>>>>>=20
>>>>>>=20
>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
>>>>>> has been successfully submitted by Jinwei Xia and posted to
>>>>>> the IETF repository.
>>>>>>=20
>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
>>>>>> Revision:	 00
>>>>>> Title:		 Content Splicing for RTP Sessions
>>>>>> Creation_date:	 2010-10-16
>>>>>> WG ID:		 Independent Submission
>>>>>> Number_of_pages: 12
>>>>>>=20
>>>>>> Abstract:
>>>>>> This memo outlines RTP splicing.  Splicing is a process that
>>>>>> allows a new multimedia stream to be inserted into current
>>>>>> multimedia stream and to be conveyed to receiver for a period
>>>>>> of time.  This memo discusses the requirements of RTP
>>>>>> splicing.  In order to satisfy the requirements, this memo
>>>>>> lists several existing intermediary network elements as
>>>>>> alternatives and evaluates whether one of alternatives can be
>>>>>> used to perform RTP splicing.
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> The IETF Secretariat.
>>>>>>=20
>>>>>>=20
>>>>>=20
>>>>> _______________________________________________
>>>>> Audio/Video Transport Working Group
>>>>> avt@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>=20
>>>> --
>>>> Colin Perkins
>>>> http://csperkins.org/
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From peter.musgrave@magorcorp.com  Thu Oct 21 13:20:01 2010
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Subject: [AVT] Errata Updates for AVT RFC4696/RFC5109
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--00163630f66dc6e168049326457c
Content-Type: text/plain; charset=ISO-8859-1

Hi Rob/AVT,

Here are my notes on some of the reported errata:

RFC4695
http://www.rfc-editor.org/errata_search.php?eid=861
Errata correctly notes error in ASCII characters for P and Q
Action:Hold For Doc Update (unlikely to stump anyone implementing)

RFC5109
http://www.rfc-editor.org/errata_search.php?eid=1225
Editorial: clarrifies a sentence in the introduction
Action: Hold for doc update

http://www.rfc-editor.org/errata_search.php?eid=1226
Editorial: rephrases a step in the instructions for RTP header
reconstruction
Action: Rejected (In my opinion the new text is not significantly clearer
and the use of the word cumulative is a bit confusing).

http://www.rfc-editor.org/errata_search.php?eid=1227
Editorial: clarifies and adds a useful internal reference
Action: Hold for Doc Update

http://www.rfc-editor.org/errata_search.php?eid=1228
Editorial: Correct figure to indicate RTP header is 12 octets (6 is shown)
Action: Hold for Doc Update

http://www.rfc-editor.org/errata_search.php?eid=1229
Editorial: Correct figure to indicate RTP header is 12 octets (6 is shown)
Action: Hold for Doc Update

http://www.rfc-editor.org/errata_search.php?eid=1230
Editorial: An error in an example with RTP head as 6 octets and a
mis-ordering of one header field
Action: Hold for Doc Update

Regards,

Peter Musgrave

--00163630f66dc6e168049326457c
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Hi Rob/AVT,=A0<div><br></div><div>Here are my notes on some of the reported=
 errata:</div><div><br></div><div>RFC4695</div><div><a href=3D"http://www.r=
fc-editor.org/errata_search.php?eid=3D861">http://www.rfc-editor.org/errata=
_search.php?eid=3D861</a></div>
<div>Errata correctly notes error in ASCII characters for P and Q</div><div=
>Action:Hold For Doc Update (unlikely to stump anyone implementing)</div><d=
iv><br></div><div>RFC5109</div><div><a href=3D"http://www.rfc-editor.org/er=
rata_search.php?eid=3D1225">http://www.rfc-editor.org/errata_search.php?eid=
=3D1225</a></div>
<div>Editorial: clarrifies a sentence in the introduction</div><div>Action:=
 Hold for doc update</div><div><br></div><div><a href=3D"http://www.rfc-edi=
tor.org/errata_search.php?eid=3D1226">http://www.rfc-editor.org/errata_sear=
ch.php?eid=3D1226</a></div>
<div>Editorial: rephrases a step in the instructions for RTP header reconst=
ruction=A0</div><div>Action: Rejected (In my opinion the new text is not si=
gnificantly clearer and the use of the word cumulative is a bit confusing).=
=A0</div>
<div><br></div><div><div><a href=3D"http://www.rfc-editor.org/errata_search=
.php?eid=3D1227">http://www.rfc-editor.org/errata_search.php?eid=3D1227</a>=
</div><div>Editorial: clarifies and adds a useful internal reference</div><=
/div>
<div>Action: Hold for Doc Update</div><div><br></div><div><div><a href=3D"h=
ttp://www.rfc-editor.org/errata_search.php?eid=3D1228">http://www.rfc-edito=
r.org/errata_search.php?eid=3D1228</a></div></div><div>Editorial: Correct f=
igure to indicate RTP header is 12 octets (6 is shown)</div>
<div>Action: Hold for Doc Update</div><div><br></div><div><div><div><a href=
=3D"http://www.rfc-editor.org/errata_search.php?eid=3D1229">http://www.rfc-=
editor.org/errata_search.php?eid=3D1229</a></div></div><div>Editorial: Corr=
ect figure to indicate RTP header is 12 octets (6 is shown)</div>
<div>Action: Hold for Doc Update</div></div><div><br></div><div><a href=3D"=
http://www.rfc-editor.org/errata_search.php?eid=3D1230">http://www.rfc-edit=
or.org/errata_search.php?eid=3D1230</a></div><div>Editorial: An error in an=
 example with RTP head as 6 octets and a mis-ordering of one header field</=
div>
<div>Action: Hold for Doc Update</div><div><br></div><div>Regards,=A0</div>=
<div><br></div><div>Peter Musgrave</div><div><br></div><div><br></div><div>=
<br></div><div><br></div><div><br></div><div><br></div>

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Cc: 'IETF AVT WG' <avt@ietf.org>, Roni Even <Even.roni@huawei.com>
Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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On Oct 21, 2010, at 4:17 PM, Colin Perkins wrote:

> Roni,
>=20
> I don't think it matters that the content arrives as an RTP stream. =
It's an entirely separate stream, and doesn't affect the translation =
process. They're logically separate.
>=20
I agree with Colin here.=20

Either you treat the streams as equal inputs to a mixer, and the case =
reduces to the problem of a video mixer doing input switching, or you =
treat all the other inputs as external stuff that gets inserted into the =
main stream using a translator.

If you choose the mixer approach, you have to be a legal mixer, which =
means you keep state, use your own SSRC, and report CSRCs.

If you choose the translator approach, you have to "consume" the other =
stream and then emit it as if it were emitted by the original stream's =
source.

I don't see what the other approaches would be or how they might work =
better.

DaveO.

> Colin
>=20
>=20
>=20
> On 21 Oct 2010, at 19:26, Roni Even wrote:
>> Hi Colin,
>> In this use case the inserted content is not an RTP stream but comes =
from a local storage. For this use case the splicer is a translator. It =
will still need to handle the RTCP RR depending on who originated the =
data that was sent by the translator.
>> My question will be if this is the typical use case or is there a use =
case where the inserted data is not local and may arrive as RTP stream. =
In which case there is more than one topology
>>=20
>> Roni
>>=20
>>> -----Original Message-----
>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>> Sent: Thursday, October 21, 2010 7:23 PM
>>> To: Roni Even
>>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>>> splicing-for-rtp-00
>>>=20
>>> Hi Roni,
>>>=20
>>> I think this problem becomes clearer if you consider a splicer that =
is
>>> reading the content to be inserted from a local file on disk. In =
that
>>> case, the splicer is unquestionably a standard RTP translator, in =
the
>>> middle of a one-to-one or one-to-many RTP session.
>>>=20
>>> That there are also some cases where an entirely separate RTP =
session
>>> is used to deliver the content to be inserted doesn't (conceptually)
>>> affect the translation process at all.
>>>=20
>>> Colin
>>>=20
>>>=20
>>>=20
>>> On 21 Oct 2010, at 07:56, Roni Even wrote:
>>>> Hi Colin,
>>>> I am not so sure about this being a regular RTP translator. =
According
>>> To RFC 5117 " A Translator always keeps the SSRC for a stream across
>>> the translation". Also in the splicing case it is not point to point =
or
>>> multipoint but more like multipoint to point or multipoint. This =
looks
>>> more like a mixer that does not provide the CSRC since one of the
>>> requirements is to make it look to the receiver that all the media =
is
>>> coming from the same source. One issue that will need to be =
addressed
>>> is how to report the RTCP RR from the receivers back to the sender =
of
>>> the primary or insertion streams. This may need some state keeping =
in
>>> the splicer.
>>>>=20
>>>> Roni Even
>>>>=20
>>>>> -----Original Message-----
>>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
>>> Of
>>>>> Colin Perkins
>>>>> Sent: Wednesday, October 20, 2010 3:52 PM
>>>>> To: Jinwei Xia
>>>>> Cc: 'IETF AVT WG'
>>>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>>>>> splicing-for-rtp-00
>>>>>=20
>>>>> Hi Jinwei,
>>>>>=20
>>>>> My impression is that this draft confuses the issue by focussing =
too
>>> much on scenarios where the content to be inserted is also delivered
>>> over RTP. Whether the content to be inserted is delivered as a real-
>>> time RTP stream or fetched from local file storage does not affect =
the
>>> insertion process, and so doesn't need to be mentioned in the draft. =
I
>>> still believe that a standard RTP translator is sufficient to solve
>>> this problem, although there may be useful things to say about how =
to
>>> efficiently implement such a translator.
>>>>>=20
>>>>> Colin
>>>>>=20
>>>>>=20
>>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
>>>>>> Hi,
>>>>>>=20
>>>>>> =46rom the discussion on IETF 78th, there were some =
misunderstandings
>>>>> of the scope of splicing draft. So we discard previous problem
>>>>> statement draft while initiate new solution draft. In this new
>>> draft,
>>>>> We emphysize content insertion issue is indeed a RTP-generic =
rather
>>>>> than MPEG2 TS-specific.
>>>>>>=20
>>>>>> Hope the new draft can address most issues AVT concern, any
>>> comments
>>>>> are very appreciated!
>>>>>>=20
>>>>>>=20
>>>>>> BR!
>>>>>>=20
>>>>>> Jinwei
>>>>>>=20
>>>>>>> -----Original Message-----
>>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
>>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
>>>>>>> To: xiajinwei@huawei.com
>>>>>>> Subject: New Version Notification for
>>>>>>> draft-xia-avt-splicing-for-rtp-00
>>>>>>>=20
>>>>>>>=20
>>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
>>>>>>> has been successfully submitted by Jinwei Xia and posted to
>>>>>>> the IETF repository.
>>>>>>>=20
>>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
>>>>>>> Revision:	 00
>>>>>>> Title:		 Content Splicing for RTP Sessions
>>>>>>> Creation_date:	 2010-10-16
>>>>>>> WG ID:		 Independent Submission
>>>>>>> Number_of_pages: 12
>>>>>>>=20
>>>>>>> Abstract:
>>>>>>> This memo outlines RTP splicing.  Splicing is a process that
>>>>>>> allows a new multimedia stream to be inserted into current
>>>>>>> multimedia stream and to be conveyed to receiver for a period
>>>>>>> of time.  This memo discusses the requirements of RTP
>>>>>>> splicing.  In order to satisfy the requirements, this memo
>>>>>>> lists several existing intermediary network elements as
>>>>>>> alternatives and evaluates whether one of alternatives can be
>>>>>>> used to perform RTP splicing.
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>>=20
>>>>>>> The IETF Secretariat.
>>>>>>>=20
>>>>>>>=20
>>>>>>=20
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>>=20
>>>>> --
>>>>> Colin Perkins
>>>>> http://csperkins.org/
>>=20
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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From Even.roni@huawei.com  Thu Oct 21 13:49:00 2010
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Date: Thu, 21 Oct 2010 22:47:08 +0200
From: Roni Even <Even.roni@huawei.com>
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To: 'David R Oran' <oran@cisco.com>, 'Colin Perkins' <csp@csperkins.org>
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Subject: Re: [AVT] FW: New Version Notification for	draft-xia-avt-splicing-for-rtp-00
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Dave,
So the first question is if this is really one stream or is it two mixed
streams. My feeling was that it is two mixed streams and not one.

Now if it is two mixed streams than one of the requirements is that the
receiver will not identify which one is the inserted stream easily. So I am
not sure if CSRC is not contradictory.

I thought that we are talking here about a mixer but the case that Colin
provided when the inserted stream is local to splicer makes it more of a
translator except maybe the handling of the RTCP.

BTW: RTP translator and mixer are two of the topologies in RFC 5117, you can
also suggest that this is a Topo-RTCP-terminating-MCU doing video and audio
switch.

Roni Even



> -----Original Message-----
> From: David R Oran [mailto:oran@cisco.com]
> Sent: Thursday, October 21, 2010 10:27 PM
> To: Colin Perkins
> Cc: Roni Even; 'IETF AVT WG'
> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
> splicing-for-rtp-00
> 
> 
> On Oct 21, 2010, at 4:17 PM, Colin Perkins wrote:
> 
> > Roni,
> >
> > I don't think it matters that the content arrives as an RTP stream.
> It's an entirely separate stream, and doesn't affect the translation
> process. They're logically separate.
> >
> I agree with Colin here.
> 
> Either you treat the streams as equal inputs to a mixer, and the case
> reduces to the problem of a video mixer doing input switching, or you
> treat all the other inputs as external stuff that gets inserted into
> the main stream using a translator.
> 
> If you choose the mixer approach, you have to be a legal mixer, which
> means you keep state, use your own SSRC, and report CSRCs.
> 
> If you choose the translator approach, you have to "consume" the other
> stream and then emit it as if it were emitted by the original stream's
> source.
> 
> I don't see what the other approaches would be or how they might work
> better.
> 
> DaveO.
> 
> > Colin
> >
> >
> >
> > On 21 Oct 2010, at 19:26, Roni Even wrote:
> >> Hi Colin,
> >> In this use case the inserted content is not an RTP stream but comes
> from a local storage. For this use case the splicer is a translator. It
> will still need to handle the RTCP RR depending on who originated the
> data that was sent by the translator.
> >> My question will be if this is the typical use case or is there a
> use case where the inserted data is not local and may arrive as RTP
> stream. In which case there is more than one topology
> >>
> >> Roni
> >>
> >>> -----Original Message-----
> >>> From: Colin Perkins [mailto:csp@csperkins.org]
> >>> Sent: Thursday, October 21, 2010 7:23 PM
> >>> To: Roni Even
> >>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
> >>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
> >>> splicing-for-rtp-00
> >>>
> >>> Hi Roni,
> >>>
> >>> I think this problem becomes clearer if you consider a splicer that
> is
> >>> reading the content to be inserted from a local file on disk. In
> that
> >>> case, the splicer is unquestionably a standard RTP translator, in
> the
> >>> middle of a one-to-one or one-to-many RTP session.
> >>>
> >>> That there are also some cases where an entirely separate RTP
> session
> >>> is used to deliver the content to be inserted doesn't
> (conceptually)
> >>> affect the translation process at all.
> >>>
> >>> Colin
> >>>
> >>>
> >>>
> >>> On 21 Oct 2010, at 07:56, Roni Even wrote:
> >>>> Hi Colin,
> >>>> I am not so sure about this being a regular RTP translator.
> According
> >>> To RFC 5117 " A Translator always keeps the SSRC for a stream
> across
> >>> the translation". Also in the splicing case it is not point to
> point or
> >>> multipoint but more like multipoint to point or multipoint. This
> looks
> >>> more like a mixer that does not provide the CSRC since one of the
> >>> requirements is to make it look to the receiver that all the media
> is
> >>> coming from the same source. One issue that will need to be
> addressed
> >>> is how to report the RTCP RR from the receivers back to the sender
> of
> >>> the primary or insertion streams. This may need some state keeping
> in
> >>> the splicer.
> >>>>
> >>>> Roni Even
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
> Behalf
> >>> Of
> >>>>> Colin Perkins
> >>>>> Sent: Wednesday, October 20, 2010 3:52 PM
> >>>>> To: Jinwei Xia
> >>>>> Cc: 'IETF AVT WG'
> >>>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-
> avt-
> >>>>> splicing-for-rtp-00
> >>>>>
> >>>>> Hi Jinwei,
> >>>>>
> >>>>> My impression is that this draft confuses the issue by focussing
> too
> >>> much on scenarios where the content to be inserted is also
> delivered
> >>> over RTP. Whether the content to be inserted is delivered as a
> real-
> >>> time RTP stream or fetched from local file storage does not affect
> the
> >>> insertion process, and so doesn't need to be mentioned in the
> draft. I
> >>> still believe that a standard RTP translator is sufficient to solve
> >>> this problem, although there may be useful things to say about how
> to
> >>> efficiently implement such a translator.
> >>>>>
> >>>>> Colin
> >>>>>
> >>>>>
> >>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> >>>>>> Hi,
> >>>>>>
> >>>>>> From the discussion on IETF 78th, there were some
> misunderstandings
> >>>>> of the scope of splicing draft. So we discard previous problem
> >>>>> statement draft while initiate new solution draft. In this new
> >>> draft,
> >>>>> We emphysize content insertion issue is indeed a RTP-generic
> rather
> >>>>> than MPEG2 TS-specific.
> >>>>>>
> >>>>>> Hope the new draft can address most issues AVT concern, any
> >>> comments
> >>>>> are very appreciated!
> >>>>>>
> >>>>>>
> >>>>>> BR!
> >>>>>>
> >>>>>> Jinwei
> >>>>>>
> >>>>>>> -----Original Message-----
> >>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> >>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
> >>>>>>> To: xiajinwei@huawei.com
> >>>>>>> Subject: New Version Notification for
> >>>>>>> draft-xia-avt-splicing-for-rtp-00
> >>>>>>>
> >>>>>>>
> >>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
> >>>>>>> has been successfully submitted by Jinwei Xia and posted to
> >>>>>>> the IETF repository.
> >>>>>>>
> >>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
> >>>>>>> Revision:	 00
> >>>>>>> Title:		 Content Splicing for RTP Sessions
> >>>>>>> Creation_date:	 2010-10-16
> >>>>>>> WG ID:		 Independent Submission
> >>>>>>> Number_of_pages: 12
> >>>>>>>
> >>>>>>> Abstract:
> >>>>>>> This memo outlines RTP splicing.  Splicing is a process that
> >>>>>>> allows a new multimedia stream to be inserted into current
> >>>>>>> multimedia stream and to be conveyed to receiver for a period
> >>>>>>> of time.  This memo discusses the requirements of RTP
> >>>>>>> splicing.  In order to satisfy the requirements, this memo
> >>>>>>> lists several existing intermediary network elements as
> >>>>>>> alternatives and evaluates whether one of alternatives can be
> >>>>>>> used to perform RTP splicing.
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> The IETF Secretariat.
> >>>>>>>
> >>>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group
> >>>>>> avt@ietf.org
> >>>>>> https://www.ietf.org/mailman/listinfo/avt
> >>>>>
> >>>>> --
> >>>>> Colin Perkins
> >>>>> http://csperkins.org/
> >>
> >> _______________________________________________
> >> Audio/Video Transport Working Group
> >> avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> >
> >
> >
> > --
> > Colin Perkins
> > http://csperkins.org/
> >
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt


