
From gorry@erg.abdn.ac.uk  Mon Jul  1 00:00:30 2013
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Date: Mon, 1 Jul 2013 08:00:20 +0100
From: gorry@erg.abdn.ac.uk
To: "John Leslie" <john@jlc.net>
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Cc: Gorry Fairhurst <gorry@erg.abdn.ac.uk>, tsvwg chair <tsvwg-chairs@tools.ietf.org>, tsvwg WG <tsvwg@ietf.org>
Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-byte-pkt-congest-10 - to conclude 28th June 2013
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> Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:
>>
>> This email announces the beginning of a short working group last call
>> for draft-ietf-tsvwg-byte-pkt-congest-10, "Byte and Packet Congestion
>> Notification". This document is now thought by the authors to be ready
>> to proceed to be published as a Proposed Standard...
>
>    Alas, we do need clarity about the intended status.
>
Sorry BCP.

>    I cannot see how this draft would qualify for Proposed Standard; so
> I will not discuss that further unless someone confirms that that indeed
> is the status that will be proposed.
>
>    The document itself claims Intended status: BCP.
>
This is correct.

>    IMHO, we have been overtaken by events over the last six years; and
> to publish a BCP on this topic today would require considerable review
> and discussion. I am willing to participate in such review; but I am
> doubtful we will see quick progress with the current authors. :^(
>
>    I suggest instead publishing pretty much as-is with intended status:
> Informational. Surely the effort which has gone into it so far justifies
> a WG Informational RFC. (I'd be happiest with an added note of the
> expected work in AQM; but I certainly won't insist on that.)
>
>    I do encourage everyone interested in this to attend the AQM BoF at
> 1700 on Tuesday, July 30th.
>
>    (As a brief suggestion of issues deserving our work to publish this
> as a BCP today, I'll mention:
> - whether we are attempting to estimate time-on-the-wire when we count
>   bytes and/or packets;
> - what AQM has come to mean over the last few years; and
> - how byte and/or packet counting will interact with time-in-queue AQM.)
>
> --
> John Leslie <john@jlc.net>
>
I'll send an email closing the WGLC and listing the people who participated.

Gorry


From weixinpeng@huawei.com  Wed Jul  3 18:30:47 2013
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From: Weixinpeng <weixinpeng@huawei.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: A new draft on tunnel congestion information feedback mechanism.
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Date: Thu, 4 Jul 2013 01:30:16 +0000
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Subject: [tsvwg] A new draft on tunnel congestion information feedback mechanism.
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--_000_C5C3BB522B1DDF478AA09545169155B43CA34426nkgeml507mbxchi_
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Hi all,
         A new draft on tunnel congestion information feedback mechanism ha=
s been submitted.

Best Regards
Xinpeng.





Abstract:

   This document describes a mechanism to calculate congestion of a

   tunnel segment based on RFC 6040 recommendations, and a feedback

   protocol by which to send the measured congestion of the tunnel from

   egress to ingress router. A basic  model for measuring tunnel

   congestion and feedback is described, and a protocol for carrying the

   feedback data is outlined.



URL:             http://www.ietf.org/internet-drafts/draft-wei-tsvwg-tunnel=
-congestion-feedback-00.txt

Status:          http://datatracker.ietf.org/doc/draft-wei-tsvwg-tunnel-con=
gestion-feedback

Htmlized:        http://tools.ietf.org/html/draft-wei-tsvwg-tunnel-congesti=
on-feedback-00


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<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi all,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; A new draft on tunnel congestion information feedback mec=
hanism has been submitted.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Best Regards<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Xinpeng.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Abstract:<o:p></o:p></span><=
/p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; This document d=
escribes a mechanism to calculate congestion of a<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; tunnel segment =
based on RFC 6040 recommendations, and a feedback<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; protocol by whi=
ch to send the measured congestion of the tunnel from<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; egress to ingre=
ss router. A basic&nbsp; model for measuring tunnel<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; congestion and =
feedback is described, and a protocol for carrying the<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; feedback data i=
s outlined.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">URL:&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://www.ietf=
.org/internet-drafts/draft-wei-tsvwg-tunnel-congestion-feedback-00.txt">
http://www.ietf.org/internet-drafts/draft-wei-tsvwg-tunnel-congestion-feedb=
ack-00.txt</a><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Status:&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://datatracker.ietf.org/do=
c/draft-wei-tsvwg-tunnel-congestion-feedback">
http://datatracker.ietf.org/doc/draft-wei-tsvwg-tunnel-congestion-feedback<=
/a><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Htmlized:&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://tools.ietf.org/html/draft-wei-tsv=
wg-tunnel-congestion-feedback-00">
http://tools.ietf.org/html/draft-wei-tsvwg-tunnel-congestion-feedback-00</a=
><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_C5C3BB522B1DDF478AA09545169155B43CA34426nkgeml507mbxchi_--

From gorry@erg.abdn.ac.uk  Thu Jul  4 06:28:51 2013
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Date: Thu, 04 Jul 2013 14:28:36 +0100
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] Agenda requests for TSVWG in Berlin
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Hi all,

If you still would like to present at the planned TSVWG meeting in 
Berlin, please let the chairs know asap:

* Presentation title / draft
* Presenter
* Presentation time

- Last meeting we saw a large number of drafts, so this time we plan to 
focus must time on drafts that have seen recent discussion on the list 
(hint: If it has not been discussed on the list, then  it probably isn't 
ready for tsvwg) and those drafts that have changed significantly since 
the last meeting.

- Time for WG drafts will be allocated in precedence of other presentations.

Best wishes,

Gorry, James, and David

++++

Here is my list of current requests received - in no particular order 
and **NO** decision yet:

Agenda Requests (Berlin)

Working Group Documents:

   Byte and Packet Congestion Notification
   draft-ietf-tsvwg-byte-pkt-congest

Individual Documents:

   (Ruediger.Geib@telekom.de)
   DiffServ interconnection classes
   draft-geib-tsvwg-diffserv-intercon

   G. Shepherd
   Multicast UDP Usage Guidelines for Application Designers
   draft-shepherd-multicast-udp-guidelines-01

   Subha Dhesikan
   DSCP and other packet markings for RTCWeb QoS
   draft-dhesikan-tsvwg-rtcweb-qos-01

New work:

  (<weixinpeng@huawei.com>)
  Tunnel Congestion Feedback
  draft-wei-tsvwg-tunnel-congestion-feedback-00

  draft-iyengar-minion-concept-00
  draft-iyengar-minion-protocol-00
  draft-zmafir-tsvwg-flow-metadata-rsvp-00 (No Draft)
  draft-choukir-tsv-flow-metadata-encoding-00 (No Draft)

  MiniStack, a minimalistic operating system stack (No Draft)

Notes on other matters:

   New AD announcement

   draft-bonica-6man-frag-deprecate-00 -- no presentation requested.


From Karen.Nielsen@tieto.com  Thu Jul  4 08:39:03 2013
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To: <nishida@sfc.wide.ad.jp>, <salvatore.loreto@ericsson.com>
Date: Thu, 4 Jul 2013 18:38:49 +0300
Thread-Topic: draft-ietf-tsvwg-sctp-failover: comments and suggestions
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Subject: Re: [tsvwg] draft-ietf-tsvwg-sctp-failover: comments and suggestions
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Hi All,

Please find some additional comments inline below.

>
>- Section 5.1: CWND Handling for Potentially Failed Destinations
>
<snip>
>
>I agree with you to stick to the behavior of RFC4960 as default behavior. =
We've
>been exploring several possibilities for performance improvement and discu=
ssing
>using 2 MTU might be an idea, although we think we'll need further discuss=
ions.
>Preethi might have some more comments on this point.
>

Please accept the following addition comments:

The motivation for the 2MTU, I now understand, is to aim to mitigate unwant=
ed effects of=20
path bouncing resulting from a spurious T3-timeout followed by failover=20
and subsequent "immediate" switchback operation of RFC4960 once HB ACK
is acknowledged within the back-off'ed RTO.=20

While I think that one can very well say in this document that the CWND cou=
ld be set to 2MTU=20
in case the first HB sent on a PF path is successfully ack'ed, then I would=
 like to note that=20
this then is motivated by a desire, first and foremost, to be backward comp=
atible with RC4960, more than it
really necessarily addresses how to best mitigate throughput degradation ef=
fects of spurious T3-timeouts.
Indeed for implementations that operate with different switchback modes tha=
n the immediate switchback operation of RFC4960,
 (e.g. operate with certain permanent failover modes), then the set of the =
CWND to 2 MTU on the previous PF path=20
is not of a big significance as it can be most optimal, from a performance =
perspective,
to stay on the new data carrying path, which at this point in time may have=
 a CWND of IW, IW+1MTU or even higher.
I think that the motivation for setting the CWND to 2MTU possibly should be=
 made conditionally on whether the SCTP indeed follows
the immediate switchback operation of RFC4960.

One further observation is that in a situation where it is only the 2dn or =
the 3rd (or 4th..) of the HBs sent on the PF path, that is successful, then=
=20
setting the CWND to 2 MTU may actually constitute a decrease of the CWND. T=
his is the case if the implementation=20
follows RC4960 and raise the CWND to IW (4MTU) after an idle period of one =
RTO following the T3 timeout.
When in the text it is said that the CWND should be set to 1MTU at the stat=
e transition from PF to ACTIVE does this then very explicitly
addresses the fact that unsuccessful HBs, sent on a PF path, shall keep the=
 CWND down at 1MTU as otherwise data failures would ?
If so then I think that it could be good to add this clarification.