From john.lazzaro@gmail.com  Thu Oct 21 13:52:51 2010
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Hi Peter,

I'm maintaining the RFC 4695 bis I-D, and had a
question on your email that was CC'd to AVT:

On Thu, Oct 21, 2010 at 1:21 PM, Peter Musgrave
<peter.musgrave@magorcorp.com> wrote:
> RFC4695
> http://www.rfc-editor.org/errata_search.php?eid=861
> Errata correctly notes error in ASCII characters for P and Q
> Action:Hold For Doc Update (unlikely to stump anyone implementing)

Does this imply that you don't think we
should moving the RFC 4695 bis out
of the current cycle of "refresh the I-D
as-is every six months" mode at this time?

Or is this "action" orthogonal to the
decision to do a "Last Call" on the bis?

Thanks,

-- 
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
john [dot] lazzaro [at] gmail [dot] com

From peter.musgrave@magorcorp.com  Thu Oct 21 14:05:06 2010
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
Date: Thu, 21 Oct 2010 17:06:18 -0400
To: John Lazzaro <john.lazzaro@gmail.com>
Cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] Errata Updates for AVT RFC4696/RFC5109
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Hi John,

I was unaware of the bis.

This was part of an orthogonal effort to deal with reported errata backlog. I=
f it gets considered next time around I see no issues...

Regards

Peter Musgrave

Sent from my iPhone

On 2010-10-21, at 4:54 PM, John Lazzaro <john.lazzaro@gmail.com> wrote:

> Hi Peter,
>=20
> I'm maintaining the RFC 4695 bis I-D, and had a
> question on your email that was CC'd to AVT:
>=20
> On Thu, Oct 21, 2010 at 1:21 PM, Peter Musgrave
> <peter.musgrave@magorcorp.com> wrote:
>> RFC4695
>> http://www.rfc-editor.org/errata_search.php?eid=3D861
>> Errata correctly notes error in ASCII characters for P and Q
>> Action:Hold For Doc Update (unlikely to stump anyone implementing)
>=20
> Does this imply that you don't think we
> should moving the RFC 4695 bis out
> of the current cycle of "refresh the I-D
> as-is every six months" mode at this time?
>=20
> Or is this "action" orthogonal to the
> decision to do a "Last Call" on the bis?
>=20
> Thanks,
>=20
> --=20
> John Lazzaro
> http://www.cs.berkeley.edu/~lazzaro
> john [dot] lazzaro [at] gmail [dot] com

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On Oct 21, 2010, at 4:47 PM, Roni Even wrote:

> Dave,
> So the first question is if this is really one stream or is it two =
mixed
> streams. My feeling was that it is two mixed streams and not one.
>=20
> Now if it is two mixed streams than one of the requirements is that =
the
> receiver will not identify which one is the inserted stream easily.
So he can't skip ads, or something else?

> So I am
> not sure if CSRC is not contradictory.
>=20
Interesting conundrum, no?

> I thought that we are talking here about a mixer but the case that =
Colin
> provided when the inserted stream is local to splicer makes it more of =
a
> translator except maybe the handling of the RTCP.
>=20
> BTW: RTP translator and mixer are two of the topologies in RFC 5117, =
you can
> also suggest that this is a Topo-RTCP-terminating-MCU doing video and =
audio
> switch.
>=20
Right, in which case we're back to the CSRC-related conflicting goals.

> Roni Even
>=20
>=20
>=20
>> -----Original Message-----
>> From: David R Oran [mailto:oran@cisco.com]
>> Sent: Thursday, October 21, 2010 10:27 PM
>> To: Colin Perkins
>> Cc: Roni Even; 'IETF AVT WG'
>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>> splicing-for-rtp-00
>>=20
>>=20
>> On Oct 21, 2010, at 4:17 PM, Colin Perkins wrote:
>>=20
>>> Roni,
>>>=20
>>> I don't think it matters that the content arrives as an RTP stream.
>> It's an entirely separate stream, and doesn't affect the translation
>> process. They're logically separate.
>>>=20
>> I agree with Colin here.
>>=20
>> Either you treat the streams as equal inputs to a mixer, and the case
>> reduces to the problem of a video mixer doing input switching, or you
>> treat all the other inputs as external stuff that gets inserted into
>> the main stream using a translator.
>>=20
>> If you choose the mixer approach, you have to be a legal mixer, which
>> means you keep state, use your own SSRC, and report CSRCs.
>>=20
>> If you choose the translator approach, you have to "consume" the =
other
>> stream and then emit it as if it were emitted by the original =
stream's
>> source.
>>=20
>> I don't see what the other approaches would be or how they might work
>> better.
>>=20
>> DaveO.
>>=20
>>> Colin
>>>=20
>>>=20
>>>=20
>>> On 21 Oct 2010, at 19:26, Roni Even wrote:
>>>> Hi Colin,
>>>> In this use case the inserted content is not an RTP stream but =
comes
>> from a local storage. For this use case the splicer is a translator. =
It
>> will still need to handle the RTCP RR depending on who originated the
>> data that was sent by the translator.
>>>> My question will be if this is the typical use case or is there a
>> use case where the inserted data is not local and may arrive as RTP
>> stream. In which case there is more than one topology
>>>>=20
>>>> Roni
>>>>=20
>>>>> -----Original Message-----
>>>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>>>> Sent: Thursday, October 21, 2010 7:23 PM
>>>>> To: Roni Even
>>>>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
>>>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-avt-
>>>>> splicing-for-rtp-00
>>>>>=20
>>>>> Hi Roni,
>>>>>=20
>>>>> I think this problem becomes clearer if you consider a splicer =
that
>> is
>>>>> reading the content to be inserted from a local file on disk. In
>> that
>>>>> case, the splicer is unquestionably a standard RTP translator, in
>> the
>>>>> middle of a one-to-one or one-to-many RTP session.
>>>>>=20
>>>>> That there are also some cases where an entirely separate RTP
>> session
>>>>> is used to deliver the content to be inserted doesn't
>> (conceptually)
>>>>> affect the translation process at all.
>>>>>=20
>>>>> Colin
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> On 21 Oct 2010, at 07:56, Roni Even wrote:
>>>>>> Hi Colin,
>>>>>> I am not so sure about this being a regular RTP translator.
>> According
>>>>> To RFC 5117 " A Translator always keeps the SSRC for a stream
>> across
>>>>> the translation". Also in the splicing case it is not point to
>> point or
>>>>> multipoint but more like multipoint to point or multipoint. This
>> looks
>>>>> more like a mixer that does not provide the CSRC since one of the
>>>>> requirements is to make it look to the receiver that all the media
>> is
>>>>> coming from the same source. One issue that will need to be
>> addressed
>>>>> is how to report the RTCP RR from the receivers back to the sender
>> of
>>>>> the primary or insertion streams. This may need some state keeping
>> in
>>>>> the splicer.
>>>>>>=20
>>>>>> Roni Even
>>>>>>=20
>>>>>>> -----Original Message-----
>>>>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
>> Behalf
>>>>> Of
>>>>>>> Colin Perkins
>>>>>>> Sent: Wednesday, October 20, 2010 3:52 PM
>>>>>>> To: Jinwei Xia
>>>>>>> Cc: 'IETF AVT WG'
>>>>>>> Subject: Re: [AVT] FW: New Version Notification for draft-xia-
>> avt-
>>>>>>> splicing-for-rtp-00
>>>>>>>=20
>>>>>>> Hi Jinwei,
>>>>>>>=20
>>>>>>> My impression is that this draft confuses the issue by focussing
>> too
>>>>> much on scenarios where the content to be inserted is also
>> delivered
>>>>> over RTP. Whether the content to be inserted is delivered as a
>> real-
>>>>> time RTP stream or fetched from local file storage does not affect
>> the
>>>>> insertion process, and so doesn't need to be mentioned in the
>> draft. I
>>>>> still believe that a standard RTP translator is sufficient to =
solve
>>>>> this problem, although there may be useful things to say about how
>> to
>>>>> efficiently implement such a translator.
>>>>>>>=20
>>>>>>> Colin
>>>>>>>=20
>>>>>>>=20
>>>>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
>>>>>>>> Hi,
>>>>>>>>=20
>>>>>>>> =46rom the discussion on IETF 78th, there were some
>> misunderstandings
>>>>>>> of the scope of splicing draft. So we discard previous problem
>>>>>>> statement draft while initiate new solution draft. In this new
>>>>> draft,
>>>>>>> We emphysize content insertion issue is indeed a RTP-generic
>> rather
>>>>>>> than MPEG2 TS-specific.
>>>>>>>>=20
>>>>>>>> Hope the new draft can address most issues AVT concern, any
>>>>> comments
>>>>>>> are very appreciated!
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> BR!
>>>>>>>>=20
>>>>>>>> Jinwei
>>>>>>>>=20
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
>>>>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
>>>>>>>>> To: xiajinwei@huawei.com
>>>>>>>>> Subject: New Version Notification for
>>>>>>>>> draft-xia-avt-splicing-for-rtp-00
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
>>>>>>>>> has been successfully submitted by Jinwei Xia and posted to
>>>>>>>>> the IETF repository.
>>>>>>>>>=20
>>>>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
>>>>>>>>> Revision:	 00
>>>>>>>>> Title:		 Content Splicing for RTP Sessions
>>>>>>>>> Creation_date:	 2010-10-16
>>>>>>>>> WG ID:		 Independent Submission
>>>>>>>>> Number_of_pages: 12
>>>>>>>>>=20
>>>>>>>>> Abstract:
>>>>>>>>> This memo outlines RTP splicing.  Splicing is a process that
>>>>>>>>> allows a new multimedia stream to be inserted into current
>>>>>>>>> multimedia stream and to be conveyed to receiver for a period
>>>>>>>>> of time.  This memo discusses the requirements of RTP
>>>>>>>>> splicing.  In order to satisfy the requirements, this memo
>>>>>>>>> lists several existing intermediary network elements as
>>>>>>>>> alternatives and evaluates whether one of alternatives can be
>>>>>>>>> used to perform RTP splicing.
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>> The IETF Secretariat.
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> _______________________________________________
>>>>>>>> Audio/Video Transport Working Group
>>>>>>>> avt@ietf.org
>>>>>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>>>>=20
>>>>>>> --
>>>>>>> Colin Perkins
>>>>>>> http://csperkins.org/
>>>>=20
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avt
>>>=20
>>>=20
>>>=20
>>> --
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>=20


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From Bing.He@alcatel-sbell.com.cn  Thu Oct 21 17:58:25 2010
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Thread-Topic: RFC4733/2833 related: Should volume field value be used in regenerating DTMF tone?
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From: "HE Bing" <Bing.He@alcatel-sbell.com.cn>
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Cc: XU Hui N <Hui.n.Xu@alcatel-sbell.com.cn>, Felizardo Jose <Jose.Felizardo@alcatel-lucent.com>
Subject: [AVT] RFC4733/2833 related: Should volume field value be used in regenerating DTMF tone?
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Hello Mr Schulzrinne, et. all

We got a different understanding from our DSP vendor regrading the DTMF
relay based on NTE packet defined in RFC 2833 and later 4733, thus we
need your help for clarification.

Our DSP vendor uses a hardcoded power level instead of the valid value
carried by the volume field in NTE packet and declare this does not
violate RFC 2833 and RFC 4733 due to no explicit statement telling that
value SHALL or SHOULD be used.

We can say nothing bu we do think the volume field value SHALL be used
in generating DTMF tone as long as it is valid. Each DSP vendor may have
different implementation, but if support of RFC 2833 or 4733 is
declared, use of volume field has to be supported at least as an option.
We already run into problem in a customer project where the terminal
expect DTMF signal with power level higher than the hardcoded value.

Could you share with your opinion? If the volume field is not used in
reconstructing the relayed DTMF and other tone, then any other
motivation to design such a field within NTE packet?