Final thoughts:

We have in real deployment experienced fault situations where HBs consisten=
tly were able to get through on the primary path,=20
but where the path broke due to congestion once it got loaded with data. Th=
ere are different ways in which one can improve the
protocol operation to best tackle such a situation. Our present solution, w=
hich is to deploy a permanent failover mode together with PF
and looking for a way to add reliability to the T3 timeouts, may not be the=
 ultimate way to solve this, but I would like to note that=20
in such a situation then consistently returning to the primary path and set=
ting CWND =3D 2MTU is not the right thing to do.


>- Section 5.3: Permanent Failover
>
>We have good experiences from live networks with running SCTP in a "no
>switchback" mode of operation with no automatic migration back to a prefer=
red
>primary path. Once the primary path is left due to it being potentially fa=
iled, then a
>selected alternative path automatically becomes the new primary path.
>
>This mode of operation corresponds to a simplified variant of the permanen=
t
>failover operation proposed in [CAR002] with alpha =3D beta =3D
>SCTP_PEER_ADDR:THLDS. The mode of operation is run with notifications of
>change of primary path being provided to upper layers via RFC6458
>SCTP_PEER_ADDR_CHANGE semantics (SCTP_ADDR_MADE_PRIM).
>
>The mode of operation does not suit all purposes, but in networks environm=
ents
>where a number of equally preferred paths are available, then by this mode=
 of
>operation one minimizes the effects of unnecessary path bouncing and the t=
hereby
>associated effects of throughput degradation due to path changes and slow =
start
>operation.
>
>Based on our experiences we would recommend and support that the Permanent
>Failover mode of operation, at least in the simple variant described above=
, is
>promoted as a valid alternative approach. For the purpose of activation of=
 this
>mode of operation a switchback (or failover) mode operation parameter coul=
d be
>introduced as configurable by the introduction of an appropriate new read/=
write
>socket option in the Socket API.
>
>
>Yes. I agree that there will be a situation where Permanent Failover work =
effectively.
>OTOH, supporting this=A0prevents us from keeping RFC4960 behavior as RFC49=
60
>doesn't have this mode.
>But, I believe this is a very interesting point and we would like to get m=
ore feedback
>in order to think about this more.
>

I understand to continue to support RFC4960 behaviour, but I suppose that o=
ne can add to RFC4960
behaviour :-) We have the simple "no switchback mode" described above runni=
ng in deployment.


>- Section 5.4: The impact of HB timeouts in association error counter
>
>The draft recommends for excluding HB timeouts from contributing to the
>(association) error counter. We fully support this approach for all timeou=
ts of HBs
>that are send to probe the state of destinations addresses that are in PF =
state and
>we would even propose for the draft to demand that generally timeouts stem=
ming
>from HBs, which are not sent on the Primary Path, will be excluded from th=
e
>association error counter.
>
>Thereby the association error counter and thus the association fate reflec=
ts the
>forward progress of the data transmission to peer endpoint while at the sa=
me time it
>will be appropriately isolated from timeouts of HBs sent on non-data carry=
ing paths.
>
>Allowing timeout of HBs sent on the Primary Path to impact the association=
 error
>counter will serve to validate the aliveness of the peer endpoint when the=
re is no
>data in transit.
>
>This suggestion follows the thought expressed in http://www.ietf.org/mail-
>archive/web/tsvwg/current/msg11294.html and it is thus believed to fulfil =
the original
>intend of RFC4960.
>
>This is a bit difficult point for me. I don't know much about the original=
 intention of
>RFC4960, but if we follow what is written there, i think we need to increm=
ent
>association error count. (Please correct me if I misunderstand this point)=
 So, my
>personal preference is to set bigger AMR to avoid early termination so tha=
t we can
>keep the RFC4960 behavior.
>However, we don't mind discussing to update the logic described in 4960 fo=
r more
>sophisticated solution.
>

Thanks !

Yes, I concur that the text of section 8.1 of RFC4960 stipulates that all H=
B Acks and HB timeouts=20
influence the association error counter. But from the referenced discussion=
 thread,
 http://www.ietf.org/mail-archive/web/tsvwg/current/msg11294.html  (simply =
copying it in here to avoid repeating the arguments)=20
 it also seems that by purpose no keywords were used when writing these par=
agraphs.=20
Further as the Potentially Feature was not defined at the time, I am not su=
re that
it would be valid to say that the text of RFC4960 apply (at least) to the H=
Bs sent as part of this function.

>From our perspective what we would like to support and see happening is=20
that all HBs, whether sent as part of the PF function or not, influence the=
 association error counter=20
if and only if they are transmitted on the Primary Path (and then with stan=
dard logic running making sure that all data transmissions=20
prevent (or cancel in race conditions) any influence from HBs).

Thereby the association error counter will be influenced by (and will thus =
reflect) the data transmission only.
Except in idle situation where HBs on Primary Path will serve to update the=
 association error counter in a similar manner as data would.
Thereby one get a very clean and reliable implementation of the association=
 error counter. By reliable, I mean that one can use
the same AMR value regardless of the level of HB probing that one have acti=
vated and regardless of whether the Potentially Failed feature is activated=
 or not.
Such an implementation also makes the AMR value be independent from the num=
ber of paths supported in the association context.=20
One can choose to use a value of AMR proportional with the number of paths =
in order to have all paths be tried a number of times before association cl=
osure, but
with the choice suggested, one is not forced to "artificially" raise the AM=
R value to make room for the HBs that will fire as part of the potentially =
failed feature as well
as for some fraction of the Path Monitoring HBs that can happen to fire dur=
ing the same timeframe.

An additional thought is that in the odd cases where HBs get through wherea=
s data do not, then I think that it is contra-productive to have the associ=
ation error counter be reset
as long as data consistently fail.=20

I, personally, does not so much like the suggestion to use a "maximal" or "=
bigger" AMR. I think that it will be complex to deduce exactly
what failure time (aka number of data retransmission)  that a certain AMR v=
alue will guarantee you,=20
whereas when only data transmission influences the AMR, things will be clea=
n, simple and predictable.

I would be very happy if we by this communication could solicit information=
 from others.

>
>- Section 5.1 Thoughts on the destination address state machine
>
>In RFC4960 address state machine, notification of cease of use of the curr=
ent
>primary path as transmission path for new data is provided to upper layer =
via
>(coupled with) the announcement of the destination as being UNREACHABLE. W=
ith
>the introduction of the potentially failed state as a semi-hidden state (n=
ot resulting in
>notified state transition to UL) the information on the fail-over to use a=
 new data
>carrying path is lost.
>While there may be merits in suppressing notifications on potentially fail=
ed
>addresses, then we believe that it is unfortunate that the notification of=
 failover from
>usage of the primary path for new data is lost.
>This is especially problematic in path bouncing scenarios where the primar=
y path is
>unstable but still works some. E.g., HBs may go through, but the path brea=
ks (data
>transmission fails) as soon as it is more loaded. In such a situation no n=
otifications
>may be generated to UL, even if new data transmission consistently cycles =
to and
>from the primary path. The solution to such path bouncing scenarios will h=
ave to
>come from more advanced path failover mechanism, like [CAR002] (and would =
not
>be solved just by notification), but it would be a great help if appropria=
te
>notifications were provided.
>
>I think it will be OK to send notification as long as it's a new notificat=
ion type. But
>we should not send notifications defined in RFC6458.

ok - I assume this is for backward compatible reasons.=20

>I guess having a new notification type won't cause problems, but I would l=
ike to get
>more feedback on this.

Ok - we can come back with a suggestion.

Thanks a lot !

BR, Karen

>
>Thanks!
>--
>Yoshifumi


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To: "<tsvwg@ietf.org>" <tsvwg@ietf.org>, Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [tsvwg] Fwd: Confirmation for Auto-Post of I-D draft-shepherd-multicast-udp-guidelines
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Can I have an agenda slot in Berlin to discuss
draft-shepherd-multicast-udp-guidelines?

Thanks,
Greg

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Date: Wed, Jun 12, 2013 at 10:30 AM
Subject: Confirmation for Auto-Post of I-D
draft-shepherd-multicast-udp-guidelines
To: Greg Shepherd <gjshep@gmail.com>



Hi,

The IETF datatracker draft submission service has received your draft
draft-shepherd-multicast-udp-guidelines-01, and requires a
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Please follow this link to the page where you can confirm the posting:
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        The IETF Secretariat
        through the draft submission service

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<div dir=3D"ltr">Can I have an agenda slot in Berlin to discuss draft-sheph=
erd-multicast-udp-guidelines?<div><br></div><div>Thanks,</div><div>Greg<br>=
<div><br><div class=3D"gmail_quote">---------- Forwarded message ----------=
<br>
From: <b class=3D"gmail_sendername">IETF I-D Submission Tool</b> <span dir=
=3D"ltr">&lt;<a href=3D"mailto:idsubmission@ietf.org">idsubmission@ietf.org=
</a>&gt;</span><br>Date: Wed, Jun 12, 2013 at 10:30 AM<br>Subject: Confirma=
tion for Auto-Post of I-D draft-shepherd-multicast-udp-guidelines<br>
To: Greg Shepherd &lt;<a href=3D"mailto:gjshep@gmail.com">gjshep@gmail.com<=
/a>&gt;<br><br><br><br>
Hi,<br>
<br>
The IETF datatracker draft submission service has received your draft<br>
draft-shepherd-multicast-udp-guidelines-01, and requires a<br>
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Please follow this link to the page where you can confirm the posting:<br>
Confirmation URL: <a href=3D"http://datatracker.ietf.org/submit/status/5114=
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<br>
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<br>
=A0 =A0 =A0 =A0 The IETF Secretariat<br>
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This email summarises the results of the WGLC for 
draft-ietf-tsvwg-byte-pkt-congest-10.

This document had been approved by the IESG for publication as an RFC, 
butthe just-concluded WGLC revealed that updates to the document are
still potentially needed. The document is therefore returned to the WG 
work on these changes.

The resulting ID will requirereview in a short WGLCto confirm the final 
document, followed by an IETF last call by the community at large.

We hope a new revision will be available shortly.

Best wishes,

Gorry (as document shepherd)

From madanmohanm@huawei.com  Sun Jul  7 23:00:06 2013
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From: Madan Mohan Mishra <madanmohanm@huawei.com>
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--_000_2F3BA9016646EE43862F506BAC2D49A92CE484C8szxeml537mbxchi_
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Hi Group,

    I have a doubt in one of the scenario related to restart duing shutdown=
 which is not very clear to my understanding as per the RFC4960. The scenar=
io is as below:



Clause as per the RFC:

---------------------------------------------------------------------------=
----------

9.2.  Shutdown of an Association
"If an endpoint is in the SHUTDOWN-ACK-SENT state and receives an INIT
   chunk (e.g., if the SHUTDOWN COMPLETE was lost) with source and
   destination transport addresses (either in the IP addresses or in the
   INIT chunk) that belong to this association, it should discard the
   INIT chunk and retransmit the SHUTDOWN ACK chunk."

---------------------------------------------------------------------------=
----------



Network scenario as per the above RFC snippet:

SCTP-A                                                                    S=
CTP-B

Local IP List: L-IP1, L-IP2                                             Loc=
al IP List: R-IP1, R-IP2

ESTABLISHED                                                            ESTA=
BLISHED

---------------------------SHUT-DOWN (L-IP1->R-IP1)-------------------->

<-------------------------SHUT-DOWN-ACK (R-IP1<- L-IP1)--------------

---------------------------SHUT-DOWN-COMP (L-IP1->R-IP1)-----lost in networ=
k

Now, local end point rebooted

It might create a new association as below:

local IP addresses.

Local IP List: L-IP2, L-IP3

----------------------------INIT (L-IP2->R-IP1)----------------------------=
---->

                                                                           =
     SHUT-DOWN-ACK sent state

                                                                           =
     As per RFC, upon receiving INIT message

                                                                           =
     in SHUT-DOWN-ACK-SENT state, it should

                                                                           =
     retransmit SHUT-DOWN-ACK and discard INIT message.



<-----------SHUT-DOWN-ACK (L-IP1<-R-IP1)------------------- This mesage wil=
 be discarded at L-IP1, since there will be

                                                                           =
                     no association in L-IP1



The destination IP address of last SHUT-DOWN-ACK message is no more exist. =
So , all

the SHUT-DOWN-ACK messages sent by SCTP-B will be discarded in network.



RFC4960 does not mention explicitly to handle this kind of situation?



As, retransmission of SHUT-DOWN-ACK message is not due to T2 timer expiry, =
path switching

is not applicable. So, SHUT-DOWN-ACK message will always be sent on the pri=
mary address.



So, I believe, SCTP-B should send SHUT-DOWN-ACK message to the same destina=
tion address

which is exist in both INIT and current association destination address lis=
t so that the message will

not be discarded.



Please share your opinion.



Thanks and Regards,

Madan



--_000_2F3BA9016646EE43862F506BAC2D49A92CE484C8szxeml537mbxchi_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style id=3D"owaParaStyle">P {
	MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px
}
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</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi Group,</p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;I have a doubt in one of the scenario related to=
 restart duing shutdown which is not very clear to my understanding as per =
the&nbsp;RFC4960. The scenario is as below:</p>
<p>&nbsp;</p>
<p>Clause as per the RFC:</p>
<p>------------------------------------------------------------------------=
-------------</p>
<p>9.2.&nbsp; Shutdown of an Association<br>
&quot;If an endpoint is in the SHUTDOWN-ACK-SENT state and receives an INIT=
<br>
&nbsp;&nbsp; chunk (e.g., if the SHUTDOWN COMPLETE was lost) with source an=
d<br>
&nbsp;&nbsp; destination transport addresses (either in the IP addresses or=
 in the<br>
&nbsp;&nbsp; INIT chunk) that belong to this association, it should discard=
 the<br>
&nbsp;&nbsp; INIT chunk and retransmit the SHUTDOWN ACK chunk.&quot;</p>
<p>------------------------------------------------------------------------=
-------------</p>
<p>&nbsp;</p>
<p>Network scenario as per the above RFC snippet:</p>
<p>SCTP-A&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;SCTP-B</p>
<p>Local IP List: L-IP1, L-IP2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
Local IP List: R-IP1, R-IP2</p>
<p>ESTABLISHED&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;E=
STABLISHED</p>
<p>---------------------------SHUT-DOWN (L-IP1-&gt;R-IP1)------------------=
--&gt;</p>
<p>&lt;-------------------------SHUT-DOWN-ACK (R-IP1&lt;- L-IP1)-----------=
---</p>
<p>---------------------------SHUT-DOWN-COMP (L-IP1-&gt;R-IP1)-----<strong>=
lost in network</strong></p>
<p>Now, local end point rebooted </p>
<p>It might create a new association as below:</p>
<p>local IP addresses.</p>
<p>Local IP List: L-IP2, L-IP3</p>
<p>----------------------------INIT (L-IP2-&gt;R-IP1)----------------------=
----------&gt;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
SHUT-DOWN-ACK sent state</font></p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;As per RFC, upon receiving INIT message</p=
>
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;in SHUT-DOWN-ACK-SENT state, it should</p>
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;retransmit SHUT-DOWN-ACK and discard INIT =
message.</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"><strong></strong></font>&nbsp;=
</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">&lt;-----------SHUT-DOWN-ACK (=
L-IP1&lt;-R-IP1)-------------------
<strong>This mesage wil be discarded at L-IP1, since there will be</strong>=
</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"><strong>&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;no association in L-IP1</strong></font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">The destination IP address of =
last SHUT-DOWN-ACK message is no more exist. So , all</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">the SHUT-DOWN-ACK messages sen=
t by SCTP-B will be discarded in network.</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">RFC4960 does not mention expli=
citly to handle this kind of situation?</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">As, retransmission of SHUT-DOW=
N-ACK message is not due to T2 timer expiry, path switching</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">is not applicable. So, SHUT-DO=
WN-ACK message will always be sent on the primary address.</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">So, I believe, SCTP-B should s=
end SHUT-DOWN-ACK message to the same destination address</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">which is exist in both INIT an=
d current association destination address list so that the message will
</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">not be discarded.</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">Please share your opinion.</fo=
nt></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff"></font>&nbsp;</p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">Thanks and Regards,</font></p>
<p><font style=3D"BACKGROUND-COLOR: #ffffff">Madan</p>
</font>
<p>&nbsp;</p>
</div>
</body>
</html>

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From gorry@erg.abdn.ac.uk  Mon Jul  8 02:05:42 2013
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Subject: [tsvwg] Some comments on draft-shepherd-multicast-udp-guidelines-01.txt
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Greg,

I like the idea of publishing something on the topic in this draft, and 
think this a valuable addition to what is said in RFC 5405. It's not 
that multicast is hard - it's just slightly different.

However I do see the need for additional/different congestion control 
requirements for multicast.  Was that the intention, and that the basic 
guidance for using UDP in RFC 5405 also applies?  - I'd have expected 
that RFC 5405 is normatively cited as a list of basic UDP requirements?

This email was as a TSVWG contributor, I'm interested in multicast 
transport/infrastructure (not as a WG Chair).

Gorry


---

Introduction:
I suspect any advice also applies to use of UDP-Lite with multicast?

---

At the end of section 2.1 I see the assertion here that multicast 
tunnels should not be congestion-controlled, making this the 
responsibility of applications that use the tunnels.

While I can see this has been the case with traditional tunnels, the key 
question that needs to be thought about here (I think raised by Magnus?) 
is whether this is right also for automatic multicast tunnels - or 
whether a circuit-breaker approach, as discussed for example with PW3E, 
is more appropriate.

I'm not sure if the IETF should simply ignore protocols designed for 
"private" well-managed networks. For unicast, such traffic now 
frequently traverses VPNs and tunnels via the general Internet. Good 
advice is important, although we may of course be unable to place 
"requirements".

As above for both unicast and multicast, I think the issue here is that 
if the "private" traffic is not CC'ed then the tunnels between private 
networks do not use CC. This then means that we need to think about how 
to protect the unicast CC traffic from this impact. Is this a network 
operational responsibility (e.g. alarms and rate limits) or a transport 
responsibility (e.g. circuit-breakers and CC)?

As I recall, the aggregate multicast tunnel traffic may be argued to 
result in a significant proportion of the internet traffic through a 
home gateway. This question therefore seems relevant to various drafts 
including: AMT and draft-ietf-pwe3-p2mp-pw-requirements-02.txt !!!

---

The multicast congestion control space faces similar challenges to the 
multicast reliability space.

I think the solution space is slightly bigger than the present draft 
says. To me, the key issue with Multicast in the General Internet is 
that there are (at least) three fundamental differences:

* The multicast tree is often not heterogeneous - this implies there may 
be a wide range of path characteristics. This has implications on the 
methods used for RMT e.g. in 3.1 of RFC 5405, it says many applications 
"cannot maintain a reliable RTT estimate for a destination". This often 
applies to the set of destinations in an RMT session or a multicast RTP 
session.

* The heterogeneity also means that some receivers may be receiving 
significantly more loss than the rest of the group. In talks on 
multicast I've called these "crying babies" (by which I mean recipients 
that can gain the attention of the sender and may need special care). It 
seems to me that many multicast apps try to exclude these from the group 
- e.g. start a unicast session to this particular endpoint, or simply 
don't distribute a security key to these receivers.