Thanks a lot for your clarification!
Best regard.
He Bing

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<TITLE>RFC4733/2833 related: Should volume field value be used in =
regenerating DTMF tone?</TITLE>
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<BODY>
<!-- Converted from text/rtf format -->

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">Hello Mr =
Schulzrinne, et. all</FONT>
</P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">We got a =
different understanding from our DSP vendor regrading the DTMF relay =
based on NTE packet defined in RFC 2833 and later 4733, thus we need =
your help for clarification.</FONT></P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">Our DSP vendor =
uses a hardcoded power level instead of the valid value carried by the =
volume field in NTE packet and declare this does not violate RFC 2833 =
and RFC 4733 due to no explicit statement telling that value SHALL or =
SHOULD be used.</FONT></P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">We can say =
nothing bu we do think the volume field value SHALL be used in =
generating DTMF tone as long as it is valid. Each DSP vendor may have =
different implementation, but if support of RFC 2833 or 4733 is =
declared, use of volume field has to be supported at least as an =
option.</FONT></P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">We already run =
into problem in a customer project where the terminal expect DTMF signal =
with power level higher than the hardcoded value.</FONT></P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">Could you =
share with your opinion? If the volume field is not used in =
reconstructing the relayed DTMF and other tone, then any other =
motivation to design such a field within NTE packet?</FONT></P>

<P><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">Thanks a lot =
for your clarification!</FONT>

<BR><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">Best =
regard.</FONT>

<BR><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Trebuchet MS">He =
Bing</FONT>
</P>

</BODY>
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From yangpeilin@huawei.com  Thu Oct 21 18:48:37 2010
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From: Peilin YANG <yangpeilin@huawei.com>
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Hi=2C

Please check the attached -00 draft on =22Switching from unicast to multi=
cast for multicast short time-shift=22=2E Also=2C URI is as below=3A
http=3A//www=2Eietf=2Eorg/id/draft-yang-avt-switch-multicast-short-timesh=
ift-00=2Etxt

Any comments are welcome=2E =


Best regards=2C

Peilin


----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A IETF I-D Submission Tool =3Cidsubmission=40ietf=2Eo=
rg=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=D2=BB=2C =CA=AE=D4=C2 18=C8=D5=2C 2010 =CF=C2=
=CE=E74=3A37
=D6=F7=CC=E2=3A New Version Notification for draft-yang-avt-switch-multic=
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=CA=D5=BC=FE=C8=CB=3A yangpeilin=40huawei=2Ecom
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-ftgroup=2Ecom=2C tena=40huawei=2Ecom=2C guyingjie=40huawei=2Ecom

A new version of I-D=2C draft-yang-avt-switch-multicast-short-timeshift-0=
0=2Etxt has been successfully submitted by Peilin Yang and posted to the =
IETF repository=2E

Filename=3A         draft-yang-avt-switch-multicast-short-timeshift
Revision=3A         00
Title=3A                 Switching from unicast to multicast for multicas=
t short time-shift
Creation=5Fdate=3A         2010-10-18
WG ID=3A                 Independent Submission
Number=5Fof=5Fpages=3A 11

Abstract=3A
When a client requests a time-shift service for a multicast session
using RTP for media transport=2C like pause=2C instant replay=2C slow-
motion video=2C frame-by-frame viewing=2C rewind=2C fast-forward=2C etc=2E=
=2C it
needs to switch from multicast session to unicast session=2E  This
unicast session will always serve for the time-shift service unless
the client manually switches back to the multicast session=2E  Actually
a time- shift request may happens for all clients=2E  That is=2C
multicast session stream will be replaced by many unicast time-shift
streams having significant impact on network bandwidth usage=2E

In this document=2C we describe a method using existing RTP and RTCP
protocol infrastructure that switches back to multicast session from
unicast session of time-shift=2E  In this method=2C a burst RTP flow
which is a little faster than the primary multicast rate may be
transmitted so that unicast session may catch up and switch back to
multicast session=2E
                                                                         =
         =



The IETF Secretariat=2E


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Audio/Video Transport                                            P. Yang
Internet-Draft                                                   R. Even
Intended status: Standards Track           Huawei Technologies Co., Ltd.
Expires: April 21, 2011                                      H. Moustafa
                                                 France Telecom - Orange
                                                        October 18, 2010


   Switching from unicast to multicast for multicast short time-shift
         draft-yang-avt-switch-multicast-short-timeshift-00.txt

Abstract

   When a client requests a time-shift service for a multicast session
   using RTP for media transport, like pause, instant replay, slow-
   motion video, frame-by-frame viewing, rewind, fast-forward, etc., it
   needs to switch from multicast session to unicast session.  This
   unicast session will always serve for the time-shift service unless
   the client manually switches back to the multicast session.  Actually
   a time- shift request may happens for all clients.  That is,
   multicast session stream will be replaced by many unicast time-shift
   streams having significant impact on network bandwidth usage.

   In this document, we describe a method using existing RTP and RTCP
   protocol infrastructure that switches back to multicast session from
   unicast session of time-shift.  In this method, a burst RTP flow
   which is a little faster than the primary multicast rate may be
   transmitted so that unicast session may catch up and switch back to
   multicast session.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 21, 2011.

Copyright Notice



Yang, et al.             Expires April 21, 2011                 [Page 1]

Internet-Draft  Switching for multicast short time-shift    October 2010


   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  RTSP Time-shift and MST Reverse Switch . . . . . . . . . .  5
     1.2.  RAMS and MST Reverse Switch  . . . . . . . . . . . . . . .  6
   2.  Conventions  . . . . . . . . . . . . . . . . . . . . . . . . .  6
   3.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  6
   4.  Reverse switch for multicast short time-shift  . . . . . . . .  7
     4.1.  Overview . . . . . . . . . . . . . . . . . . . . . . . . .  7
     4.2.  Message Flows  . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10
   8.  Normative References . . . . . . . . . . . . . . . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 10






















Yang, et al.             Expires April 21, 2011                 [Page 2]

Internet-Draft  Switching for multicast short time-shift    October 2010


1.  Introduction

   Interactive time-shift is an important service for media application.
   Through time-shift control operations (e.g. pause, instant replay,
   slow-motion video, frame-by-frame viewing, rewind and fast-forward),
   viewers can access recorded programming and live streams.  The time-
   shift service is available for recorded media and real-time
   streaming.  For the real-time streaming, time-shift service requires
   recording/caching of the information.  The time-shifted information
   can be sent by the original media source or by a server in the
   network that caches the stream and provides the time-shift service.

   Note that time-shift service may run on the client side but this
   requires the client to be able to cache the stream and
   synchronization to the main multicast stream needs to be managed by
   the client and does not require any specific protocol.

   This document looks at time-shift services where the original content
   is delivered using multicast.  In this use case the time-shift
   service is using a unicast stream from a Multicast Short Time-shift
   (MST) server to the client.  The synchronization between the
   multicast and unicast stream can be achieved by the client if he is
   using control commands like fast forward or RTSP [ref] Scale request.
   The MST can notice that the client is receiving in the unicast stream
   the same information that is current in the multicast stream.

   Network traffic will rise with the increase in the number of clients
   using time-shift service, because the time-shift service traffic
   changes from a multicast stream (one multicast stream for all
   clients) to a large number of unicast streams (one unicast stream per
   client).  Time- shift services like slow-motion view, instant replay,
   frame-by-frame viewing, pause and rewind may often occur for most of
   viewers in prime-time (e.g. when watching sports events).

   We can separate the time-shift services of primary live streams
   delivered over multicast/broadcast into multicast long time-shift and
   multicast short time-shift.  Some of the manipulations requested by
   the viewers are multicast short time-shift (e.g. short pause, instant
   replay, slow-motion video and frame-by-frame viewing).  For the
   multicast short time-shift, the time offset between the unicast time-
   shift playback, after the time-shift request, and the current time of
   the primary multicast stream is small enabling the unicast stream to
   catch up with primary multicast stream by speeding up the playback,
   and then switch back to the original multicast session.

   In this document, we propose a solution for enabling the time-shifted
   stream receivers to catch up with the original multicast stream using
   the tools offered by the existing RTP and RTCP protocols.  The



Yang, et al.             Expires April 21, 2011                 [Page 3]

Internet-Draft  Switching for multicast short time-shift    October 2010


   document also describes how to switch back to the primary multicast
   session from the unicast session initiated by either the client or
   the server.  Using this solution allows the network bandwidth to be
   effectively saved for the time-shift service of multicast
   application.

   In the scenario that we consider in this draft, an intermediary
   network element (that we refer to as Multicast Short Time-shift
   server) joins the multicast session and continuously caches the
   multicast stream.  When an RTP receiver sends a time-shift request,
   the MST server starts a new unicast RTP session and transmits the
   unicast stream to the RTP receiver over that session.  A simplified
   network diagram showing this scenario employing an intermediary
   network element is shown in Figure 1, where the hashed lines show the
   unicast stream.
                  +--------------------------------+
                  |  Intermediary Network Element  |
                  |                                |
                  |          MST Server            |
                  +--------------------------------+
                                 ^ :
                                 | :
                                 | :
                                 | v
      +-----------+        +-------------+           +-------------+
      | Multicast |        |             |---------->|  Time-shift |
      |           |------->|   Router    |           |     RTP     |
      |  Source   |        |             |..........>|   Receiver  |
      +-----------+        +-------------+           +-------------+
                                   |
                                   |                 +-------------+
                                   |                 |   Existing  |
                                   +---------------->|     RTP     |
                                                     |   Receiver  |
                                                     +-------------+
         Figure 1: Reverse switch for multicast short time-shift
               through an intermediary network element

   The proposed solution is not dependent on a specific streaming
   control protocol like RTSP [ref].  It addresses the synchronization
   between a primary multicast RTP stream and a parallel time-shifted
   unicast RTP stream.  A principle design goal is to use an existing
   protocol to define this solution.  This improves the versatility of
   the existing implementations, and promotes faster deployment and
   better interoperability.  To this effect, the proposal is to use the
   switching of flows from unicast stream to multicast stream described
   in RAMS [draft-ietf-avt-rapid-acquisition-for-rtp], and use the
   capabilities of RTCP to handle the signaling needed to accomplish



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   this automatic reverse switch for multicast short time-shift.

1.1.  RTSP Time-shift and MST Reverse Switch

   RTSP 2.0[draft-ietf-mmusic-rfc2326bis] is an application-level
   protocol for setup and control of the media delivery with real-time
   properties, and provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, typically streaming
   media.

   RTSP defines any necessary media transport signalling with regards to
   RTP.  In appendix C.1 it defines the interaction of RTSP with respect
   to the RTP protocol [RFC3550].  Yet RTSP is not limited to RTP media
   delivery control but any delivery type of data.  It provides a
   general media delivery control mechanism to play out the media, pause
   media, or stop media over different delivery channels, such as UDP,
   multicast UDP, TCP, RTP over UDP, etc.

   RTSP can also provide interactive time-shift function with different
   scale and speed.  Section 13.4.8 in [draft-ietf-mmusic-rfc2326bis]
   has a scenario for Playing Live with time-shift where a certain media
   server may offer time-shift services to their clients.  The usage of
   this play method can implement time- shift for Live Media or On-
   demand Media.  Section 16.44 in [draft-ietf-mmusic-rfc2326bis]
   addresses the scaling for video and audio, it suggests sending only
   key frames for video but this may not achieve the right scale speed
   since it depends on the locality of such frames in the stream.  It
   also requires analysis of the media payload for all supported codecs
   to find the key frames.  In this document we call the interactive
   time-shift function in [draft-ietf-mmusic-rfc2326bis] as "RTSP Time-
   shift".

   For Live Media using RTP multicast over UDP, RTSP Time-shift can
   provide an initial switch from multicast session to unicast session
   when time-shift happens, but it cannot complete the reverse switch
   from unicast session of time-shift to the original multicast session.

   An MST server can for example transmit the unicast stream of the
   time-shift by a short burst stream and indicate to the receivers to
   speed up the playback through Non-perceptual Speedup Playback.  After
   the burst of the unicast stream of time-shift catches up with the
   primary multicast stream, the client can switch back to the primary
   multicast stream so that the network bandwidth can be effectively
   saved.  The synchronization due to catch up with the primary
   multicast session may also happen due to the client operation like
   the fast forward.





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1.2.  RAMS and MST Reverse Switch

   RAMS [draft-ietf-avt-rapid-acquisition-for-rtp] describes a method
   using the existing RTP and RTCP protocol for reducing the acquisition
   delay, allowing fast channel switch.  RAMS defines the reverse switch
   from unicast burst session to the primary multicast session.


2.  Conventions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].


3.  Definitions

   This document uses the following acronyms and definitions frequently:

   Time-shift Control: Refers to the manipulation control of time-shift
   service (e.g. pause, slow-motion video, frame-by-frame viewing,
   rewind, fast- forward, etc.), not include the play method.

   Time-shift Playback: Refers to the normal playback with a natural
   playback rate after the time-shift control.

   Multicast Time-shift Interval: Refers to the time interval or the
   number of packets between the primary multicast stream and the normal
   playback during the unicast session after the time-shift control.  In
   other words, it presents the time interval between the start point
   frame of the unicast session and the frame of the current location of
   the primary multicast session.

   Multicast Short Time-shift: The case when the multicast Time-shift
   Interval is short, such as 30 seconds.  The normal playback during
   the unicast session after the time-shift control means that viewers
   finish the time-shift control (e.g. pause, rewind and fast-
   forward)and start to receive the unicast stream and play back again
   at the speed of the Primary multicast stream.

   Primary multicast stream: The multicast stream before time-shift
   request.

   Instant Replay: It is the case of the replaying of video footage of
   an event or incident directly after its occurrence.  In television
   broadcasting of sports events, instant replay is often used during
   live broadcast, to show a passage of play which was important or
   remarkable, or which was unclear on the first sight.



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   Slow-motion Video: The case when the playback of a video clip appears
   to be slower than the natural speed of the events.

   Non-perceptual Speedup Playback: During the speedup playback, after
   each interval of some frames, one frame is skipped as if it was not
   present, the next frame is directly rendered to take the place of the
   skipped frame, while keeping a fixed Frame Per Second (FPS)during the
   playback speedup.


4.  Reverse switch for multicast short time-shift

   This section gives an overview on the proposed method for reverse
   multicast time- shift switch from unicast time-shift RTP Sessions.

4.1.  Overview

   RTP Control Protocol (RTCP) Extensions for Single-Source Multicast
   Sessions with Unicast Feedback [RFC5760] specifies an extension to
   the RTCP to use unicast feedback to a multicast sender.  The (Unicast
   RTCP) Feedback Target is a logical function to which RTCP unicast
   feedback traffic is addressed.  The functions of the Feedback Target
   and the Distribution Source MAY be co-located or integrated in the
   same entity.  In this case, The (Unicast RTCP) Feedback Target MAY be
   co-located or integrated in the Multicast Time-shift Server.

   This section presents a proposed method to switch back to the
   multicast session from the unicast session of time-shift considering
   multicast short time-shift.  The proposed method allows the network
   bandwidth to be effectively saved when an RTP receiver finishes its
   time-shift request and starts its time- shift playback by unicast
   stream.  A Multicast Short Time-shift Server (MST) and new RTCP
   feedback messages are also introduced for the proposed method.

   The MST server has a cache where the most recent packets from the
   primary multicast stream are continuously stored for some time.  When
   a viewer wants to normally playback the unicast stream after time-
   shift request, the RTP receiver needs to send a playback request to
   the feedback target, and then receives the unicast stream from the
   MST server.  In order to switch back to the original multicast
   stream, the MST server needs to transmit a burst unicast stream to
   RTP receivers.  Using an accelerated bitrate (as compared to the
   bitrate of the original multicast stream).  This means that at a
   certain point in time, the unicast burst will catch up with the
   original multicast stream.  At this point, the RTP receiver no longer
   needs to receive the unicast burst and can switch back to the
   original multicast session.




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   The transmission of the unicast burst stream of time-shift playback
   depends on the time-shift buffer size of RTP receivers and the
   Multicast Short Time-shift Interval.  In the case when the time-shift
   buffer size of an RTP receiver can inadequately accommodate the
   number of packets of the Multicast Short Time-shift Interval, the
   receiver needs to playback for the duration of the speeding-up until
   the remaining number of packets does not overflow at the time-shift
   buffer.  During the playback of the speeding-up, the receiver may
   speed up video playback by the non-perceptual speedup playback so
   that viewers could never perceived the existence of the speeding-up.
   Refer to [I-D.yang-avt-rtp-synced-playback]

4.2.  Message Flows

   This section introduces the different messages for the proposed
   method and the related messages flows.

   - MST(Multicast Short Time-shift) synchronization
   transmission(MSTST): This is the generic form of the different
   messages that will be transmitted during the proposed method in this
   draft for reverse switch multicast short time-shift.

   -Time-Shift-Control-Commands: This message presents the request for
   the Time-Shift service (e.g. fast forward, pause, reverse, ..etc.).
   This message may be addressed by the MST server or other time-shift
   server.

   - MSTST Playback Request(MSTST-PReq): It's the message sent from the
   RTP receiver to the MST to request the playback after the time-shift
   request.

   - MSTST Playback Response(MSTST-PRes): It's the response message sent
   from the MST to the RTP receiver to confirm the playback process
   initiation.  Following this message, the RTP receiver will receive
   the unicast burst RTP stream from the MST.

   - MSTST Synchronization Indication(MSTST-SInd): When the unicast
   burst catch up with the original multicast stream, this message is
   sent from the MST to the RTP receiver to indicate this fact.

   - MSTST Synchronization Response(MSTST-SRes): This message is sent by
   the RTP receiver as a response to the MST confirming the need to
   switch back to the original multicast session.  Following this
   message the RTP receiver is able to receive the multicast stream.

   - MSTST Termination Request(MSTST-TReq): This message is sent by the
   RTP receiver to the MST to terminate the process of the time-shift.




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   - MSTST Termination Response(MSTST-TRes): This message is a reply
   sent by the MST to the RTP to confirm to the MSTST-TReq.