* Any receiver *anywhere* can potentially try to receive any group. 
That's not a transport problem. It may imply the need for rate limits on 
the aggregate multicast service, but there is no way at the transport 
layer to stop a join message propagating to the next hop router. It also 
points to the need for multicast operators to monitor their 
infrastructure (which we do, and I'm sure others do - since specific 
multicast transport issues can only be debugged when you understand the 
health of the underlying network infrastructure).

While all this may initially seem problematic, the overall goal of 
multicast is for users to get useful data. This implies there is an 
incentive for apps to not to do silly things that will render their 
service useless.

---

Effective MTU: I saw Lars' comment on path MTU discovery (relating to 
3.2 of RFC 5405) - somehow this needs to be combined with the 1200 byte 
rule of thumb that many of us have been using for multicast for years.

I recall some multicast deployments shying away from using PMTUD because 
it was hard to protect the infrastructure when there was no specific 
destination endpoint. Hence this was a DoS vulnerability. (Just because 
a receiver can see a packet from a multicast tree doesn't mean it should 
be allowed to drive-down the sender MTU to a trivial value). Could the 
draft explain whether this is an issue or non-issue?

---

Security

See above on the responsibility for sharing the limited capacity 
available in a typical home gateway.

One of the great things about multicast is RPF checks - a great way to 
protect from some DOS attacks. RPF checks in multicast REALLY help to 
make this manageable!

IPsec and other security mechanisms work with multicast - but they need 
multicast key management techniques.

Maybe the draft should say something about firewalls - pointing them to 
support multicast!

---

NITS

I suspect you do not need to cite this? 
[I-D.narten-iana-considerations-rfc2434bis].



From Michael.Tuexen@lurchi.franken.de  Mon Jul  8 02:33:46 2013
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: Madan Mohan Mishra <madanmohanm@huawei.com>
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On Jul 8, 2013, at 7:59 AM, Madan Mohan Mishra <madanmohanm@huawei.com> =
wrote:

> Hi Group,
>     I have a doubt in one of the scenario related to restart duing =
shutdown which is not very clear to my understanding as per the RFC4960. =
The scenario is as below:
> =20
> Clause as per the RFC:
> =
--------------------------------------------------------------------------=
-----------
> 9.2.  Shutdown of an Association
> "If an endpoint is in the SHUTDOWN-ACK-SENT state and receives an INIT
>    chunk (e.g., if the SHUTDOWN COMPLETE was lost) with source and
>    destination transport addresses (either in the IP addresses or in =
the
>    INIT chunk) that belong to this association, it should discard the
>    INIT chunk and retransmit the SHUTDOWN ACK chunk."
> =
--------------------------------------------------------------------------=
-----------
> =20
> Network scenario as per the above RFC snippet:
> SCTP-A                                                                 =
   SCTP-B
> Local IP List: L-IP1, L-IP2                                            =
 Local IP List: R-IP1, R-IP2
> ESTABLISHED                                                            =
ESTABLISHED
> ---------------------------SHUT-DOWN =
(L-IP1->R-IP1)-------------------->
> <-------------------------SHUT-DOWN-ACK (R-IP1<- L-IP1)--------------
> ---------------------------SHUT-DOWN-COMP (L-IP1->R-IP1)-----lost in =
network
> Now, local end point rebooted
> It might create a new association as below:
> local IP addresses.
> Local IP List: L-IP2, L-IP3
> ----------------------------INIT =
(L-IP2->R-IP1)-------------------------------->
>                                                                        =
         SHUT-DOWN-ACK sent state
>                                                                        =
         As per RFC, upon receiving INIT message
>                                                                        =
         in SHUT-DOWN-ACK-SENT state, it should
>                                                                        =
         retransmit SHUT-DOWN-ACK and discard INIT message.
> =20
> <-----------SHUT-DOWN-ACK (L-IP1<-R-IP1)------------------- This =
mesage wil be discarded at L-IP1, since there will be
>                                                                        =
                         no association in L-IP1
> =20
> The destination IP address of last SHUT-DOWN-ACK message is no more =
exist. So , all
> the SHUT-DOWN-ACK messages sent by SCTP-B will be discarded in =
network.
> =20
> RFC4960 does not mention explicitly to handle this kind of situation?
Well, you can consider the SHUTDOWN-ACK as a response to the received =
INIT in
this particular case. That would mean that it is a good idea to send the =
SHUTDOWN-ACK
to the source address of the received INIT. This would resolve your =
issue here.

Best regards
Michael
> =20
> As, retransmission of SHUT-DOWN-ACK message is not due to T2 timer =
expiry, path switching
> is not applicable. So, SHUT-DOWN-ACK message will always be sent on =
the primary address.
> =20
> So, I believe, SCTP-B should send SHUT-DOWN-ACK message to the same =
destination address
> which is exist in both INIT and current association destination =
address list so that the message will
> not be discarded.
> =20
> Please share your opinion.
> =20
> Thanks and Regards,
> Madan
> =20


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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
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From fluffy@iii.ca  Mon Jul  8 16:36:37 2013
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Subject: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when =
it was an individual draft then WG draft in another WG and I don't think =
there is much more to say about it. This is one of the normative =
dependencies of the webrtc work and that stuff is already getting =
implemented in browsers so I want to move this along in an expedient =
fashion.=20

I think it is ready for WGLC. If there are any issues people think need =
to be discussed before WGLC, please let us know so we can resolve them =
in Berlin.

Thanks, Cullen


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On Mon, Jul 8, 2013 at 4:35 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>
> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when =
it was an individual draft then WG draft in another WG and I don't think th=
ere is much more to say about it. This is one of the normative dependencies=
 of the webrtc work and that stuff is already getting implemented in browse=
rs so I want to move this along in an expedient fashion.
>
> I think it is ready for WGLC. If there are any issues people think need t=
o be discussed before WGLC, please let us know so we can resolve them in Be=
rlin.
>
> Thanks, Cullen
>

Note: The sample code in the document does not include correct code
for ipv6, which is more or less:

IPv6 DiffServ

IPv4   IPPROTO_IP     IP_TOS
IPv6   IPPROTO_IPV6   IPV6_TCLASS

So, to set the DSCP in IPv6 you must use IPv6 Traffic Class field.

int ds =3D IPTOS_DSCP_AF11;
int rc =3D setsockopt(fd, IPPROTO_IPV6, IPV6_TCLASS, &ds, sizeof(ds));

see:

http://www.bufferbloat.net/issues/249

for more details

--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From fluffy@iii.ca  Mon Jul  8 17:28:30 2013
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From: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Good catch - thank you.

On Jul 8, 2013, at 5:00 PM, Dave Taht <dave.taht@gmail.com> wrote:

> On Mon, Jul 8, 2013 at 4:35 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>>=20
>> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively =
when it was an individual draft then WG draft in another WG and I don't =
think there is much more to say about it. This is one of the normative =
dependencies of the webrtc work and that stuff is already getting =
implemented in browsers so I want to move this along in an expedient =
fashion.
>>=20
>> I think it is ready for WGLC. If there are any issues people think =
need to be discussed before WGLC, please let us know so we can resolve =
them in Berlin.
>>=20
>> Thanks, Cullen
>>=20
>=20
> Note: The sample code in the document does not include correct code
> for ipv6, which is more or less:
>=20
> IPv6 DiffServ
>=20
> IPv4   IPPROTO_IP     IP_TOS
> IPv6   IPPROTO_IPV6   IPV6_TCLASS
>=20
> So, to set the DSCP in IPv6 you must use IPv6 Traffic Class field.
>=20
> int ds =3D IPTOS_DSCP_AF11;
> int rc =3D setsockopt(fd, IPPROTO_IPV6, IPV6_TCLASS, &ds, sizeof(ds));
>=20
> see:
>=20
> http://www.bufferbloat.net/issues/249
>=20
> for more details
>=20
> --=20
> Dave T=E4ht
>=20
> Fixing bufferbloat with cerowrt: =
http://www.teklibre.com/cerowrt/subscribe.html


From hannes.tschofenig@gmx.net  Tue Jul  9 00:51:16 2013
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA512

Hi Cullen,=20

this document is still an individual draft and so I am not sure what you =
mean by working group last call.=20

- =46rom a content point of view I am not sure how useful it is. I =
shared my views about how this applies to LTE; IMHO it is not =
applicable.

It would be so great if we could solve all QoS challenges by defining =
yet another set of DiffServ code points.=20

Ciao
Hannes

On Jul 9, 2013, at 2:35 AM, Cullen Jennings wrote:

>=20
> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively =
when it was an individual draft then WG draft in another WG and I don't =
think there is much more to say about it. This is one of the normative =
dependencies of the webrtc work and that stuff is already getting =
implemented in browsers so I want to move this along in an expedient =
fashion.=20
>=20
> I think it is ready for WGLC. If there are any issues people think =
need to be discussed before WGLC, please let us know so we can resolve =
them in Berlin.
>=20
> Thanks, Cullen
>=20

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From gorry@erg.abdn.ac.uk  Tue Jul  9 01:03:41 2013
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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On 09/07/2013 00:35, Cullen Jennings wrote:
>
> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when it was an individual draft then WG draft in another WG and I don't think there is much more to say about it. This is one of the normative dependencies of the webrtc work and that stuff is already getting implemented in browsers so I want to move this along in an expedient fashion.
>
> I think it is ready for WGLC. If there are any issues people think need to be discussed before WGLC, please let us know so we can resolve them in Berlin.
>
> Thanks, Cullen
>

In the email above you ask for WGLC. The first step is for the group to 
consider adoption. As I recall, this draft was brought to TSVWG for 
consideration because it discusses transport mappings to QoS.