                 -------------------------
                 |        MST  Server      |
    -----------  |  ------   ------------  |   --------    ------------
   | Multicast | | |  FT  | | Burst/Ret. | |  |        |  |    RTP     |
   |  Source   | | |      | |   Source   | |  | Router |  |  Receiver  |
   |           | |  ------   ------------  |  |        |  |  (RTP_Rx)  |
    -----------  |                         |   --------    ------------
         |        -------------------------       |             |
         |                   |                    |             |
         |--- RTP Multicast->-------------------->|             |
         |                   |                    |             |
         |             ................................................
         |             .                                              .
         |             .      <::Time-shift control commands ::>      .
         |             .       ..Time-shift media stream     ..>      .
         |             ................................................
         |                   |                    |             |
         |                   |<''''''''' RTCP MSTST-PReq''''''''|
         |                   |'''''''''' RTCP MSTST-PRes'''''''>|
         |                   |                    |             |
         |                   |                    |             |
         |                   |....... Unicast RTP Burst .......>|
         |                   |                    |             |
         |                   |                    |             |
         |                   |'''''''''' RTCP MSTST-SInd'''''''>|
         |                   |<''''''''' RTCP MSTST-SRes''''''''|
         |                   |                    |             |
         |                   |                    |             |
         |                   |                    |<~SFGMP Join~|
         |                   |                    |             |
         |                   |                    |             |
         |------------------ RTP Multicast -------------------->|
         |                   |                    |             |
         |                   |                    |             |
         |                   |<''''''''' RTCP MSTST-TReq''''''''|
         |                   |'''''''''' RTCP MSTST-TRes ''''''>|
         |                   |                    |             |
         |                   |                    |             |
         |                   |<''''''''' (RTCP BYE) ''''''''''''|
         |                   |                    |             |








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5.  Security Considerations

   TBD.


6.  IANA Considerations

   TBD.


7.  Acknowledgements

   TBD.


8.  Normative References

   [I-D.ietf-mmusic-rfc2326bis]
              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-24 (work in
              progress), July 2010.

   [I-D.yang-avt-rtp-synced-playback]
              Yang, P. and Y. Wang, "Synchronized Playback in Rapid
              Acquisition of Multicast Sessions",
              draft-yang-avt-rtp-synced-playback-04 (work in progress),
              March 2010.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.


Authors' Addresses

   Peilin Yang
   Huawei Technologies Co., Ltd.
   101 Software Avenue, Yuhua District, Nanjing 210012
   P.R.China

   Phone: +86-25-56622638
   Email: yangpeilin@huawei.com





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   Roni Even
   Huawei Technologies Co., Ltd.
   14 David Hamelech, Tel Aviv 64953
   Israel

   Email: even.roni@huawei.com


   Hassnaa Moustafa
   France Telecom - Orange
   38-40 rue du General Leclerc Issy Les Moulineaux, 92794 Cedex 9
   France

   Email: hassnaa.moustafa@orange-ftgroup.com


   Tina Tsou (editor)
   Huawei Technologies
   Section F, Huawei Industrial Base
   Bantian Longgang, Shenzhen  518129
   P.R. China

   Phone: +86 755 28972912
   Email: tena@huawei.com


   Gu Yingjie
   Huawei Technologies Co., Ltd.
   101 Software Avenue, Yuhua District, Nanjing 210012
   P.R.China

   Email: guyingjie@huawei.com



















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From xiajinwei@huawei.com  Thu Oct 21 20:28:52 2010
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Date: Fri, 22 Oct 2010 11:30:15 +0800
From: Jinwei Xia <xiajinwei@huawei.com>
In-reply-to: <72B2F4CF-DC7A-4A03-B358-276E8C5DECFE@cisco.com>
To: 'David R Oran' <oran@cisco.com>, 'Colin Perkins' <csp@csperkins.org>
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Cc: 'IETF AVT WG' <avt@ietf.org>, 'Roni Even' <Even.roni@huawei.com>
Subject: Re: [AVT] FW: New Version Notification	fordraft-xia-avt-splicing-for-rtp-00
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Hi Dave,

Please see inline. 

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of David R Oran
> Sent: Friday, October 22, 2010 4:27 AM
> To: Colin Perkins
> Cc: 'IETF AVT WG'; Roni Even
> Subject: Re: [AVT] FW: New Version Notification 
> fordraft-xia-avt-splicing-for-rtp-00
> 
> 
> On Oct 21, 2010, at 4:17 PM, Colin Perkins wrote:
> 
> > Roni,
> > 
> > I don't think it matters that the content arrives as an RTP 
> stream. It's an entirely separate stream, and doesn't affect 
> the translation process. They're logically separate.
> > 
> I agree with Colin here. 
> 
> Either you treat the streams as equal inputs to a mixer, and 
> the case reduces to the problem of a video mixer doing input 
> switching, or you treat all the other inputs as external 
> stuff that gets inserted into the main stream using a translator.
> 
> If you choose the mixer approach, you have to be a legal 
> mixer, which means you keep state, use your own SSRC, and 
> report CSRCs.

 A standard mixer may not work well, it mix all the streams rather than
select one of them.

> 
> If you choose the translator approach, you have to "consume" 
> the other stream and then emit it as if it were emitted by 
> the original stream's source.

Translator acts as a client and terminate inserted stream from inserted
stream source view compared to acting as translator from original stream
source view.  But the RTCP RR issue mentioned by Roni is unsolved, I also
mark this issue as TBD in draft.  

> 
> I don't see what the other approaches would be or how they 
> might work better.

MCU is not well defined in IETF, I agree to limit the scope in mixer and
translator.

> 
> DaveO.
> 
> > Colin
> > 
> > 
> > 
> > On 21 Oct 2010, at 19:26, Roni Even wrote:
> >> Hi Colin,
> >> In this use case the inserted content is not an RTP stream 
> but comes from a local storage. For this use case the splicer 
> is a translator. It will still need to handle the RTCP RR 
> depending on who originated the data that was sent by the translator.
> >> My question will be if this is the typical use case or is 
> there a use 
> >> case where the inserted data is not local and may arrive as RTP 
> >> stream. In which case there is more than one topology
> >> 
> >> Roni
> >> 
> >>> -----Original Message-----
> >>> From: Colin Perkins [mailto:csp@csperkins.org]
> >>> Sent: Thursday, October 21, 2010 7:23 PM
> >>> To: Roni Even
> >>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
> >>> Subject: Re: [AVT] FW: New Version Notification for 
> draft-xia-avt- 
> >>> splicing-for-rtp-00
> >>> 
> >>> Hi Roni,
> >>> 
> >>> I think this problem becomes clearer if you consider a 
> splicer that 
> >>> is reading the content to be inserted from a local file 
> on disk. In 
> >>> that case, the splicer is unquestionably a standard RTP 
> translator, 
> >>> in the middle of a one-to-one or one-to-many RTP session.
> >>> 
> >>> That there are also some cases where an entirely separate RTP 
> >>> session is used to deliver the content to be inserted doesn't 
> >>> (conceptually) affect the translation process at all.
> >>> 
> >>> Colin
> >>> 
> >>> 
> >>> 
> >>> On 21 Oct 2010, at 07:56, Roni Even wrote:
> >>>> Hi Colin,
> >>>> I am not so sure about this being a regular RTP translator. 
> >>>> According
> >>> To RFC 5117 " A Translator always keeps the SSRC for a 
> stream across 
> >>> the translation". Also in the splicing case it is not 
> point to point 
> >>> or multipoint but more like multipoint to point or 
> multipoint. This 
> >>> looks more like a mixer that does not provide the CSRC 
> since one of 
> >>> the requirements is to make it look to the receiver that all the 
> >>> media is coming from the same source. One issue that will 
> need to be 
> >>> addressed is how to report the RTCP RR from the receivers back to 
> >>> the sender of the primary or insertion streams. This may 
> need some 
> >>> state keeping in the splicer.
> >>>> 
> >>>> Roni Even
> >>>> 
> >>>>> -----Original Message-----
> >>>>> From: avt-bounces@ietf.org 
> [mailto:avt-bounces@ietf.org] On Behalf
> >>> Of
> >>>>> Colin Perkins
> >>>>> Sent: Wednesday, October 20, 2010 3:52 PM
> >>>>> To: Jinwei Xia
> >>>>> Cc: 'IETF AVT WG'
> >>>>> Subject: Re: [AVT] FW: New Version Notification for 
> draft-xia-avt- 
> >>>>> splicing-for-rtp-00
> >>>>> 
> >>>>> Hi Jinwei,
> >>>>> 
> >>>>> My impression is that this draft confuses the issue by 
> focussing 
> >>>>> too
> >>> much on scenarios where the content to be inserted is 
> also delivered 
> >>> over RTP. Whether the content to be inserted is delivered 
> as a real- 
> >>> time RTP stream or fetched from local file storage does 
> not affect 
> >>> the insertion process, and so doesn't need to be mentioned in the 
> >>> draft. I still believe that a standard RTP translator is 
> sufficient 
> >>> to solve this problem, although there may be useful things to say 
> >>> about how to efficiently implement such a translator.
> >>>>> 
> >>>>> Colin
> >>>>> 
> >>>>> 
> >>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
> >>>>>> Hi,
> >>>>>> 
> >>>>>> From the discussion on IETF 78th, there were some 
> >>>>>> misunderstandings
> >>>>> of the scope of splicing draft. So we discard previous problem 
> >>>>> statement draft while initiate new solution draft. In this new
> >>> draft,
> >>>>> We emphysize content insertion issue is indeed a RTP-generic 
> >>>>> rather than MPEG2 TS-specific.
> >>>>>> 
> >>>>>> Hope the new draft can address most issues AVT concern, any
> >>> comments
> >>>>> are very appreciated!
> >>>>>> 
> >>>>>> 
> >>>>>> BR!
> >>>>>> 
> >>>>>> Jinwei
> >>>>>> 
> >>>>>>> -----Original Message-----
> >>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> >>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
> >>>>>>> To: xiajinwei@huawei.com
> >>>>>>> Subject: New Version Notification for 
> >>>>>>> draft-xia-avt-splicing-for-rtp-00
> >>>>>>> 
> >>>>>>> 
> >>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
> >>>>>>> has been successfully submitted by Jinwei Xia and 
> posted to the 
> >>>>>>> IETF repository.
> >>>>>>> 
> >>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
> >>>>>>> Revision:	 00
> >>>>>>> Title:		 Content Splicing for RTP Sessions
> >>>>>>> Creation_date:	 2010-10-16
> >>>>>>> WG ID:		 Independent Submission
> >>>>>>> Number_of_pages: 12
> >>>>>>> 
> >>>>>>> Abstract:
> >>>>>>> This memo outlines RTP splicing.  Splicing is a process that 
> >>>>>>> allows a new multimedia stream to be inserted into current 
> >>>>>>> multimedia stream and to be conveyed to receiver for 
> a period of 
> >>>>>>> time.  This memo discusses the requirements of RTP 
> splicing.  In 
> >>>>>>> order to satisfy the requirements, this memo lists several 
> >>>>>>> existing intermediary network elements as alternatives and 
> >>>>>>> evaluates whether one of alternatives can be used to 
> perform RTP 
> >>>>>>> splicing.
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> The IETF Secretariat.
> >>>>>>> 
> >>>>>>> 
> >>>>>> 
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group avt@ietf.org 
> >>>>>> https://www.ietf.org/mailman/listinfo/avt
> >>>>> 
> >>>>> --
> >>>>> Colin Perkins
> >>>>> http://csperkins.org/
> >> 
> >> _______________________________________________
> >> Audio/Video Transport Working Group
> >> avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> > 
> > 
> > 
> > --
> > Colin Perkins
> > http://csperkins.org/
> > 
> > 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> 


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From: "Piers O'Hanlon" <p.ohanlon@cs.ucl.ac.uk>
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Subject: Re: [AVT] draft-ietf-avt-ecn-for-rtp-02 Other Tech Concerns
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Bob,

I took another look at things and realised there was an error in your
calculations regarding T6 - see below.

On 19 October 2010 15:45, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Hi Bob,
>
> See inline for my comments on the three issues.
>
> Bob Briscoe skrev 2010-08-06 00:50:
>> Magnus, Ingemar, Colin, Piers, Ken,
>>
>> Below are my main technical concerns with
>> draft-ietf-avt-ecn-for-rtp-02 as promised.
>> I have used the line numbering here:
>> <http://tools.ietf.org/idnits?url=3Dhttp://tools.ietf.org/id/draft-ietf-=
avt-ecn-for-rtp-02.txt>
>>
>> T1/ Combining pkts
>> T2/ ECN-IP-RTCP
>> T3/ Anti-Cheating
>> T4/ Logging Not-ECT fall-back =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 <<<---=
THIS EMAIL
>> T5/ Delivery %age ECT vs Not-ECT =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0<<<---TH=
IS EMAIL
>> T6/ Pkt Reordering around session start <<<---THIS EMAIL
>>
>> T4/ Not just fall-back to Not-ECT, but logging or alarms too.
>> -------------------
>> "
>> 783 =A0 =A0 =A0 =A0loss of all packets with ECT or ECN-CE markings. =A0I=
f the
>> path between
>> 784 =A0 =A0 =A0 =A0the sender and the receivers exhibits either of these=
 behaviours one
>> 785 =A0 =A0 =A0 =A0needs to stop using ECN immediately to protect both t=
he network and
>> 786 =A0 =A0 =A0 =A0the application.
>> "
>> "
>> 1332 =A0 =A0 =A0 Detecting clearing of ECN field:
>> 1337 =A0 =A0 =A0 Dropping of ECT packets:
>> "
>> As well as detecting ECN black holes and falling back to Not-ECT, the
>> source ought to log the IP address pair exhibiting this problem. This
>> could be used for later diagnostics, if someone ever tries to
>> systematically clear up this middlemess (e.g. in a corporate
>> environment, or anywhere that a network operator has access to these log=
s).
>> The logs might also be used as a database for selective leaps-of-faith.
>>
>> If an appropriate management alarm can be raised, it should be too.
>
> Agreed, I have no issues including that in the spec.
>
That's fine.

>>
>> T5/ Comparing Delivery percentage of ECT vs Not-ECT
>> -------------------
>> "
>> 1337 =A0 =A0 =A0 Dropping of ECT packets: To determine if the packet dro=
p ratio is
>> 1338 =A0 =A0 =A0 different between not-ECT and ECT marked transmission r=
equires a mix
>> 1339 =A0 =A0 =A0 of transmitted traffic. =A0The sender should compare if=
 the delivery
>> 1340 =A0 =A0 =A0 percentage (delivered / transmitted) between ECT and no=
t-ECT is
>> 1341 =A0 =A0 =A0 significantly different. =A0Care must be taken if the n=
umber
>> of packets
>> 1342 =A0 =A0 =A0 are low in either of the categories.
>> "
>> Eh? This surely won't work. Under normal circumstances (ie no
>> ECN-Blocking Muddleboxes) Not-ECT packets will be dropped more due to
>> congestion and ECT packets will hardly ever be dropped. So the sender
>> cannot detect if ECT packets are getting some extra drop due to
>> Muddleboxes, by comparing with Not-ECT which will have extra drop for
>> another valid reason (congestion).
>
> Well, that depends on the relative frequency between the ECT and non-ECT
> and the actual packet loss rates. But, yes I think you might have
> spotted a short coming in our thinking. But I think this is not
> completely wrong but is lacking the factor from CE marking, see below.
>
> A very direct method of resolving this issue is to use explicit loss
> vectors by using RTCP XR packet loss vectors. The ECN Nonce one doesn't
> separate between ECN-CE and packet loss and although possible requires a
> bit of going around to find the packet loss rate for ECT. What I can
> understand the sender would need to keep its actual transmission
> pattern. First count and eliminate the not-ect packets that are lost
> from the NONCE vector. Then the remaining marked in the NONCE vector are
> CE and packet loss of ECT packets. By removing the number of CE marked
> ones from the same report interval based on the ECN report blocks you
> would arrive at an approximate number of lost ECT packets. I say
> approximate, because you still have the reordering issue that make the
> sender uncertain on exactly which packets has been reported in the ECN
> report between two RTCP packets.
>
> But lets look at what level of statistics is needed to be able to draw
> some conclusions. I agree that for certain operation points there will
> be a clear difference between ECT and non-ECT traffic. But lets look at
> some examples:
>
> No congestion at all: NON-ECT drop rate should be 0. ECT on capable
> transport drop rate 0, one path(s) that are incapable there will be a
> drop rate >0.
>
> Low levels of congestion: Non-ECT drop rate at a low level (0.5%). ECT
> on capable transport drop rate 0, with CE marking at low level (0.5%),
> ECT on incapable path will have a drop rate >0.5.
>
> High level of congestion: NON-ECT drop rate at high level (10%). ECT on
> capable transport will have some drop rate x, where 10 > x > 0, The CE
> marking rate will be approx 10 - x. ECT on incapable path will see a
> drop rate that is > 10.
>
> Based on the above, I think you can determine when there drops on ECT
> packets that are not related. This as you can take the CE marking rates
> into account. NON-ECT droped packets - CE marked packets should
> approximately equal the number of ECT marked drop packets. What I am
> unclear on in this case is how big fudge factor you do need. But that
> needs to be derived based on statics.
>
> Comments on the above reasoning.
>
>
>>
>> T6/ Pkt Reordering around session start-------------------
>> [Just for the record - I've already sent this point off list.]
>>
>> I will use the term "RTCP ECN Reports" for both the RTCP AVPF NACK
>> transport feedback format and the RTCP XR ECN summary report.
>> "
>> 1281 =A0 =A04.4.2. =A0Interpretation of ECN Summary information
>>
>> 1306 =A0 =A0 =A0 The counters will be initiated to zero to provide value=
 for the RTP
>> 1307 =A0 =A0 =A0 stream sender from the very first report. =A0After the =
first
>> report the
>> 1308 =A0 =A0 =A0 changes between the latest received and the previous on=
e is
>> 1309 =A0 =A0 =A0 determined by simply taking the values of the latest mi=
nus the
>> 1310 =A0 =A0 =A0 previous one, taking field wrapping into account. =A0Th=
is
>> definition is
>> 1311 =A0 =A0 =A0 also robust to packet losses, since if one report is mi=
ssing, the
>> 1312 =A0 =A0 =A0 reporting interval becomes longer, but is otherwise equ=
ally valid.
>> "
>> Both proposed RTCP ECN Reports insufficiently define the range of
>> packets they cover. They both define the range by the highest
>> sequence number and the number of packets covered by the report (the
>> sum of the various couters in the report, all counted since the start
>> of the receiver's session).
>>
>> This approach makes report accuracy vulnerable to reordering around
>> the start of the session. For instance, suppose teh receiver receives
>> the following packets, and imagine the receiver starts its session at
>> sequence #126. Then =A0sends its first report after seq #135:
>>
>> ...119,122,123,124,125,126,127,128,120,121,129,130,132,133,134,135,...
>>
>> Highest seq no =A0=3D #135
>> Number of losses =A0 =A0 =A0 =A0=3D 1 (#131)

This is not what is conveyed in this RTCP report - The 'Lost packets
counter' is a signed quantity and is calculated as follows from
RFC3550 (S6.4.1 and/or A.3):

Lost packets counter  =3D expected - num_received
In your example:  expected =3D 135 - 126 +1 =3D 10
So the 'Lost packets counter' (=3D 10 - 11) =3D -1

Thus one can always calculate the correct number of expected packets
at the sender. So there is not a mismatch of state between the sender
and receiver.

>> packets received =A0 =A0 =A0 =A0=3D 11 (which will be broken down into C=
E, ECT etc)
>> (Total pkts in report =A0 =3D 12)
>>
>> The sender will calculate that this report refers to the 12 packets
>> starting at 135-12=3D#123 and ending at #135.
>>
>> Then, the sender cannot accurately check whether the number of ECT &
>> CE (or Not-ECT) packets reported by the receiver matches the number
>> of ECT (or Not-ECT) it sent, because they are both talking about a
>> different set of packets. For instance, if the sender sets #123 and
>> #124 as ECT probes, these will be counted as ECT by the sender, but
>> the receiver puts them outside the range it is reporting on.
>>
>> This error will be repeated each time the receiver sends a report,
>> because it is always cumulative from the start of the session, and
>> the problem is an ill-defined session start.
>>
I'm not sure if this is the case as the cumulative counts allow for
relative checks by comparing the max_seq numbers (and losses) of two
(or more) reports, thus one can ignore the ill-defined start. The
absolute numbers may be wrong but they're less useful than keeping
tabs on the current reporting periods. In RFC3550 it talks of making
such a comparison (e.g. S6.4.4) which means one needs to wait till
there are at least two RTCP reports before one can make such a
comparison.

>> There are three possible solutions:
>> a) Add text to mandate that the receiver considers the session start
>> to be the lowest sequence number after it initialises its counters.
>> In the example, the receiver would count the session start as #120
>> not #126, even though it initialised its counters at #126. When it
>> receives packet #120 it immediately rolls back the start seqno to
>> #120 and considers 121-125 as 5 losses. Ultimately, because it later
>> receives #121 it only reports #122-125 as 4 losses - even if these
>> packets did actually arrive before the counter was initialised.
>>
>> b) The receiver does not include any packets with seqnos below the
>> seqno when it initialised its variables (unless there has been a wrap
>> of course). In the example, the receever would not include packets
>> #120 & #121 in its RTCP ECN Reports.
>>
>> c) The RTCP ECN Report formats include an explicit start seqno. I
>> don't think this is necessary, given either a) or b) are easy to do.
>>
>> Am I correct, or have I overlooked something?
>
> Yes, you are correct that it is important that a receiver do keep track
> of which packet is the first to count from and thus either doing A or B.
> I think we should add a clarification on this issue. As long as you do
> either of A or B the sender wouldn't be fooled about the set.
>
> I think from a probing point of view doing B would actually be better.
> If one take this as an example the packet sender may could come to the
> conclusion that the ECT probes are lost due to incapability rather than
> re-ordering. The potential down side is that a probe that has been
> received may fall outside of the window and need an additional round.
>
> There are an alternative to B that I think will work, but most likely
> are more complex to implement. That is that the sender will move its
> start number back as long as possible from the initially received number
> as long as there is no hole.
>
I have looked more closely at RFC3550 and it seems that I was wrong
when I said it covered this but our draft's approach is different as
it also includes the number of received packets. Looking more closely
at rfc3550 it seems to me that this may instead be an issue for it -
In Appendix A.3 the base_seq is mentioned as being the "first sequence
number received". Thus reordering at the start of the receiver's
session would lead to mismatch between the sender and receiver state,
though in the case of general RTCP this probably isn't such of a
problem - as one can always check relative counts.

So it seems to me that the existing approach would work ok.

Piers


> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

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Subject: Re: [AVT] FW: New Version Notification	fordraft-xia-avt-splicing-for-rtp-00
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On Oct 21, 2010, at 11:30 PM, Jinwei Xia wrote:

>=20
> Hi Dave,
>=20
> Please see inline.=20
>=20
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
>> Behalf Of David R Oran
>> Sent: Friday, October 22, 2010 4:27 AM
>> To: Colin Perkins
>> Cc: 'IETF AVT WG'; Roni Even
>> Subject: Re: [AVT] FW: New Version Notification=20
>> fordraft-xia-avt-splicing-for-rtp-00
>>=20
>>=20
>> On Oct 21, 2010, at 4:17 PM, Colin Perkins wrote:
>>=20
>>> Roni,
>>>=20
>>> I don't think it matters that the content arrives as an RTP=20
>> stream. It's an entirely separate stream, and doesn't affect=20
>> the translation process. They're logically separate.
>>>=20
>> I agree with Colin here.=20
>>=20
>> Either you treat the streams as equal inputs to a mixer, and=20
>> the case reduces to the problem of a video mixer doing input=20
>> switching, or you treat all the other inputs as external=20
>> stuff that gets inserted into the main stream using a translator.
>>=20
>> If you choose the mixer approach, you have to be a legal=20
>> mixer, which means you keep state, use your own SSRC, and=20
>> report CSRCs.
>=20
> A standard mixer may not work well, it mix all the streams rather than
> select one of them.
>=20
Mixers are perfectly allowed to do simple stream selection/switching. =
Nearly all video "mixers" in fact do that rather than actually =
compositing or other forms of "mixing". I think you're being misled by =
the name.

>>=20
>> If you choose the translator approach, you have to "consume"=20
>> the other stream and then emit it as if it were emitted by=20
>> the original stream's source.
>=20
> Translator acts as a client and terminate inserted stream from =
inserted
> stream source view compared to acting as translator from original =
stream
> source view.  But the RTCP RR issue mentioned by Roni is unsolved, I =
also
> mark this issue as TBD in draft. =20
>=20
Ok, but I'm still not seeing much of an issue.. Maybe I'm being na=EFve...=


>>=20
>> I don't see what the other approaches would be or how they=20
>> might work better.
>=20
> MCU is not well defined in IETF, I agree to limit the scope in mixer =
and
> translator.
>=20
Well, let's not get hung up too much on terminology; what's important is =
the function that have to be performed (or not performed), ensuring =
interoperability among the elements, and not making architectural =
changes to RTP unless no other approach satisfies the requirments.


>>=20
>> DaveO.
>>=20
>>> Colin
>>>=20
>>>=20
>>>=20
>>> On 21 Oct 2010, at 19:26, Roni Even wrote:
>>>> Hi Colin,
>>>> In this use case the inserted content is not an RTP stream=20
>> but comes from a local storage. For this use case the splicer=20
>> is a translator. It will still need to handle the RTCP RR=20
>> depending on who originated the data that was sent by the translator.
>>>> My question will be if this is the typical use case or is=20
>> there a use=20
>>>> case where the inserted data is not local and may arrive as RTP=20
>>>> stream. In which case there is more than one topology
>>>>=20
>>>> Roni
>>>>=20
>>>>> -----Original Message-----
>>>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>>>> Sent: Thursday, October 21, 2010 7:23 PM
>>>>> To: Roni Even
>>>>> Cc: 'Jinwei Xia'; 'IETF AVT WG'
>>>>> Subject: Re: [AVT] FW: New Version Notification for=20
>> draft-xia-avt-=20
>>>>> splicing-for-rtp-00
>>>>>=20
>>>>> Hi Roni,
>>>>>=20
>>>>> I think this problem becomes clearer if you consider a=20
>> splicer that=20
>>>>> is reading the content to be inserted from a local file=20
>> on disk. In=20
>>>>> that case, the splicer is unquestionably a standard RTP=20
>> translator,=20
>>>>> in the middle of a one-to-one or one-to-many RTP session.
>>>>>=20
>>>>> That there are also some cases where an entirely separate RTP=20
>>>>> session is used to deliver the content to be inserted doesn't=20
>>>>> (conceptually) affect the translation process at all.
>>>>>=20
>>>>> Colin
>>>>>=20
>>>>>=20
>>>>>=20
>>>>> On 21 Oct 2010, at 07:56, Roni Even wrote:
>>>>>> Hi Colin,
>>>>>> I am not so sure about this being a regular RTP translator.=20
>>>>>> According
>>>>> To RFC 5117 " A Translator always keeps the SSRC for a=20
>> stream across=20
>>>>> the translation". Also in the splicing case it is not=20
>> point to point=20
>>>>> or multipoint but more like multipoint to point or=20
>> multipoint. This=20
>>>>> looks more like a mixer that does not provide the CSRC=20
>> since one of=20
>>>>> the requirements is to make it look to the receiver that all the=20=

>>>>> media is coming from the same source. One issue that will=20
>> need to be=20
>>>>> addressed is how to report the RTCP RR from the receivers back to=20=

>>>>> the sender of the primary or insertion streams. This may=20
>> need some=20
>>>>> state keeping in the splicer.
>>>>>>=20
>>>>>> Roni Even
>>>>>>=20
>>>>>>> -----Original Message-----
>>>>>>> From: avt-bounces@ietf.org=20
>> [mailto:avt-bounces@ietf.org] On Behalf
>>>>> Of
>>>>>>> Colin Perkins
>>>>>>> Sent: Wednesday, October 20, 2010 3:52 PM
>>>>>>> To: Jinwei Xia
>>>>>>> Cc: 'IETF AVT WG'
>>>>>>> Subject: Re: [AVT] FW: New Version Notification for=20
>> draft-xia-avt-=20
>>>>>>> splicing-for-rtp-00
>>>>>>>=20
>>>>>>> Hi Jinwei,
>>>>>>>=20
>>>>>>> My impression is that this draft confuses the issue by=20
>> focussing=20
>>>>>>> too
>>>>> much on scenarios where the content to be inserted is=20
>> also delivered=20
>>>>> over RTP. Whether the content to be inserted is delivered=20
>> as a real-=20
>>>>> time RTP stream or fetched from local file storage does=20
>> not affect=20
>>>>> the insertion process, and so doesn't need to be mentioned in the=20=

>>>>> draft. I still believe that a standard RTP translator is=20
>> sufficient=20
>>>>> to solve this problem, although there may be useful things to say=20=

>>>>> about how to efficiently implement such a translator.
>>>>>>>=20
>>>>>>> Colin
>>>>>>>=20
>>>>>>>=20
>>>>>>> On 16 Oct 2010, at 03:06, Jinwei Xia wrote:
>>>>>>>> Hi,
>>>>>>>>=20
>>>>>>>> =46rom the discussion on IETF 78th, there were some=20
>>>>>>>> misunderstandings
>>>>>>> of the scope of splicing draft. So we discard previous problem=20=

>>>>>>> statement draft while initiate new solution draft. In this new
>>>>> draft,
>>>>>>> We emphysize content insertion issue is indeed a RTP-generic=20
>>>>>>> rather than MPEG2 TS-specific.
>>>>>>>>=20
>>>>>>>> Hope the new draft can address most issues AVT concern, any
>>>>> comments
>>>>>>> are very appreciated!
>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> BR!
>>>>>>>>=20
>>>>>>>> Jinwei
>>>>>>>>=20
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
>>>>>>>>> Sent: Saturday, October 16, 2010 9:58 AM
>>>>>>>>> To: xiajinwei@huawei.com
>>>>>>>>> Subject: New Version Notification for=20
>>>>>>>>> draft-xia-avt-splicing-for-rtp-00
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>> A new version of I-D, draft-xia-avt-splicing-for-rtp-00.txt
>>>>>>>>> has been successfully submitted by Jinwei Xia and=20
>> posted to the=20
>>>>>>>>> IETF repository.
>>>>>>>>>=20
>>>>>>>>> Filename:	 draft-xia-avt-splicing-for-rtp
>>>>>>>>> Revision:	 00
>>>>>>>>> Title:		 Content Splicing for RTP Sessions
>>>>>>>>> Creation_date:	 2010-10-16
>>>>>>>>> WG ID:		 Independent Submission
>>>>>>>>> Number_of_pages: 12
>>>>>>>>>=20
>>>>>>>>> Abstract:
>>>>>>>>> This memo outlines RTP splicing.  Splicing is a process that=20=

>>>>>>>>> allows a new multimedia stream to be inserted into current=20
>>>>>>>>> multimedia stream and to be conveyed to receiver for=20
>> a period of=20
>>>>>>>>> time.  This memo discusses the requirements of RTP=20
>> splicing.  In=20
>>>>>>>>> order to satisfy the requirements, this memo lists several=20
>>>>>>>>> existing intermediary network elements as alternatives and=20
>>>>>>>>> evaluates whether one of alternatives can be used to=20
>> perform RTP=20
>>>>>>>>> splicing.
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>> The IETF Secretariat.
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> _______________________________________________
>>>>>>>> Audio/Video Transport Working Group avt@ietf.org=20
>>>>>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>>>>=20
>>>>>>> --
>>>>>>> Colin Perkins
>>>>>>> http://csperkins.org/
>>>>=20
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avt
>>>=20
>>>=20
>>>=20
>>> --
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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From Even.roni@huawei.com  Fri Oct 22 14:12:17 2010
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Date: Fri, 22 Oct 2010 23:10:24 +0200
From: Roni Even <Even.roni@huawei.com>
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Cc: "'DRAGE, Keith \(Keith\)'" <keith.drage@alcatel-lucent.com>, avt@ietf.org
Subject: [AVT] draft-brandenburg-avt-rtcp-for-idms-02 - going forward
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Hi,
I read the draft and looked also at annex W of TS 183 063 v3.4.1 which
specifies this XR block and SDP. 
I appreciate you brining this document to the IETF AVT WG even though it
would have been enough to have it in TS 183 063 according to the
specification required policy in RFC 3611 section 6.2.

I think it will be good to present it in Beijing and we can ask if people in
AVT would like to have it as a work item or just help by reviewing leaving
it in TS 183 063.

Regards
Roni Even
AVT co-chair




From sunseawq@huawei.com  Sun Oct 24 18:54:17 2010
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Date: Mon, 25 Oct 2010 09:55:48 +0800
From: Qin Wu <sunseawq@huawei.com>
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Subject: Re: [AVT] I-D Action:draft-wu-avt-retransmission-supression-rtp-06.txt
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Hi, folks:
06 version of draft-wu-avt-retransmission-supression-rtp is posted which address the remaining comments:
* shorten the abstract
* simplify the introduction section
* remove the appdendix
* rephrase the section 6.1.1 to define only the behaviour of a single distribution source
* keeps retransmission as packet loss repair mechanism out.
* Add RAMS case to explain how feedback suppression works.
A URL for this Internet-Draft is:
 http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-06.txt
Thank for Ali, Colin and Tom's comments.