I understand from the draft this proposes default mappings for an 
RTCWeb use cases. As I see it, the present draft:

* Advocates use of separate QoS code points for different streaming 
components

* Describes both endpoint handling of priority for the media and 
transport priority/precedence (aka DSCP).

* Proposes formalising a mapping between specific DSCPs and subnetwork 
QoS functions (LTE, WiFi - but not general Ethernet).

Please read this draft and comment on the list - we will table this for 
discussion at the next IETF meeting.

Gorry



From andrewmcgr@gmail.com  Tue Jul  9 02:48:20 2013
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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So, there is a significant bug in the mappings proposed.  That is this:
there is a substantial deployed base of home routers and other network
devices that use the Linux kernel's default output queue pfifo_fast, which
uses a mapping derived from RFC 1349 (documented in the linux man page
tc_prio, see for example http://linux.die.net/man/8/tc-prio)   If the
proposed mappings don't work with that queue discipline, they won't work on
the deployed base.  pfifo_fast has then unfortunate properties of ignoring
the top 3 bits of the tos byte, and sometimes paying attention to one of
the ECN bits in choosing a queueing band.

In effect, since Linux has produced a de-facto standard codepoint
assignment for the edge of the network, we had better recommend something
compatible.  The proposed mapping uses EF for all audio, which will create
a priority inversion and have all the Linux devices on the edge classify
that traffic as 'bulk interactive', lower than the interactive video at
AF4x in the 'interactive' band.  This is probably not what we want.

I've filled out the linux band numbers in table 1 from the draft; the third
number in each table cell is the linux band, smaller number is higher
priority:
+-----------------------+-------------+-------------+-------------+
| Media Type            |    Low      |   Medium    |    High     |
+-----------------------+-------------+-------------+-------------+
| Audio                 |  46 (EF)  1 |  46 (EF)  1 |  46 (EF)  1 |
| Interactive Video     | 38 (AF43) 1 | 36 (AF42) 0 | 34 (AF41) 2 |
| Non-Interactive Video | 26 (AF33) 1 | 28 (AF32) 0 | 30 (AF31) 2 |
| Data                  |  8 (CS1)  1 |   0 (BE)  1 | 10 (AF11) 2 |
+-----------------------+-------------+-------------+-------------+
      Table 1 - Media Type, Priority, and DSCP Mapping

As can readily be seen, there are a number of priority inversions there.
 Effectively, RFC 4594 doesn't work as designed at the edge of the network,
and we have to go back to RFC 1349, or at least a restricted set of RFC
4594 codepoints.

Effectively, Linux has given us a priority series that reads:
AFx2>AFx3>AFx1>(CS1 or BE)

So, for the purpose of working with both Linux routers and RFC 4594, the
useful series of codepoints are AF42>AF33>AF21>(CS1 or BE), and it becomes
an exercise of deciding where to place those codepoints in the table.  I'd
rather use CS1 than BE because it at least states that the application is
doing QoS marking.

Aside from fixing the actual mappings, though, the draft is a great idea
and really should be published as soon as reasonably possible.


On Tue, Jul 9, 2013 at 6:03 PM, Gorry Fairhurst <gorry@erg.abdn.ac.uk>wrote:

> On 09/07/2013 00:35, Cullen Jennings wrote:
>
>>
>> The draft-dhesikan-tsvwg-rtcweb-**qos draft was discussed extensively
>> when it was an individual draft then WG draft in another WG and I don't
>> think there is much more to say about it. This is one of the normative
>> dependencies of the webrtc work and that stuff is already getting
>> implemented in browsers so I want to move this along in an expedient
>> fashion.
>>
>> I think it is ready for WGLC. If there are any issues people think need
>> to be discussed before WGLC, please let us know so we can resolve them in
>> Berlin.
>>
>> Thanks, Cullen
>>
>>
> In the email above you ask for WGLC. The first step is for the group to
> consider adoption. As I recall, this draft was brought to TSVWG for
> consideration because it discusses transport mappings to QoS.
>
> I understand from the draft this proposes default mappings for an RTCWeb
> use cases. As I see it, the present draft:
>
> * Advocates use of separate QoS code points for different streaming
> components
>
> * Describes both endpoint handling of priority for the media and transport
> priority/precedence (aka DSCP).
>
> * Proposes formalising a mapping between specific DSCPs and subnetwork QoS
> functions (LTE, WiFi - but not general Ethernet).
>
> Please read this draft and comment on the list - we will table this for
> discussion at the next IETF meeting.
>
> Gorry
>
>
>

--089e0158b7947faabf04e1110f4b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">So, there is a significant bug in the mappings proposed. =
=A0That is this: there is a substantial deployed base of home routers and o=
ther network devices that use the Linux kernel&#39;s default output queue p=
fifo_fast, which uses a mapping derived from RFC 1349=A0(documented in the =
linux man page tc_prio, see for example <a href=3D"http://linux.die.net/man=
/8/tc-prio">http://linux.die.net/man/8/tc-prio</a>)=A0 =A0If the proposed m=
appings don&#39;t work with that queue discipline, they won&#39;t work on t=
he deployed base. =A0pfifo_fast has then unfortunate properties of ignoring=
 the top 3 bits of the tos byte, and sometimes paying attention to one of t=
he ECN bits in choosing a queueing band.<div>
<br></div><div>In effect, since Linux has produced a de-facto standard code=
point assignment for the edge of the network, we had better recommend somet=
hing compatible. =A0The proposed mapping uses EF for all audio, which will =
create a priority inversion and have all the Linux devices on the edge clas=
sify that traffic as &#39;bulk interactive&#39;, lower than the interactive=
 video at AF4x in the &#39;interactive&#39; band. =A0This is probably not w=
hat we want.</div>
<div><br></div><div>I&#39;ve filled out the linux band numbers in table 1 f=
rom the draft; the third number in each table cell is the linux band, small=
er number is higher priority:</div><div><div>+-----------------------+-----=
--------+-------------+-------------+</div>
<div>| Media Type =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0Low =A0 =A0 =A0| =A0 Medi=
um =A0 =A0| =A0 =A0High =A0 =A0 |</div><div>+-----------------------+------=
-------+-------------+-------------+</div><div>| Audio =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 | =A046 (EF) =A01 | =A046 (EF) =A01 | =A046 (EF) =A01 |</div>
<div>| Interactive Video =A0 =A0 | 38 (AF43) 1 | 36 (AF42) 0 | 34 (AF41) 2 =
|</div><div>| Non-Interactive Video | 26 (AF33) 1 | 28 (AF32) 0 | 30 (AF31)=
 2 |</div><div>| Data =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| =A08 (CS1) =A01 =
| =A0 0 (BE) =A01 | 10 (AF11) 2 |</div>
<div>+-----------------------+-------------+-------------+-------------+</d=
iv><div>=A0 =A0 =A0 Table 1 - Media Type, Priority, and DSCP Mapping</div><=
div><br></div><div style>As can readily be seen, there are a number of prio=
rity inversions there. =A0Effectively, RFC 4594 doesn&#39;t work as designe=
d at the edge of the network, and we have to go back to RFC 1349, or at lea=
st a restricted set of RFC 4594 codepoints.</div>
<div style><br></div><div style>Effectively, Linux has given us a priority =
series that reads: AFx2&gt;AFx3&gt;AFx1&gt;(CS1 or BE)</div><div style><br>=
</div><div style>So, for the purpose of working with both Linux routers and=
 RFC 4594, the useful series of codepoints are AF42&gt;AF33&gt;AF21&gt;(CS1=
 or BE), and it becomes an exercise of deciding where to place those codepo=
ints in the table. =A0I&#39;d rather use CS1 than BE because it at least st=
ates that the application is doing QoS marking.</div>
<div><br></div><div style>Aside from fixing the actual mappings, though, th=
e draft is a great idea and really should be published as soon as reasonabl=
y possible.</div></div></div><div class=3D"gmail_extra"><br><br><div class=
=3D"gmail_quote">
On Tue, Jul 9, 2013 at 6:03 PM, Gorry Fairhurst <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:gorry@erg.abdn.ac.uk" target=3D"_blank">gorry@erg.abdn.ac.uk</=
a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">On 09/07/2013 00:35, Cullen Jenning=
s wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
The draft-dhesikan-tsvwg-rtcweb-<u></u>qos draft was discussed extensively =
when it was an individual draft then WG draft in another WG and I don&#39;t=
 think there is much more to say about it. This is one of the normative dep=
endencies of the webrtc work and that stuff is already getting implemented =
in browsers so I want to move this along in an expedient fashion.<br>

<br>
I think it is ready for WGLC. If there are any issues people think need to =
be discussed before WGLC, please let us know so we can resolve them in Berl=
in.<br>
<br>
Thanks, Cullen<br>
<br>
</blockquote>
<br></div></div>
In the email above you ask for WGLC. The first step is for the group to con=
sider adoption. As I recall, this draft was brought to TSVWG for considerat=
ion because it discusses transport mappings to QoS.<br>
<br>
I understand from the draft this proposes default mappings for an RTCWeb us=
e cases. As I see it, the present draft:<br>
<br>
* Advocates use of separate QoS code points for different streaming compone=
nts<br>
<br>
* Describes both endpoint handling of priority for the media and transport =
priority/precedence (aka DSCP).<br>
<br>
* Proposes formalising a mapping between specific DSCPs and subnetwork QoS =
functions (LTE, WiFi - but not general Ethernet).<br>
<br>
Please read this draft and comment on the list - we will table this for dis=
cussion at the next IETF meeting.<span class=3D"HOEnZb"><font color=3D"#888=
888"><br>
<br>
Gorry<br>
<br>
<br>
</font></span></blockquote></div><br></div>

--089e0158b7947faabf04e1110f4b--

From turners@ieca.com  Tue Jul  9 04:06:04 2013
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From: Sean Turner <turners@ieca.com>
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Subject: [tsvwg] wg review/adoption of draft-turner-rsvp-auth-update
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Hi,

I'd like to solicit additional review of this draft:
http://datatracker.ietf.org/doc/draft-turner-rsvp-auth-update/

We got some initial comments from David Black that we incorporated in 
the -01 version.  The issue we're most worried about is whether using 
unused reserved bits will cause problems.