Regards!
-Qin
----- Original Message ----- 
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Subject: I-D Action:draft-wu-avt-retransmission-supression-rtp-06.txt


>A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> Title           : RTCP Report Extension for Feedback Suppression
> Author(s)       : Q. Wu, et al.
> Filename        : draft-wu-avt-retransmission-supression-rtp-06.txt
> Pages           : 17
> Date            : 2010-10-24
> 
> In a large RTP session using the RTCP feedback mechanism defined in
> RFC 4585, a media source or middlebox may experience transient
> overload if some event causes a large number of receivers to send
> feedback at once.  This feedback implosion can be mitigated if the
> device suffering from overload can send a feedback suppression
> message to the receivers to inhibit further feedback.  This memo
> defines RTCP extensions for feedback suppression, to suppress NACK
> and FIR feedback requests.  It also defines associated SDP
> signalling."
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-avt-retransmission-supression-rtp-06.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
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> 
> Below is the data which will enable a MIME compliant mail reader
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From csp@csperkins.org  Mon Oct 25 04:00:43 2010
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References: <201008052251.o75Mp7BL005112@bagheera.jungle.bt.co.uk>	<662B899F-09B5-48E5-B164-C61D278E4670@g11.org.uk>	<201008252336.o7PNafwl030089@bagheera.jungle.bt.co.uk>	<6D774CF6-FB7C-4710-A040-FCF40775FB8A@csperkins.org> <DBB1DC060375D147AC43F310AD987DCC180E4086A8@ESESSCMS0366.eemea.ericsson.se> <4CBFEC63.7090003@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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On 21 Oct 2010, at 08:31, Magnus Westerlund wrote:
> Ingemar Johansson S skrev 2010-10-21 09:23:
>> I am working on the draft, removing the nonce stuff. As I understand =
it, section 5.3 can be removed entirely, but I guess some text is needed =
to indicated that RFC3611 (4.1) can be used to indicate packets that are =
lost or ECN-CE marked.=20
>> ECN-CE is not mentioned in RFC3611, do you believe that it is valid =
to use RFC3611 for indication on ECN-CE. I guess it is if one stick to =
the principle that ECN-CE should be reacted on in the same way as a =
packet loss. What is your opinion.
>=20
> I at least are of the opinion that if a packet arrived with a CE mark,
> it is still delivered. It shall not be marked as lost in the RTCP XR
> packet loss vector. If we want/need to know exactly which packets =
where
> CE marked, then we should define a CE marking XR vector for that.
> Overloading of this would reduce the utility of the packet loss =
vector.


Agreed. The congestion response should be the same for a packet that's =
lost or CE marked, but the effect on the user experience will be quite =
different, so they should be reported differently.=20

--=20
Colin Perkins
http://csperkins.org/




From rbriscoe@jungle.bt.co.uk  Mon Oct 25 07:50:14 2010
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To: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVT] Removing nonce (WAS RE: draft-ietf-avt-ecn-for-rtp-02 T3/ Anti-Cheating)
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Folks,

May not get time to respond to all these responses on ECN-for-RTP. 
Too many of my own drafts in process for today's deadlines.


Bob

At 12:02 25/10/2010, Colin Perkins wrote:
>On 21 Oct 2010, at 08:31, Magnus Westerlund wrote:
> > Ingemar Johansson S skrev 2010-10-21 09:23:
> >> I am working on the draft, removing the nonce stuff. As I 
> understand it, section 5.3 can be removed entirely, but I guess 
> some text is needed to indicated that RFC3611 (4.1) can be used to 
> indicate packets that are lost or ECN-CE marked.
> >> ECN-CE is not mentioned in RFC3611, do you believe that it is 
> valid to use RFC3611 for indication on ECN-CE. I guess it is if one 
> stick to the principle that ECN-CE should be reacted on in the same 
> way as a packet loss. What is your opinion.
> >
> > I at least are of the opinion that if a packet arrived with a CE mark,
> > it is still delivered. It shall not be marked as lost in the RTCP XR
> > packet loss vector. If we want/need to know exactly which packets where
> > CE marked, then we should define a CE marking XR vector for that.
> > Overloading of this would reduce the utility of the packet loss vector.
>
>
>Agreed. The congestion response should be the same for a packet 
>that's lost or CE marked, but the effect on the user experience will 
>be quite different, so they should be reported differently.
>
>--
>Colin Perkins
>http://csperkins.org/

________________________________________________________________
Bob Briscoe,                                BT Innovate & Design 


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : RTP Control Protocol (RTCP) Port for Source-Specific Multicast (SSM) Sessions
	Author(s)       : A. Begen
	Filename        : draft-ietf-avt-rtcp-port-for-ssm-03.txt
	Pages           : 7
	Date            : 2010-10-25

The Session Description Protocol (SDP) has an attribute that allows
RTP applications to specify an address and a port associated with the
RTP Control Protocol (RTCP) traffic.  In RTP-based source-specific
multicast (SSM) sessions, the same attribute is used to designate the
address and the RTCP port of the Feedback Target in the SDP
description.  However, the RTCP port associated with the SSM session
itself cannot be specified by the same attribute to avoid ambiguity,
and thus, is required to be derived from the "m=" line of the media
description.  Deriving the RTCP port from the "m=" line imposes an
unnecessary restriction.  This document removes this restriction by
introducing a new SDP attribute.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.


	Title           : Explicit Congestion Notification (ECN) for RTP over UDP
	Author(s)       : M. Westerlund, et al.
	Filename        : draft-ietf-avt-ecn-for-rtp-03.txt
	Pages           : 44
	Date            : 2010-10-25

This document specifies how explicit congestion notification (ECN)
can be used with RTP/UDP flows that use RTCP as feedback mechanism.

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From yangpeilin@huawei.com  Mon Oct 25 20:14:10 2010
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Hi=2C

New Version for draft-xia-avt-mpeg2ts-preamble-04 has been submitted=2E =


This new version mainly adds the text about the timing of the MPEG2-TS Pr=
eamble at section 6=2E

Any comments are welcome=2E

Peilin




----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A IETF I-D Submission Tool =3Cidsubmission=40ietf=2Eo=
rg=3E
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A new version of I-D=2C draft-xia-avt-mpeg2ts-preamble-04=2Etxt has been =
successfully submitted by Peilin Yang and posted to the IETF repository=2E=


Filename=3A         draft-xia-avt-mpeg2ts-preamble
Revision=3A         04
Title=3A                 Preamble Acquisition of MPEG2-TS Multicast Sessi=
ons
Creation=5Fdate=3A         2010-10-25
WG ID=3A                 Independent Submission
Number=5Fof=5Fpages=3A 23

Abstract=3A
The ITU-T Rec=2E H=2E222=2E0 =7C ISO/IEC 13818-1 Transport Stream (MPEG2-=
TS)
addresses the combining of one or more elementary streams of video
and audio=2C as well as other data=2C into single or multiple streams
which are suitable for storage or transmission=2E  The necessary and
sufficient information contained in the Program Specific
Information(PSI) tables to demultiplex and present programs must be
acquired before a RTP receiver can process any data received in
MPEG2-TS=2E  In this document=2C a Retransmission Server is specified to
deliver MPEG2-TS preamble prior to unicast burst RTP packets=2E  The
Retransmission Server caches raw RTP packets with MPEG2-TS preamble
information=2C and sends them to the RTP receiver which initiates rapid
acquisition of MPEG2-TS multicast sessions=2E
                                                                         =
         =



The IETF Secretariat=2E


From Even.roni@huawei.com  Mon Oct 25 22:11:52 2010
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Date: Tue, 26 Oct 2010 07:12:58 +0200
From: Roni Even <Even.roni@huawei.com>
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Hi,

I assume that the problem is because in TRAU you can send either CMR or =
CMI while RFC4867 allows both in the same packet. Is this correct?

Roni Even

=20

From: Xie Lei [mailto:xielei57471@huawei.com]=20
Sent: Monday, October 25, 2010 9:14 AM
To: avt@ietf.org
Cc: keith.drage@alcatel-lucent.com; 'Roni Even'
Subject: [AVT]The interworking issue for RTP Payload Format for AMR

=20

Hello All

I work on one interworking task between AMR RTP Payload and TRAU frame =
in TDM network, and find it is very difficult to map the =
Codec_Mode_Request(CMR) value in RTP to Request or Indication Flag (RIF) =
in TRAU frame which is defined in 3GPP 48.060/48.061.=20

As defined in RFC4867, RTP frame can take CMR and CMI(FT field) both in =
payload and the codec mode change period could be 20ms. However,TRAU =
speech frame can only take one of CMR or CMI, and the RIF should follow =
rigorous mode change peroid, that means, TRAU speech frame could only =
take CMR every 40ms and between each CMR frame, the TRAU speech frame =
must be CMI frame.=20

The issue happens when two consequent RTP payload takeing valide CMR =
value in each payload, at this scenario, we do not know how to set the =
RIF value. For instance, the first RTP payload takes CMR=3D4, and we map =
this CMR value to RIF flag to indicate the TRAU speech frame is for CMR, =
and the second RTP payload comes with CMR=3D1, then we really do not =
know how to do now. according to the RTP payload CMR valude, we should =
indicate RIF as CMR. However, according to the TRAU frame definition, =
the RIF must be CMI which following with last CMR frame.

RFC4867 has some decription on reqirements for encoder in GSM. However, =
it is not enough, i think there needs some description on how to map CMR =
and CMI value in RTP frame to RIF defined TRAU when interworking between =
IP and TDM network. currentlly , we can not find such mapping table in =
any specification. We hope avt groud could work on this and resolve this =
issue.

=20

Best Regards

Rock

=20

=20

=20

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<div class=3DWordSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I assume that the problem is because in TRAU you can send =
either
CMR or CMI while RFC4867 allows both in the same packet. Is this =
correct?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Xie Lei
[mailto:xielei57471@huawei.com] <br>
<b>Sent:</b> Monday, October 25, 2010 9:14 AM<br>
<b>To:</b> avt@ietf.org<br>
<b>Cc:</b> keith.drage@alcatel-lucent.com; 'Roni Even'<br>
<b>Subject:</b> [AVT]The interworking issue for RTP Payload Format for =
AMR<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Hello
All</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I&nbsp;work
on one interworking task between AMR RTP Payload and </span><span =
lang=3DFR
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>TRAU frame =
in TDM
network, and find it is very difficult to map the </span><span lang=3DFR
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:black'>C=
odec_Mode_Request(CMR)
value in RTP to </span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>Request</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:red'> </span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>or Indication Flag</span><span =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif";color:red'> </span><span style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:black'>(RIF) in TRAU frame which =
is
defined in</span><span style=3D'font-size:10.5pt;color:black'> =
</span><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:black'>3=
GPP
48.060/48.061. </span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>As defined in RFC4867, RTP frame can take CMR and CMI(FT =
field)
both in payload and the codec mode change period&nbsp;could =
be&nbsp;20ms.
However,</span><span style=3D'color:black'>TRAU speech frame can only =
take one of
CMR or CMI, and the RIF </span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>should follow&nbsp;rigorous mode change peroid, that means, =
</span><span
style=3D'color:black'>TRAU speech frame</span><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:black'> could only take CMR every =
40ms
and between each CMR frame, the </span><span style=3D'color:black'>TRAU =
speech
frame&nbsp;</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>must be CMI frame. </span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>The&nbsp;issue happens when two consequent RTP&nbsp;payload
takeing valide CMR value in each payload, at this scenario, we do not =
know how
to set the RIF value. For instance, the first RTP payload takes CMR=3D4, =
and we
map this CMR value to RIF flag to indicate the </span><span =
style=3D'color:black'>TRAU
speech frame</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'> </span><span style=3D'color:black'>is for CMR, and the =
second RTP
payload comes with CMR=3D1, then we really do not know how to do now. =
according
to the RTP payload CMR valude, we should indicate RIF as CMR. However,
according to the TRAU frame definition, the RIF&nbsp;must be CMI
which&nbsp;following with last CMR frame.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>RFC4867 has some decription on reqirements for encoder in =
GSM.
However, it is not enough,&nbsp;i think there needs some description
on&nbsp;how to map CMR and CMI value in RTP frame&nbsp;to
RIF&nbsp;defined&nbsp;TRAU when interworking between IP and TDM network.
currentlly , we can not find such mapping table in any specification. We =
hope
avt groud could work on this and&nbsp;resolve this =
issue.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>Best Regards</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'color:black'>Rock</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial =
Unicode MS","sans-serif";
color:gray'>=E5=8D=8E=E4=B8=BA=E6=8A=80=E6=9C=AF=E6=9C=89=E9=99=90=E5=85=AC=
</span><span style=3D'font-size:10.0pt;font-family:"MS Mincho";
color:gray'>=E5=8F=B8</span><span =
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<p class=3DMsoNormal><br>
<span style=3D'font-size:7.5pt;font-family:"MS =
Mincho";color:gray'>=E5=9C=B0=E5=9D=80=EF=BC=9A=E6=B7=B1=E5=9C=B3=E5=B8=82=
</span><span
style=3D'font-size:7.5pt;font-family:"Arial Unicode =
MS","sans-serif";color:gray'>=E9=BE=99=E5=B2=97=E5=8C=BA=E5=9D=82=E7=94=B0=
=E5=8D=8E=E4=B8=BA=E5=9F=BA=E5=9C=B0</span><span
style=3D'font-size:7.5pt;color:gray'> </span><span =
style=3D'font-size:7.5pt;
font-family:"Arial Unicode =
MS","sans-serif";color:gray'>=E9=82=AE=E7=BC=96=EF=BC=9A</span><span
style=3D'font-size:7.5pt;color:gray'>518129<br>
</span><span =
style=3D'font-size:7.5pt;font-family:=E5=8D=8E=E6=96=87=E7=BB=86=E9=BB=91=
;color:gray'><a
href=3D"http://www.huawei.com">http://www.huawei.com</a><br>
-------------------------------------------------------------------------=
------------------------------------------------------------<br>
</span><span style=3D'font-size:7.5pt;font-family:"MS =
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Date: Tue, 26 Oct 2010 14:04:28 +0800
From: Xie Lei <xielei57471@huawei.com>
To: Roni Even <Even.roni@huawei.com>, avt@ietf.org
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Subject: Re: [AVT] The interworking issue for RTP Payload Format for AMR
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Hi
Yes, the TRAU can only take either CMR or CMI, and in GSM network, TRAU frame sequence must strictly be one CMR frame following with one CMI frame , however, RFC4867 allows take both CMR and CMI one packet and there is no indication to indentify the packet is CMR frame or not, that makes the problem.

BR
Rock
  ----- Original Message ----- 
  From: Roni Even 
  To: avt@ietf.org 
  Cc: 'Xie Lei' 
  Sent: Tuesday, October 26, 2010 1:12 PM
  Subject: FW: [AVT]The interworking issue for RTP Payload Format for AMR


  Hi,

  I assume that the problem is because in TRAU you can send either CMR or CMI while RFC4867 allows both in the same packet. Is this correct?

  Roni Even

   

  From: Xie Lei [mailto:xielei57471@huawei.com] 
  Sent: Monday, October 25, 2010 9:14 AM
  To: avt@ietf.org
  Cc: keith.drage@alcatel-lucent.com; 'Roni Even'
  Subject: [AVT]The interworking issue for RTP Payload Format for AMR

   

  Hello All

  I work on one interworking task between AMR RTP Payload and TRAU frame in TDM network, and find it is very difficult to map the Codec_Mode_Request(CMR) value in RTP to Request or Indication Flag (RIF) in TRAU frame which is defined in 3GPP 48.060/48.061. 

  As defined in RFC4867, RTP frame can take CMR and CMI(FT field) both in payload and the codec mode change period could be 20ms. However,TRAU speech frame can only take one of CMR or CMI, and the RIF should follow rigorous mode change peroid, that means, TRAU speech frame could only take CMR every 40ms and between each CMR frame, the TRAU speech frame must be CMI frame. 

  The issue happens when two consequent RTP payload takeing valide CMR value in each payload, at this scenario, we do not know how to set the RIF value. For instance, the first RTP payload takes CMR=4, and we map this CMR value to RIF flag to indicate the TRAU speech frame is for CMR, and the second RTP payload comes with CMR=1, then we really do not know how to do now. according to the RTP payload CMR valude, we should indicate RIF as CMR. However, according to the TRAU frame definition, the RIF must be CMI which following with last CMR frame.

  RFC4867 has some decription on reqirements for encoder in GSM. However, it is not enough, i think there needs some description on how to map CMR and CMI value in RTP frame to RIF defined TRAU when interworking between IP and TDM network. currentlly , we can not find such mapping table in any specification. We hope avt groud could work on this and resolve this issue.