I've also asked for a couple of minutes on the tsvwg's agenda to discuss 
this draft.

Cheers,

spt

From gorry@erg.abdn.ac.uk  Tue Jul  9 06:31:30 2013
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Subject: [tsvwg] Preliminary TSVWG Agenda
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Here is a link to the draft agenda - this is preliminary, we have still to
schedule some talks and we may yet adjust this agenda.

http://tools.ietf.org/wg/tsvwg/agenda?item=agenda-87-tsvwg.html

Let us know of corrections, conflicts or omissions.

Gorry, David and James
(TSVWG Co-Chairs)


From eckelcu@cisco.com  Tue Jul  9 14:05:13 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] Preliminary TSVWG Agenda
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Hi Chairs,

Here are links to the two missing drafts identified in the draft agenda.

http://tools.ietf.org/html/draft-choukir-tsvwg-flow-metadata-encoding-00
http://tools.ietf.org/html/draft-zamfir-tsvwg-flow-metadata-rsvp-00

I will forward the corresponding draft announcements as well.
We appreciate any comments on the drafts and hope to have some time to pres=
ent them at IETF 87.=20
The overall framework of which these drafts are a part is provided by:
http://tools.ietf.org/html/draft-eckert-intarea-flow-metadata-framework-00

Cheers,
Charles

> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> gorry@erg.abdn.ac.uk
> Sent: Tuesday, July 09, 2013 6:31 AM
> To: tsvwg@ietf.org
> Subject: [tsvwg] Preliminary TSVWG Agenda
>=20
> Here is a link to the draft agenda - this is preliminary, we have still t=
o
> schedule some talks and we may yet adjust this agenda.
>=20
> http://tools.ietf.org/wg/tsvwg/agenda?item=3Dagenda-87-tsvwg.html
>=20
> Let us know of corrections, conflicts or omissions.
>=20
> Gorry, David and James
> (TSVWG Co-Chairs)


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Subject: [tsvwg] FW: New Version Notification for draft-choukir-tsvwg-flow-metadata-encoding-00.txt
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From wes@mti-systems.com  Tue Jul  9 19:56:54 2013
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Date: Tue, 09 Jul 2013 22:56:36 -0400
From: Wesley Eddy <wes@mti-systems.com>
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Cc: DAN DRUTA <dd5826@att.com>, tsvwg@ietf.org
Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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On 7/8/2013 7:35 PM, Cullen Jennings wrote:
> 
> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when it was an individual draft then WG draft in another WG and I don't think there is much more to say about it. This is one of the normative dependencies of the webrtc work and that stuff is already getting implemented in browsers so I want to move this along in an expedient fashion. 


Without checking ... it makes no sense to me that there would be
a normative dependency on this since it does not impact the
interoperability or correctness of an implementation.

It's purely a performance enhancement, if/when it actually causes
the intended things to happen in the network.

That said, I think the interesting facet of this document is that
packets within the same flow (defined by 5-tuple of address-port
pairs and protocol number) are receiving different codepoints, and
the implications of that for a CC that may be on top of them need
to be delved into.  There isn't really text or recommendations
going into that, which seems to be the important part, as if you
can figure that out, these mappings are the more obvious part.

So, I think it might make sense to do this, but the current
document isn't addressing the implications or how this works in
a bigger system sense, and I don't think the particular marking
tables are the hard or important part.


-- 
Wes Eddy
MTI Systems

From gorry@erg.abdn.ac.uk  Wed Jul 10 00:11:35 2013
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Date: Wed, 10 Jul 2013 08:11:28 +0100
From: gorry@erg.abdn.ac.uk
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Cc: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] Preliminary TSVWG Agenda
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I updated the reading list on the Agenda page.

We'll issue a revised Agenda with the final allocations.

Gorry

> Hi Chairs,
>
> Here are links to the two missing drafts identified in the draft agenda.
>
> http://tools.ietf.org/html/draft-choukir-tsvwg-flow-metadata-encoding-00
> http://tools.ietf.org/html/draft-zamfir-tsvwg-flow-metadata-rsvp-00
>
> I will forward the corresponding draft announcements as well.
> We appreciate any comments on the drafts and hope to have some time to
> present them at IETF 87.
> The overall framework of which these drafts are a part is provided by:
> http://tools.ietf.org/html/draft-eckert-intarea-flow-metadata-framework-00
>
> Cheers,
> Charles
>
>> -----Original Message-----
>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf
>> Of
>> gorry@erg.abdn.ac.uk
>> Sent: Tuesday, July 09, 2013 6:31 AM
>> To: tsvwg@ietf.org
>> Subject: [tsvwg] Preliminary TSVWG Agenda
>>
>> Here is a link to the draft agenda - this is preliminary, we have still
>> to
>> schedule some talks and we may yet adjust this agenda.
>>
>> http://tools.ietf.org/wg/tsvwg/agenda?item=agenda-87-tsvwg.html
>>
>> Let us know of corrections, conflicts or omissions.
>>
>> Gorry, David and James
>> (TSVWG Co-Chairs)
>



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Hi,

Looks good.
I have an unrelated question since a mention occurs in the
sprstat_abandoned_sent stats):
does it make sense to abandon a message that is partially sent? At the
receiver, I am
assuming if such a  partial message was to make it, partial delivery will
be made to
the socket user?


cheers,
jamal

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<div dir=3D"ltr"><br><div>Hi,</div><div><br></div><div>Looks good.=A0</div>=
<div>I have an unrelated question since a mention occurs in the sprstat_aba=
ndoned_sent stats):=A0</div><div>does it make sense to abandon a message th=
at is partially sent? At the receiver, I am</div>

<div>assuming if such a =A0partial message was to make it, partial delivery=
 will be made to</div><div>the socket user?</div><div><br></div><div><br></=
div><div>cheers,</div><div>jamal</div><div><br></div><div><br></div></div>


--bcaec51a7c7ad39e5a04e11e1dc5--

From tuexen@fh-muenster.de  Wed Jul 10 00:42:27 2013
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On Jul 10, 2013, at 3:22 AM, Jamal Hadi Salim <hadi@mojatatu.com> wrote:

>=20
> Hi,
>=20
> Looks good.=20
> I have an unrelated question since a mention occurs in the =
sprstat_abandoned_sent stats):=20
> does it make sense to abandon a message that is partially sent? At the =
receiver, I am
> assuming if such a  partial message was to make it, partial delivery =
will be made to
> the socket user?
Ji Jamal,

you have to distinguish between the sender and receiver side.

Assume the sender sends a user message which is approximately of size =
3*MTU.
So SCTP will fragment it into 3 DATA chunk and send them. If the user
has specified that the number of retransmissions is limited to 0 and
the last fragment gets lost in the network, the sender will increment
sprstat_abandoned_sent.

What will happen at the receiver side if you assume that the first two
fragments have been received. If partial delivery was started, you will
get a SCTP_PARTIAL_DELIVERY_EVENT event indicating =
SCTP_PARTIAL_DELIVERY_ABORTED.
Without partial delivery being started, the receiver could also drop the
first part of the message.

Best regards
Michael
>=20
>=20
> cheers,
> jamal
>=20
>=20


From Michael.Tuexen@lurchi.franken.de  Wed Jul 10 03:44:04 2013
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On Jul 10, 2013, at 11:48 AM, Jamal Hadi Salim <hadi@mojatatu.com> =
wrote:

>=20
> Hi Michael,
>=20
> On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen =
<tuexen@fh-muenster.de> wrote:
>=20
> Assume the sender sends a user message which is approximately of size =
3*MTU.
> So SCTP will fragment it into 3 DATA chunk and send them. If the user
> has specified that the number of retransmissions is limited to 0 and
> the last fragment gets lost in the network, the sender will increment
> sprstat_abandoned_sent.
>=20
> Ok,  so the last fragment essentially doesnt get acknowledged before =
we=20
> abandon ship. And the other stat indicates we abandoned while we still =
had some
> sitting locally. Is the intersection of the two (some are sent but not =
acked and
> some are still sitting locally) implying both stats will be =
incremented?
I think there is no intersection:
Either a fragment has been sent or no fragment has been sent. Depending
on this, the corresponding counter has incremented.
>=20
> Other than the perceived ambiguity -  why do i care to know about the
> two types of abandoned messages in a user app? I  will have to sum
> those two, so why not just give me one stat?
Differentiating between unsent and sent makes sense. For unsent you
know for sure that the message has not been received by the peer.
For sent messages, you don't know (for example, the SACK could have
been lost).
I'm not giving you one stat because:
* if you want only one, adding them is not too hard.
* the SCTP_SEND_FAILED_EVENT event gives you also an indication
  whether the user message was sent or not. This stats is something
  like a summary.
> =20
> BTW: the doc will benefit from such a small description as you gave me
> above.
OK. I'll add some text.
>=20
> What will happen at the receiver side if you assume that the first two
> fragments have been received. If partial delivery was started, you =
will
> get a SCTP_PARTIAL_DELIVERY_EVENT event indicating =
SCTP_PARTIAL_DELIVERY_ABORTED.
> Without partial delivery being started, the receiver could also drop =
the
> first part of the message.
>=20
>=20
> You answered the essence of my question.
> Silly follow up: I am assuming that event will come with an EOR?
Events should have the MSG_EOR flag set if the buffer provided was
large enough.

Best regards
Michael
> To use your example (at receiver):
> EPOLLIN, recvmsg chunk1, chunk2, EAGAIN
> EPOLLIN, recvmsg SCTP_PARTIAL_DELIVERY_EVENT, EOR
>=20
> cheers,
> jamal


From eckert@cisco.com  Wed Jul 10 11:02:34 2013
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Date: Wed, 10 Jul 2013 11:01:47 -0700
From: "Toerless Eckert (eckert)" <eckert@cisco.com>
To: tsvwg@ietf.org, rtcweb@ietf.org, rmcat@ietf.org
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Subject: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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When dealing with congestion for real-time datagram (UDP/RTP) based video flows, one idea
that we think has shown to provide good value is to intelligently drop in the network
preferentially the least important packets within a video flow, for example non-referenced
P-frames, because their loss creates the lowest impact to video quality.

One of the main concerns raised against such schemes was one of fairness. If flows 
would start to indicate different priorities for packets in them, how would we
be able to avoid that flows would indicate all their packets to be of high priority
to shift packet drops to other flows. Or even more importantly: how could we motivate
flows to indicate that some of their packets are low priority without them having to
fear that they would experience more loss than than flows that don't do this.

So, there is one draft that we would like to present (presumably in TSVWG)  explaining
how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.txt

(Actually, this is how we did solve the problem in our implementatin of intelligent dropping,
 so this is not theoretical).

Now granted, the problem of fairness is but one element of building a complete ssytem
around the idea of intelligent dropping, so if folks here on the lists would like to
see for example 

  a) how well can it work in routers to shift packet drops to low-prio packets
  b) whats the evidence that it's good for video quality
  c) How should we do the marking best going forward

Then please let us know. We'd be very interested to provide insight into the parts of
those pieces that we have also worked out or discuss the ones we have not (primarily c ;-)

Thanks!
    Toerless

In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>

On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> Hi all,
> 
> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> 
> Best regards,
> CJ
> 
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
> Sent: Friday, February 08, 2013 6:25 PM
> To: Cheng-Jia Lai (chelai)
> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> 
> 
> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> has been successfully submitted by Cheng-Jia Lai and posted to the
> IETF repository.
> 
> Filename:	 draft-lai-tsvwg-normalizer
> Revision:	 00
> Title:		 Normalization Marker for AF PHB Group in DiffServ
> Creation date:	 2013-02-08
> WG ID:		 Individual Submission
> Number of pages: 15
> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> 
> 
> Abstract:
>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>    useful for traffic conditioning with Active Queue Management (AQM)
>    when the AF PHB group is deployed for multimedia service classes that
>    carry video packets with advanced codec technologies.  Using NM can
>    offer an incentive that is however missing in today's ecosystem of
>    practical deployment, i.e., when congestion occurs in the AF PHB
>    group, the network should allow a video codec to benefit from its
>    effort of generating finer layers of intra-flow precedence (IFP) with
>    discardable packets in a more balanced distribution.
> 
>                                                                                   
> 
> 
> The IETF Secretariat
> 

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


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Date: Wed, 10 Jul 2013 12:02:26 -0700
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From: Dave Taht <dave.taht@gmail.com>
To: "Toerless Eckert (eckert)" <eckert@cisco.com>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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On Wed, Jul 10, 2013 at 11:01 AM, Toerless Eckert (eckert)
<eckert@cisco.com> wrote:
> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.

I've begun to look at webrtc of late in the context of fq_codel.

One thing that might work would be ECN marking important frames and
not ECN marking
others. Problem overall is that ecn can be gamed and is largely not
enabled, and so on...

> One of the main concerns raised against such schemes was one of fairness.=
 If flows
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.

same gaming problem...

> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
>  so this is not theoretical).

In the case of the fq_codel algorithm I can see a "delayed drop"
mechanism happening,
where upon deciding to drop, it could defer a drop up to a few packets
based on the classification
of the packet, say 3 packets at most. Since flows are already
identified, this is kind of easy
by adding another state variable...

At the moment what I have deployed and tested most is a simple 3 tier
fq_codel based shaper,
which has a priority, best effort, and background queue. The priority
queue is generally only DNS traffic,
the background is only CS1, and fq_codel does such a good job with
sparse streams in
the general case that for VOIP, etc on the networks I fiddle with,
there is no point in having
much more than the BE queue. And presently all other markings are ignored.

video is another story. (or could be, I don't know enough about
webrtc's behaviors to
"see" what will happen)

So I can see trying to utilize the AF markings as you describe in your
draft in codel's
drop decision. While this is also subject to being gamed, the
disincentives are slight.

This is a little different from using "drop probability" as the
decider, however, but I
would hope that this mapping of the old AF style random drop ideas
into the new style delay
ideas could work...

And what codel depends on is a definition of "TCP friendly". That if a
stream is not
going to behave in a tcp friendly manner, then an entirely different
strategy for such
flows seems necessary, and glomming on some additional state isn't going to
be the right thing.

> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example
>
>   a) how well can it work in routers to shift packet drops to low-prio pa=
ckets

I guess where I fall apart here is how to shift the encoders to send thinne=
r
streams at  any level of packet drop.

>   b) whats the evidence that it's good for video quality
>   c) How should we do the marking best going forward

> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)

I'd be interested in specific codecs, example video streams, and test
scripts to try
and fiddle with this, as well as viable metrics.

>
> Thanks!
>     Toerless
>
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>
>> Best regards,
>> CJ
>>
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>
>>
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>
>> Filename:      draft-lai-tsvwg-normalizer
>> Revision:      00
>> Title:                 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:         2013-02-08
>> WG ID:                 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>
>>
>> Abstract:
>>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>    useful for traffic conditioning with Active Queue Management (AQM)
>>    when the AF PHB group is deployed for multimedia service classes that
>>    carry video packets with advanced codec technologies.  Using NM can
>>    offer an incentive that is however missing in today's ecosystem of
>>    practical deployment, i.e., when congestion occurs in the AF PHB
>>    group, the network should allow a video codec to benefit from its
>>    effort of generating finer layers of intra-flow precedence (IFP) with
>>    discardable packets in a more balanced distribution.
>>
>>
>>
>>
>> The IETF Secretariat
>>
>
> --
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From chelai@cisco.com  Wed Jul 10 16:43:00 2013
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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: Dave Taht <dave.taht@gmail.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Date: Wed, 10 Jul 2013 23:42:52 +0000
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References: <20130710180147.GA614@cisco.com> <CAA93jw68E5DgtzvE5N=2-QnGzZQzYQkevFwqWmiSGxiBZk0h4Q@mail.gmail.com>
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Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Yes, NM could also work with ECN (and it should be made so). In fact, we di=
d implementation using WRED on routers that support ECN markings. If RTP en=
dpoints support ECN markings, WRED will mark ECN instead of dropping the pa=
ckets as avg Q size is between min & max threshold. But it still suffers fr=
om unfairness at congestion, where the avg Q size starts to exceed the max =
threshold for the lowest priority packet class...

We did experience a hard time trying to properly configure WRED parameters =
for preferential dropping of video packets with our NM implementation... It=
 is a good insight you said here: maybe it is worth exploring how our work =
can perhaps use other AQM algorithms like CoDel, PIE, etc.

In fact, NM doesn't need specific AQM to drop by probabilities... it will b=
e interesting to see how it can work with delay-based dropping. However, NM=
 does require some "weighted" version of those AQMs. Hopefully a weighted v=
ersion (if any) won't introduce new parameters (or too many) to configure t=
he AQM.

Regards,
CJ


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of D=
ave Taht
Sent: Wednesday, July 10, 2013 12:02 PM
To: Toerless Eckert (eckert)
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

On Wed, Jul 10, 2013 at 11:01 AM, Toerless Eckert (eckert)
<eckert@cisco.com> wrote:
> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.

I've begun to look at webrtc of late in the context of fq_codel.

One thing that might work would be ECN marking important frames and
not ECN marking
others. Problem overall is that ecn can be gamed and is largely not
enabled, and so on...

> One of the main concerns raised against such schemes was one of fairness.=
 If flows
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.

same gaming problem...

> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
>  so this is not theoretical).

In the case of the fq_codel algorithm I can see a "delayed drop"
mechanism happening,
where upon deciding to drop, it could defer a drop up to a few packets
based on the classification
of the packet, say 3 packets at most. Since flows are already
identified, this is kind of easy
by adding another state variable...

At the moment what I have deployed and tested most is a simple 3 tier
fq_codel based shaper,
which has a priority, best effort, and background queue. The priority
queue is generally only DNS traffic,
the background is only CS1, and fq_codel does such a good job with
sparse streams in
the general case that for VOIP, etc on the networks I fiddle with,
there is no point in having
much more than the BE queue. And presently all other markings are ignored.

video is another story. (or could be, I don't know enough about
webrtc's behaviors to
"see" what will happen)

So I can see trying to utilize the AF markings as you describe in your
draft in codel's
drop decision. While this is also subject to being gamed, the
disincentives are slight.

This is a little different from using "drop probability" as the
decider, however, but I
would hope that this mapping of the old AF style random drop ideas
into the new style delay
ideas could work...