   

  Best Regards

  Rock

   



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From: Varun Singh <varun.singh@gmail.com>
Date: Tue, 26 Oct 2010 09:53:37 +0300
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Cc: Roni Even <ron.even.tlv@gmail.com>, =?UTF-8?B?SsO2cmcgT3R0?= <jo@comnet.tkk.fi>, Keith Drage <drage@alcatel-lucent.com>, =?UTF-8?B?VGVlbXUgS8Okcmtrw6RpbmVu?= <teemuk@cc.hut.fi>, Varun Singh <varun@comnet.tkk.fi>, Lars Eggert <lars.eggert@nokia.com>, Saba Ahsan <sahsan@cc.hut.fi>
Subject: [AVT] Fwd: New Version Notification for draft-singh-avt-mprtp-01
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Hi,

This is an update to the Multipath RTP draft presented at Maastricht.
We have made additions in Section 8 and 9. (diff:
http://tools.ietf.org/rfcdiff?url2=draft-singh-avt-mprtp-01.txt).
Comments are welcome.

A brief summary, Multipath RTP picks up ideas from MPTCP and SCTP, it
aims to exploit the multiple interfaces for increasing throughput
and/or robustness.

Regards,
Varun

Begin forwarded message:

From: IETF I-D Submission Tool <idsubmission@ietf.org>
Date: October 25, 2010 10:34:20 PM GMT+03:00
To: varun@comnet.tkk.fi
Cc: sahsan@cc.hut.fi,jo@comnet.tkk.fi,teemuk@comnet.tkk.fi,lars.eggert@nokia.com
Subject: New Version Notification for draft-singh-avt-mprtp-01


A new version of I-D, draft-singh-avt-mprtp-01.txt has been
successfully submitted by Varun Singh and posted to the IETF
repository.

Filename:	 draft-singh-avt-mprtp
Revision:	 01
Title:		 Multipath RTP (MPRTP)
Creation_date:	 2010-10-25
WG ID:		 Independent Submission
Number_of_pages: 23

Abstract:
The Real-time Transport Protocol (RTP) is used to deliver real-time
content and, along with the RTP Control Protocol (RTCP), forms the
control channel between the sender and receiver.  However, RTP and
RTCP assume a single delivery path between the sender and receiver
and make decisions based on the measured characteristics of this
single path.  Increasingly, endpoints are becoming multi-homed, which
means that they are connected via multiple Internet paths.  Network
utilization can be improved when endpoints use multiple parallel
paths for communication.  The resulting increase in reliability and
throughput can also enhance the user experience.  This document
extends the Real-time Transport Protocol (RTP) so that a single
session can take advantage of the availability of multiple paths
between two endpoints.



The IETF Secretariat.


Varun Singh
http://www.netlab.tkk.fi/~varun

From 2mkristensen@gmail.com  Tue Oct 26 02:04:17 2010
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From: Tom Kristensen <2mkristensen@gmail.com>
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Subject: Re: [AVT] I-D Action:draft-kristensen-avt-rtp-h241param-01.txt
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On 25 October 2010 18:00,  <Internet-Drafts@ietf.org> wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>
> =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : Additional H.241 Parameter in =
the RTP Payload Format for H.264 Video
> =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : T. Kristensen, M. Walters
> =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-kristensen-avt-rtp-h241par=
am-01.txt
> =A0 =A0 =A0 =A0Pages =A0 =A0 =A0 =A0 =A0 : 12
> =A0 =A0 =A0 =A0Date =A0 =A0 =A0 =A0 =A0 =A0: 2010-10-25
>
> As systems increasingly are able to run video at 60 frames per
> second, there is a need to signal desired frame rate to optimise
> resolution choices and to avoid unnecessary resource or energy usage.
> This document defines a new optional parameter addressing recent
> extensions currently supported in H.323 systems: The signalling of
> the maximum frames per second that can be efficiently received or
> that can be sent.
>
> A URL for this Internet-Draft is:
http://tools.ietf.org/html/draft-kristensen-avt-rtp-h241param-01

The -01 version was submitted yesterday, before the IETF-79 great wall
surrounded the draft submission tool.

This version addresses the feedback and handles the issues from the
AVT mailing list discussion on this draft. Most important:
- Emphasize the resolution choice optimization and energy
consumption/battery duration as motivation for max-fps
- Underline ease of gatewaying between SIP and H.323 as a reason
- Added graphs as an aid visualizing the effect of max-fps, together
with macroblock and frame size signalling
- Changed the normative text in the media type registration along the
lines of for instance sar-supported in RFC3984bis
- Tightened text describing "a=3Dmaxprate" and H.26x frame rate
signalling parameters, to get the real message clearer through
- A better text regarding co-existence with "a=3Dframerate" added

(The -00/-01 diff:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-kristensen-avt-rtp-h241param-01.=
txt
)

-- Tom

From tom.kristensen@tandberg.com  Tue Oct 26 02:15:11 2010
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Cc: Malcolm Walters <malcowal@cisco.com>, IETF AVT WG <avt@ietf.org>, Tom Kristensen <tomkrist@cisco.com>
Subject: Re: [AVT] I-D Action:draft-kristensen-avt-rtp-h241param-01.txt
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On 10/26/2010 11:06 AM, Tom Kristensen wrote:
> On 25 October 2010 18:00,<Internet-Drafts@ietf.org>  wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>
>>         Title           : Additional H.241 Parameter in the RTP Payload Format for H.264 Video
>>         Author(s)       : T. Kristensen, M. Walters
>>         Filename        : draft-kristensen-avt-rtp-h241param-01.txt
[...]

I forgot to flag this "errata" in the announce message. Noticed just as 
-01 was submitted that we accidentally changed the normative text for 
the mediatype parameter registration.

Cf. 
http://tools.ietf.org/html/draft-kristensen-avt-rtp-h241param-01#section-3.1

As max-fps is an OPTIONAL parameter the right thing to say is
"This parameter MAY be used to indicate the maximum [...]" not
"This parameter SHOULD be used to indicate the maximum [...]"
as currently said in -01. Will be fixed in next version.

-- Tom

From ray.vanbrandenburg@tno.nl  Tue Oct 26 02:58:32 2010
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
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Hi all,

Roni, thanks for the support. I will prepare a presentation for Beijing.


For those that would like some more background on IDMS; on request I can
send you some papers on the topic.

Regards,

Ray van Brandenburg


-----Original Message-----
From: Roni Even [mailto:Even.roni@huawei.com]=20
Sent: vrijdag 22 oktober 2010 23:10
To: Brandenburg, R. (Ray) van
Cc: avt@ietf.org; 'DRAGE, Keith (Keith)'
Subject: draft-brandenburg-avt-rtcp-for-idms-02 - going forward

Hi,
I read the draft and looked also at annex W of TS 183 063 v3.4.1 which
specifies this XR block and SDP.=20
I appreciate you brining this document to the IETF AVT WG even though it
would have been enough to have it in TS 183 063 according to the
specification required policy in RFC 3611 section 6.2.

I think it will be good to present it in Beijing and we can ask if
people in AVT would like to have it as a work item or just help by
reviewing leaving it in TS 183 063.

Regards
Roni Even
AVT co-chair



This e-mail and its contents are subject to the DISCLAIMER at http://www.tn=
o.nl/disclaimer/email.html


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Cc: 'Keith Drage' <keith.drage@alcatel-lucent.com>
Subject: [AVT] Initial AVT Agenda for IETF 79
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Hi,

This is the initial agenda for AVT in the Beijing meeting.

Please review 

Regards

Keith Drage and Roni Even

 

 

Audio/Video Transport (AVT) Working Group

=========================================

 

CHAIRS:  Keith Drage        <keith.drage@alcatel-lucent.com>

         Roni Even         <even.roni@huawei.com>

 

AGENDA

 

Monday, 8 November 2010 at 13:00-15:00 (Valley Ballroom B)

----------------------------------------------------------

13:00   Introduction and Status Update                      (Chairs, 15)

 

13:15   Port Mapping Between Unicast and Multicast RTP Sessions(Begen, 20)

        draft-begen-avt-token-for-portmapping-01

 

13:35   Explicit Congestion Notification for RTP over UDP  (Perkins, 20)

        draft-ietf-avt-ecn-for-rtp-03

 

13:55   RTCP Report Extension for Feedback Suppression      (Qin Wu, 15)

        draft-wu-avt-retransmission-supression-rtp-05

 

14:10   RTCP XR for inter-destination media synchronization(Ray van
Brandenburg , 10)

        draft-brandenburg-avt-rtcp-for-idms-02

 

14:20   Additional Parameters in for H.264 Video    (Tom Kristensen, 10)

        draft-kristensen-avt-rtp-h241param-01

 

14:30   End

 

 

Thursday, 11 November 2010 at 13:00-15:00 (Garden Ballroom 3)

-------------------------------------------------------------

13:00   Introduction and Status Update                       (Chairs, 5)

 

13:05   Audio Levels status                                 (Chairs, 10)

 

13:15   Content Splicing for RTP Sessions               (Jinwei Xia, 15)

        draft-xia-avt-splicing-for-rtp-00

 

13:30   Monitoring Architectures for RTP                      (Even, 20)

        draft-hunt-avt-monarch-01

 

13:50   RTCP XR Report Blocks for Realtime Video Quality Monitoring(Qin Wu,
10)

        draft-wu-avt-rtcp-xr-quality-monitoring-04

 

14:00   Switching from unicast to multicast for multicast short
time-shift(Peilin Yang, 15)

        draft-yang-avt-switch-multicast-short-timeshift-00

 

14:15   Using approximate authentication SRTP          (Gabor Feher, 10)

        draft-feher-avt-approx-auth-srtp-00

 

14:25   End

 


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<p class=3DMsoNormal>Hi,<o:p></o:p></p>

<p class=3DMsoNormal>This is the initial agenda for AVT in the Beijing =
meeting.<o:p></o:p></p>

<p class=3DMsoNormal>Please review <o:p></o:p></p>

<p class=3DMsoNormal>Regards<o:p></o:p></p>

<p class=3DMsoNormal>Keith Drage and Roni Even<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Audio/Video Transport (AVT) Working =
Group<o:p></o:p></p>

<p =
class=3DMsoNormal>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></=
o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>CHAIRS:&nbsp; Keith =
Drage&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;keith.drage@alcatel-lucent.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Roni Even&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;even.roni@huawei.com&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>AGENDA<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Monday, 8 November 2010 at 13:00-15:00 (Valley =
Ballroom B)<o:p></o:p></p>

<p =
class=3DMsoNormal>-------------------------------------------------------=
---<o:p></o:p></p>

<p class=3DMsoNormal>13:00&nbsp;&nbsp; Introduction and Status =
Update&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Chairs, 15)<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:15&nbsp;&nbsp; Port Mapping Between Unicast and =
Multicast RTP
Sessions(Begen, 20)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-begen-avt-token-for-portmapping-01<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:35&nbsp;&nbsp; Explicit Congestion Notification =
for RTP over UDP&nbsp;
(Perkins, 20)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-ietf-avt-ecn-for-rtp-03<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:55&nbsp;&nbsp; RTCP Report Extension for =
Feedback Suppression&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Qin Wu, 15)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-wu-avt-retransmission-supression-rtp-05<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:10&nbsp;&nbsp; RTCP XR for inter-destination =
media
synchronization(Ray van Brandenburg , 10)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-brandenburg-avt-rtcp-for-idms-02<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:20&nbsp;&nbsp; Additional Parameters in for =
H.264 Video&nbsp;&nbsp;&nbsp; (Tom
Kristensen, 10)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-kristensen-avt-rtp-h241param-01<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:30&nbsp;&nbsp; End<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thursday, 11 November 2010 at 13:00-15:00 (Garden =
Ballroom
3)<o:p></o:p></p>

<p =
class=3DMsoNormal>-------------------------------------------------------=
------<o:p></o:p></p>

<p class=3DMsoNormal>13:00&nbsp;&nbsp; Introduction and Status =
Update&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Chairs, 5)<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:05&nbsp;&nbsp; Audio Levels =
status&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Chairs, 10)<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:15&nbsp;&nbsp; Content Splicing for RTP =
Sessions&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
(Jinwei Xia, 15)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-xia-avt-splicing-for-rtp-00<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:30&nbsp;&nbsp; Monitoring Architectures for
RTP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (Even, =
20)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-hunt-avt-monarch-01<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>13:50&nbsp;&nbsp; RTCP XR Report Blocks for =
Realtime Video Quality
Monitoring(Qin Wu, 10)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-wu-avt-rtcp-xr-quality-monitoring-04<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:00&nbsp;&nbsp; Switching from unicast to =
multicast for multicast
short time-shift(Peilin Yang, 15)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-yang-avt-switch-multicast-short-timeshift-00<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:15&nbsp;&nbsp; Using approximate authentication =
SRTP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(Gabor Feher, 10)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-feher-avt-approx-auth-srtp-00<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>14:25&nbsp;&nbsp; End<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

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I have just requested publication of draft-ietf-avt-rtcp-port-for-ssm-03 as=
 proposed standard.

The writeup follows at the end of this mail.

-03 was produced as a result of this review. It contains two small changes.=
 One reference changes from normative to informative. An additional sentenc=
e is introduced "Except where stated otherwise in this document, the rules =
of [RFC3550] apply." as RFC 3550 is currently a normative reference, and th=
is text is only implied in the introduction.


regards

Keith

-----------------------------------------------------------
Shepherd writeup for draft-ietf-avt-rtcp-port-for-ssm-03 "RTP Control=20
Protocol (RTCP) Port for Multicast Sessions" as proposed standard.

  (1.a) Who is the Document Shepherd for this document? Has the
        Document Shepherd personally reviewed this version of the=20
        document and, in particular, does he or she believe this=20
        version is ready for forwarding to the IESG for publication?=20

The document shepherd for this document is Keith Drage.

The document shepherd has reviewed the document and believes it is ready fo=
r=20
forwarding to the IESG for publication.

Document history:

-	draft-begen-avt-rtcp-port-for-ssm-00 was submitted 22nd March=20
2010 and expires 23rd September 2010;
-	draft-begen-avt-rtcp-port-for-ssm-01 was submitted 3rd April 2010=20
and expires 5th October 2010;
-	draft-ietf-avt-rtcp-port-for-ssm-00 was submitted 16th June 2010=20
and expires 18th December 2010;
-	draft-ietf-avt-rtcp-port-for-ssm-01 was submitted 3rd August 2010=20
and expires 4th February 2010;
-	draft-ietf-avt-rtcp-port-for-ssm-02 was submitted 25th August=20
2010 and expires 26th February 2011;
-	draft-ietf-avt-rtcp-port-for-ssm-03 was submitted 25th October=20
2010 and expires 28th April 2011;

Call for adoption of baseline as WG item was made 4th June 2010 and=20
acceptance declared 16th June 2010.

Working group last call was initiated on -01 version on 4th August 2010=20
for completion 25th August 2010 as proposed standard. The MMUSIC was=20
also asked to review this version at the same time as working group=20
last call as this document is substantially an SDP definition. Reviews=20
were received from Keith Drage.

Prior to working group last call, the document received comments from=20
Peter Musgrave.

  (1.b) Has the document had adequate review both from key WG members=20
        and from key non-WG members? Does the Document Shepherd have=20
        any concerns about the depth or breadth of the reviews that=20
        have been performed? =20

The document has been adequately reviewed (see 1a above). This is a very si=
mple=20
document and substantial comments would not be expected.

  (1.c) Does the Document Shepherd have concerns that the document=20
        needs more review from a particular or broader perspective,=20
        e.g., security, operational complexity, someone familiar with=20
        AAA, internationalization or XML?=20

There are no such concerns from the document shepherd.

  (1.d) Does the Document Shepherd have any specific concerns or=20
        issues with this document that the Responsible Area Director
        and/or the IESG should be aware of? For example, perhaps he=20
        or she is uncomfortable with certain parts of the document, or=20
        has concerns whether there really is a need for it. In any=20
        event, if the WG has discussed those issues and has indicated=20
        that it still wishes to advance the document, detail those=20
        concerns here. Has an IPR disclosure related to this document=20
        been filed? If so, please include a reference to the=20
        disclosure and summarize the WG discussion and conclusion on=20
        this issue.=20

There are no concerns from the document shepherd from this perspective with=
 the=20
document.

No IPR disclosures have been made against this document (and indeed none=20
previously in the area of SSM).

  (1.e) How solid is the WG consensus behind this document? Does it=20
        represent the strong concurrence of a few individuals, with=20
        others being silent, or does the WG as a whole understand and=20
        agree with it?  =20

>From a reviewing perspective, the interest in this specific document has be=
en=20
relatively small, but it forms an essential part of draft-ietf-avt-rapid-
acquisition-for-rtp which has been well and substantially reviewed. It=20
is assumed that the interested parties in that document have also reviewed =
this=20
document.

  (1.f) Has anyone threatened an appeal or otherwise indicated extreme=20
        discontent? If so, please summarise the areas of conflict in=20
        separate email messages to the Responsible Area Director. (It=20
        should be in a separate email because this questionnaire is=20
        entered into the ID Tracker.)=20

No appeals or areas of conflict or discontent have been identified.

  (1.g) Has the Document Shepherd personally verified that the=20
        document satisfies all ID nits? (See the Internet-Drafts Checklist=
=20
        and http://tools.ietf.org/tools/idnits/). Boilerplate checks are=20
        not enough; this check needs to be thorough. Has the document=20
        met all formal review criteria it needs to, such as the MIB=20
        Doctor, media type and URI type reviews?=20

Version 2.12.05 of ID nits identifies no issues.

No other formal reviews outside of the AVT working group are perceived as=20
necessary for this document.

  (1.h) Has the document split its references into normative and=20
        informative? Are there normative references to documents that=20
        are not ready for advancement or are otherwise in an unclear=20
        state? If such normative references exist, what is the=20
        strategy for their completion? Are there normative references=20
        that are downward references, as described in [RFC3967]? If=20
        so, list these downward references to support the Area=20
        Director in the Last Call procedure for them [RFC3967].=20

The document does split normative and informative references. All the norma=
tive=20
references have been reviewed and are correctly allocated as normative=20
references. None of these normative references constitute a down reference.

  (1.i) Has the Document Shepherd verified that the document IANA=20
        consideration section exists and is consistent with the body=20
        of the document? If the document specifies protocol=20
        extensions, are reservations requested in appropriate IANA=20
        registries? Are the IANA registries clearly identified? If=20
        the document creates a new registry, does it define the=20
        proposed initial contents of the registry and an allocation=20
        procedure for future registrations? Does it suggest a=20
        reasonable name for the new registry? See [RFC5226]. If the=20
        document describes an Expert Review process has Shepherd=20
        conferred with the Responsible Area Director so that the IESG=20
        can appoint the needed Expert during the IESG Evaluation?=20

An IANA considerations section is included and the sole IANA registration, =
the=20
registration of a new SDP attribute, has been verified against the registra=
tion=20
requirements of RFC 4566.

  (1.j) Has the Document Shepherd verified that sections of the=20
        document that are written in a formal language, such as XML=20
        code, BNF rules, MIB definitions, etc., validate correctly in=20
        an automated checker?=20

The only formal language is contained in one line of ABNF. This has been=20
verified manually by the document shepherd.

  (1.k) The IESG approval announcement includes a Document=20
        Announcement Write-Up. Please provide such a Document=20
        Announcement Write-Up? Recent examples can be found in the
        "Action" announcements for approved documents. The approval=20
        announcement contains the following sections:=20
     Technical Summary=20
        Relevant content can frequently be found in the abstract=20
        and/or introduction of the document. If not, this may be=20
        an indication that there are deficiencies in the abstract=20
        or introduction.=20
     Working Group Summary=20
        Was there anything in WG process that is worth noting? For=20
        example, was there controversy about particular points or=20
        were there decisions where the consensus was particularly=20
        rough?=20
     Document Quality=20
        Are there existing implementations of the protocol? Have a=20
        significant number of vendors indicated their plan to=20
        implement the specification? Are there any reviewers that=20
        merit special mention as having done a thorough review,=20
        e.g., one that resulted in important changes or a=20
        conclusion that the document had no substantive issues? If=20
        there was a MIB Doctor, Media Type or other expert review,=20
        what was its course (briefly)? In the case of a Media Type=20
        review, on what date was the request posted?

The Session Description Protocol (SDP) has an attribute that allows RTP=20
applications to specify an address and a port associated with the RTP=20
Control Protocol (RTCP) traffic.  In RTP-based source-specific=20
multicast (SSM) sessions, the same attribute is used to designate the=20
address and the RTCP port of the Feedback Target in the SDP description. =20
However, the RTCP port associated with the SSM session itself cannot be=20
specified by the same attribute to avoid ambiguity, and thus, is=20
required to be derived from the "m=3D" line of the media description. =20
Deriving the RTCP port from the "m=3D" line imposes an unnecessary=20
restriction.  This document removes this restriction by introducing a=20
new SDP attribute.

The document achieved consensus in the AVT working group.

From wwwrun@core3.amsl.com  Wed Oct 27 09:37:30 2010
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From: IETF Secretariat <ietf-ipr@ietf.org>
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Subject: [AVT] IPR Disclosure: Nokia Corporation's Statement about IPR related to draft-ietf-avt-rtp-rfc3984bis-12
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Dear Tom Kristensen, Ye-Kui Wang, Roni Even, Randell Jesup:

An IPR disclosure that pertains to your Internet-Draft entitled "RTP Payload
Format for H.264 Video" (draft-ietf-avt-rtp-rfc3984bis) was submitted to the
IETF Secretariat on 2010-10-27 and has been posted on the "IETF Page of
Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/1431/). The title of the IPR disclosure is
"Nokia Corporation's Statement about IPR related to
draft-ietf-avt-rtp-rfc3984bis-12."

The IETF Secretariat



From stewe@stewe.org  Fri Oct 29 09:56:40 2010
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Date: Fri, 29 Oct 2010 09:57:34 -0700
From: Stephan Wenger <stewe@stewe.org>
To: <rai@ietf.org>, <fecframe@ietf.org>, <rmt@ietf.org>, <avt@ietf.org>, <mmusic@ietf.org>
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Thread-Topic: MPEG liaison statements received
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Subject: [AVT] MPEG liaison statements received
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

--B_3371191060_918748
Content-type: text/plain;
	charset="US-ASCII"
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Hi all,

The IETF has received from MPEG two liaison statements for our information.
No actions from the IETF side are requested or required.  Both statements
will appear shortly on the IETF's liaison website
https://datatracker.ietf.org/liaison/

n11632 informs us that their http streaming project has reached the
Committee Draft" (CD) approval level.

n11633 informs us that the joint IPTV project with the ITU Q13/16 has also
reached CD level.  Some of you may remember those assorted IPTV workshops of
the ITU and MPEG in 2008 and 2009; this is the outcome of the project
started back then.  In contrast to n11632, n11633 contains the CD text of
four future standards in winword format.

For those unaware of ISO's approval process, a CD is the first step in a
five (?) step approval process that ultimately ends in an International
Standard (IS).  At CD level, the key architectural design decisions are
frozen, but other technical input is still possible.  In IETF terms, it's
probably a maturity level that a WG I-D has after a couple of iterations.

Regards,
Stephan




--B_3371191060_918748
Content-type: text/html;
	charset="US-ASCII"
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div><br></div>=
<div>The IETF has received from MPEG two liaison statements for our informat=
ion. &nbsp;No actions from the IETF side are requested or required. &nbsp;Bo=
th statements will appear shortly on the IETF's liaison website&nbsp;<a href=
=3D"https://datatracker.ietf.org/liaison">https://datatracker.ietf.org/liaison=
</a>/</div><div><br></div><div>n11632 informs us that their http streaming p=
roject has reached the Committee Draft" (CD) approval level. &nbsp;</div><di=
v><br></div><div>n11633 informs us that the joint IPTV project with the ITU =
Q13/16 has also reached CD level. &nbsp;Some of you may remember those assor=
ted IPTV workshops of the ITU and MPEG in 2008 and 2009; this is the outcome=
 of the project started back then. &nbsp;In contrast to n11632, n11633 conta=
ins the CD text of four future standards in winword format.</div><div><br></=
div><div>For those unaware of ISO's approval process, a CD is the first step=
 in a five (?) step approval process that ultimately ends in an Internationa=
l Standard (IS). &nbsp;At CD level, the key architectural design decisions a=
re frozen, but other technical input is still possible. &nbsp;In IETF terms,=
 it's probably a maturity level that a WG I-D has after a couple of iteratio=
ns.</div><div><br></div><div>Regards,</div><div>Stephan</div><div><br></div>=
</body></html>

--B_3371191060_918748--



From garysull@microsoft.com  Fri Oct 29 11:29:43 2010
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From: Gary Sullivan <garysull@microsoft.com>
To: Stephan Wenger <stewe@stewe.org>, "rai@ietf.org" <rai@ietf.org>, "fecframe@ietf.org" <fecframe@ietf.org>, "rmt@ietf.org" <rmt@ietf.org>,  "avt@ietf.org" <avt@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
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Date: Fri, 29 Oct 2010 18:31:31 +0000
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FYI (responding to the comment about the approval process steps):  If you c=
ount producing a CD ballot text as being the first step, I would say that t=
here are basically three main steps in the current ISO/IEC approval process=
 - where the other two are the production of DIS and FDIS. (For those who m=
ay have been familiar with the way things were a couple years ago, the DIS =
stage was previously called an FCD.) After an FDIS is produced, there are n=
o changes (other than such things as typo fixes or rephrasing a garbled sen=
tence or two) - except, of course, by issuing subsequent amendments and cor=
rigenda. There is a ballot of an FDIS before it is considered a fully-appro=
ved standard, but the only outcome of that ballot is "Yes" or "No", not "Le=
t's change section 7" - and I have personally never witnessed a "No" outcom=
e on such a ballot. So, for practical purposes, an FDIS is the final standa=
rd.

The time between issuing a CD and issuing an FDIS takes a minimum of about =
9 months, but would more typically take about one year (or longer).

Best regards,

Gary Sullivan

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Steph=
an Wenger
Sent: Friday, October 29, 2010 09:58 AM
To: rai@ietf.org; fecframe@ietf.org; rmt@ietf.org; avt@ietf.org; mmusic@iet=
f.org
Cc: ietfdbh@comcast.net; lars.eggert@nokia.com
Subject: [AVT] MPEG liaison statements received

Hi all,

The IETF has received from MPEG two liaison statements for our information.=
  No actions from the IETF side are requested or required.  Both statements=
 will appear shortly on the IETF's liaison website https://datatracker.ietf=
.org/liaison/

n11632 informs us that their http streaming project has reached the Committ=
ee Draft" (CD) approval level.

n11633 informs us that the joint IPTV project with the ITU Q13/16 has also =
reached CD level.  Some of you may remember those assorted IPTV workshops o=
f the ITU and MPEG in 2008 and 2009; this is the outcome of the project sta=
rted back then.  In contrast to n11632, n11633 contains the CD text of four=
 future standards in winword format.

For those unaware of ISO's approval process, a CD is the first step in a fi=
ve (?) step approval process that ultimately ends in an International Stand=
ard (IS).  At CD level, the key architectural design decisions are frozen, =
but other technical input is still possible.  In IETF terms, it's probably =
a maturity level that a WG I-D has after a couple of iterations.

Regards,
Stephan


--_000_A7561C1FA8E37045B3D795A072417429B656TK5EX14MBXW652wingr_
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<div class=3DWordSection1>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>FYI (responding to the comment about the approval process st=
eps):&nbsp;
If you count producing a CD ballot text as being the first step, I would sa=
y
that there are basically three main steps in the current ISO/IEC approval
process &#8211; where the other two are the production of DIS and FDIS. (Fo=
r
those who may have been familiar with the way things were a couple years ag=
o,
the DIS stage was previously called an FCD.) After an FDIS is produced, the=
re
are no changes (other than such things as typo fixes or rephrasing a garble=
d
sentence or two) &#8211; except, of course, by issuing subsequent amendment=
s
and corrigenda. There is a ballot of an FDIS before it is considered a full=
y-approved
standard, but the only outcome of that ballot is &quot;Yes&quot; or &quot;N=
o&quot;,
not &quot;Let's change section 7&quot; &#8211; and I have personally never =
witnessed
a &quot;No&quot; outcome on such a ballot. So, for practical purposes, an F=
DIS
is the final standard.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>The time between issuing a CD and issuing an FDIS takes a
minimum of about 9 months, but would more typically take about one year (or=
 longer).<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Best regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Gary Sullivan<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Step=
han
Wenger<br>
<b>Sent:</b> Friday, October 29, 2010 09:58 AM<br>
<b>To:</b> rai@ietf.org; fecframe@ietf.org; rmt@ietf.org; avt@ietf.org;
mmusic@ietf.org<br>
<b>Cc:</b> ietfdbh@comcast.net; lars.eggert@nokia.com<br>
<b>Subject:</b> [AVT] MPEG liaison statements received<o:p></o:p></span></p=
>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>Hi all,<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>The IETF has received from MPEG two liaison statements for our
information. &nbsp;No actions from the IETF side are requested or required.
&nbsp;Both statements will appear shortly on the IETF's liaison website&nbs=
p;<a
href=3D"https://datatracker.ietf.org/liaison">https://datatracker.ietf.org/=
liaison</a>/<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>n11632 informs us that their http streaming project has reache=
d
the Committee Draft&quot; (CD) approval level. &nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>n11633 informs us that the joint IPTV project with the ITU Q13=
/16
has also reached CD level. &nbsp;Some of you may remember those assorted IP=
TV
workshops of the ITU and MPEG in 2008 and 2009; this is the outcome of the
project started back then. &nbsp;In contrast to n11632, n11633 contains the=
 CD
text of four future standards in winword format.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>For those unaware of ISO's approval process, a CD is the first
step in a five (?) step approval process that ultimately ends in an
International Standard (IS). &nbsp;At CD level, the key architectural desig=
n
decisions are frozen, but other technical input is still possible. &nbsp;In
IETF terms, it's probably a maturity level that a WG I-D has after a couple=
 of
iterations.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>Regards,<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'>Stephan<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.5pt;font-family:"Calibri",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

</div>

</body>

</html>

--_000_A7561C1FA8E37045B3D795A072417429B656TK5EX14MBXW652wingr_--

From xielei57471@huawei.com  Sat Oct 30 03:03:59 2010
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Date: Sat, 30 Oct 2010 18:05:43 +0800
From: Xie Lei <xielei57471@huawei.com>
To: avt@ietf.org
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Subject: Re: [AVT] The interworking issue for RTP Payload Format for AMR
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Hi
Are there any comments for this interworking issue? I hope any RFC4867 experts could give me some suggestions. Thanks. 

BR
Rock




  ----- Original Message ----- 
  From: Xie Lei 
  To: Roni Even ; avt@ietf.org 
  Sent: Tuesday, October 26, 2010 2:04 PM
  Subject: Re: [AVT]The interworking issue for RTP Payload Format for AMR


  Hi
  Yes, the TRAU can only take either CMR or CMI, and in GSM network, TRAU frame sequence must strictly be one CMR frame following with one CMI frame , however, RFC4867 allows take both CMR and CMI one packet and there is no indication to indentify the packet is CMR frame or not, that makes the problem.

  BR
  Rock
    ----- Original Message ----- 
    From: Roni Even 
    To: avt@ietf.org 
    Cc: 'Xie Lei' 
    Sent: Tuesday, October 26, 2010 1:12 PM
    Subject: FW: [AVT]The interworking issue for RTP Payload Format for AMR


    Hi,

    I assume that the problem is because in TRAU you can send either CMR or CMI while RFC4867 allows both in the same packet. Is this correct?

    Roni Even

     

    From: Xie Lei [mailto:xielei57471@huawei.com] 
    Sent: Monday, October 25, 2010 9:14 AM
    To: avt@ietf.org
    Cc: keith.drage@alcatel-lucent.com; 'Roni Even'
    Subject: [AVT]The interworking issue for RTP Payload Format for AMR

     

    Hello All

    I work on one interworking task between AMR RTP Payload and TRAU frame in TDM network, and find it is very difficult to map the Codec_Mode_Request(CMR) value in RTP to Request or Indication Flag (RIF) in TRAU frame which is defined in 3GPP 48.060/48.061. 

    As defined in RFC4867, RTP frame can take CMR and CMI(FT field) both in payload and the codec mode change period could be 20ms. However,TRAU speech frame can only take one of CMR or CMI, and the RIF should follow rigorous mode change peroid, that means, TRAU speech frame could only take CMR every 40ms and between each CMR frame, the TRAU speech frame must be CMI frame. 

    The issue happens when two consequent RTP payload takeing valide CMR value in each payload, at this scenario, we do not know how to set the RIF value. For instance, the first RTP payload takes CMR=4, and we map this CMR value to RIF flag to indicate the TRAU speech frame is for CMR, and the second RTP payload comes with CMR=1, then we really do not know how to do now. according to the RTP payload CMR valude, we should indicate RIF as CMR. However, according to the TRAU frame definition, the RIF must be CMI which following with last CMR frame.

    RFC4867 has some decription on reqirements for encoder in GSM. However, it is not enough, i think there needs some description on how to map CMR and CMI value in RTP frame to RIF defined TRAU when interworking between IP and TDM network. currentlly , we can not find such mapping table in any specification. We hope avt groud could work on this and resolve this issue.

     

    Best Regards

    Rock

     



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