And what codel depends on is a definition of "TCP friendly". That if a
stream is not
going to behave in a tcp friendly manner, then an entirely different
strategy for such
flows seems necessary, and glomming on some additional state isn't going to
be the right thing.

> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example
>
>   a) how well can it work in routers to shift packet drops to low-prio pa=
ckets

I guess where I fall apart here is how to shift the encoders to send thinne=
r
streams at  any level of packet drop.

>   b) whats the evidence that it's good for video quality
>   c) How should we do the marking best going forward

> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)

I'd be interested in specific codecs, example video streams, and test
scripts to try
and fiddle with this, as well as viable metrics.

>
> Thanks!
>     Toerless
>
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>
>> Best regards,
>> CJ
>>
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>
>>
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>
>> Filename:      draft-lai-tsvwg-normalizer
>> Revision:      00
>> Title:                 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:         2013-02-08
>> WG ID:                 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>
>>
>> Abstract:
>>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>    nodes to normalize the distribution of DS codepoint (DSCP) markings
>>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>    useful for traffic conditioning with Active Queue Management (AQM)
>>    when the AF PHB group is deployed for multimedia service classes that
>>    carry video packets with advanced codec technologies.  Using NM can
>>    offer an incentive that is however missing in today's ecosystem of
>>    practical deployment, i.e., when congestion occurs in the AF PHB
>>    group, the network should allow a video codec to benefit from its
>>    effort of generating finer layers of intra-flow precedence (IFP) with
>>    discardable packets in a more balanced distribution.
>>
>>
>>
>>
>> The IETF Secretariat
>>
>
> --
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From eckert@cisco.com  Wed Jul 10 18:00:36 2013
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Date: Wed, 10 Jul 2013 18:00:17 -0700
From: Toerless Eckert <eckert@cisco.com>
To: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Message-ID: <20130711010017.GA20860@cisco.com>
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Subject: Re: [tsvwg] Some comments on draft-shepherd-multicast-udp-guidelines-01.txt
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Not wanting to speak for Greg, but i think the reason for the
document is to summarize that IP multicast traffic can pefectly well comply with
Internet congestion control requirements, and that there are a bunch of documents
that explain how.

I have no strong opinion on the process, so 5405 could be normative or informational,
i guess personally speaking i'd certainly hope that its clear that ALC based
rate control is in general better than sender-rate control for Internet multicast,
so everything worked out by RMT and referred to by Gregs doc would in general
superceed the unicast centric 5405. That's how i would read those documents and
resolve conflicts between them.

Cheers
    Toerless

On Mon, Jul 08, 2013 at 10:04:53AM +0100, Gorry Fairhurst wrote:
> 
> Greg,
> 
> I like the idea of publishing something on the topic in this draft, and 
> think this a valuable addition to what is said in RFC 5405. It's not 
> that multicast is hard - it's just slightly different.
> 
> However I do see the need for additional/different congestion control 
> requirements for multicast.  Was that the intention, and that the basic 
> guidance for using UDP in RFC 5405 also applies?  - I'd have expected 
> that RFC 5405 is normatively cited as a list of basic UDP requirements?
> 
> This email was as a TSVWG contributor, I'm interested in multicast 
> transport/infrastructure (not as a WG Chair).
> 
> Gorry
> 
> 
> ---
> 
> Introduction:
> I suspect any advice also applies to use of UDP-Lite with multicast?
> 
> ---
> 
> At the end of section 2.1 I see the assertion here that multicast 
> tunnels should not be congestion-controlled, making this the 
> responsibility of applications that use the tunnels.
> 
> While I can see this has been the case with traditional tunnels, the key 
> question that needs to be thought about here (I think raised by Magnus?) 
> is whether this is right also for automatic multicast tunnels - or 
> whether a circuit-breaker approach, as discussed for example with PW3E, 
> is more appropriate.
> 
> I'm not sure if the IETF should simply ignore protocols designed for 
> "private" well-managed networks. For unicast, such traffic now 
> frequently traverses VPNs and tunnels via the general Internet. Good 
> advice is important, although we may of course be unable to place 
> "requirements".
> 
> As above for both unicast and multicast, I think the issue here is that 
> if the "private" traffic is not CC'ed then the tunnels between private 
> networks do not use CC. This then means that we need to think about how 
> to protect the unicast CC traffic from this impact. Is this a network 
> operational responsibility (e.g. alarms and rate limits) or a transport 
> responsibility (e.g. circuit-breakers and CC)?
> 
> As I recall, the aggregate multicast tunnel traffic may be argued to 
> result in a significant proportion of the internet traffic through a 
> home gateway. This question therefore seems relevant to various drafts 
> including: AMT and draft-ietf-pwe3-p2mp-pw-requirements-02.txt !!!
> 
> ---
> 
> The multicast congestion control space faces similar challenges to the 
> multicast reliability space.
> 
> I think the solution space is slightly bigger than the present draft 
> says. To me, the key issue with Multicast in the General Internet is 
> that there are (at least) three fundamental differences:
> 
> * The multicast tree is often not heterogeneous - this implies there may 
> be a wide range of path characteristics. This has implications on the 
> methods used for RMT e.g. in 3.1 of RFC 5405, it says many applications 
> "cannot maintain a reliable RTT estimate for a destination". This often 
> applies to the set of destinations in an RMT session or a multicast RTP 
> session.
> 
> * The heterogeneity also means that some receivers may be receiving 
> significantly more loss than the rest of the group. In talks on 
> multicast I've called these "crying babies" (by which I mean recipients 
> that can gain the attention of the sender and may need special care). It 
> seems to me that many multicast apps try to exclude these from the group 
> - e.g. start a unicast session to this particular endpoint, or simply 
> don't distribute a security key to these receivers.
> 
> * Any receiver *anywhere* can potentially try to receive any group. 
> That's not a transport problem. It may imply the need for rate limits on 
> the aggregate multicast service, but there is no way at the transport 
> layer to stop a join message propagating to the next hop router. It also 
> points to the need for multicast operators to monitor their 
> infrastructure (which we do, and I'm sure others do - since specific 
> multicast transport issues can only be debugged when you understand the 
> health of the underlying network infrastructure).
> 
> While all this may initially seem problematic, the overall goal of 
> multicast is for users to get useful data. This implies there is an 
> incentive for apps to not to do silly things that will render their 
> service useless.
> 
> ---
> 
> Effective MTU: I saw Lars' comment on path MTU discovery (relating to 
> 3.2 of RFC 5405) - somehow this needs to be combined with the 1200 byte 
> rule of thumb that many of us have been using for multicast for years.
> 
> I recall some multicast deployments shying away from using PMTUD because 
> it was hard to protect the infrastructure when there was no specific 
> destination endpoint. Hence this was a DoS vulnerability. (Just because 
> a receiver can see a packet from a multicast tree doesn't mean it should 
> be allowed to drive-down the sender MTU to a trivial value). Could the 
> draft explain whether this is an issue or non-issue?
> 
> ---
> 
> Security
> 
> See above on the responsibility for sharing the limited capacity 
> available in a typical home gateway.
> 
> One of the great things about multicast is RPF checks - a great way to 
> protect from some DOS attacks. RPF checks in multicast REALLY help to 
> make this manageable!
> 
> IPsec and other security mechanisms work with multicast - but they need 
> multicast key management techniques.
> 
> Maybe the draft should say something about firewalls - pointing them to 
> support multicast!
> 
> ---
> 
> NITS
> 
> I suspect you do not need to cite this? 
> [I-D.narten-iana-considerations-rfc2434bis].
> 

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


From jsaldana@unizar.es  Thu Jul 11 02:19:24 2013
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Cc: tcmtf@ietf.org, tsv-area@ietf.org, tsvwg@ietf.org
Subject: [tsvwg] I-D Action: draft-saldana-tsvwg-tcmtf-05.txt
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A new version of I-D, draft-saldana-tsvwg-tcmtf-05.txt has been successfully
submitted by Jose Saldana and posted to the IETF repository.

Filename:	 draft-saldana-tsvwg-tcmtf
Revision:	 05
Title:		 Tunneling Compressed Multiplexed Traffic Flows (TCM-TF)
Creation date:	 2013-07-11
Group:		 Individual Submission
Number of pages: 21
URL:
http://www.ietf.org/internet-drafts/draft-saldana-tsvwg-tcmtf-05.txt
Status:          http://datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf
Htmlized:        http://tools.ietf.org/html/draft-saldana-tsvwg-tcmtf-05
Diff:
http://www.ietf.org/rfcdiff?url2=draft-saldana-tsvwg-tcmtf-05

Abstract:
   Tunneling Compressed and Multiplexed Traffic Flows (TCM-TF) is a
   method for improving the bandwidth utilization of network segments
   that carry multiple flows in parallel sharing a common path.  The
   method combines standard protocols for header compression,
   multiplexing, and tunneling over a network path for the purpose of
   reducing the bandwidth used when multiple flows are carried over that
   path.  The amount of packets per second can also be reduced.

   This document describes the TCM-TF framework and the different
   options which can be used for each layer (header compression,
   multiplexing and tunneling).

 



The IETF Secretariat




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Subject: Re: [tsvwg] [Editorial Errata Reported] RFC3168 (3636)
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Hi all,

On 06/04/2013 09:00 PM, Black, David wrote:
> For more context, here's the entire paragraph that contains the sentence
> in question:
>
> 6.1.1  TCP Initialization
>
>     In the TCP connection setup phase, the source and destination TCPs
>     exchange information about their willingness to use ECN.  Subsequent
>     to the completion of this negotiation, the TCP sender sets an ECT
>     codepoint in the IP header of data packets to indicate to the network
>     that the transport is capable and willing to participate in ECN for
>     this packet. This indicates to the routers that they may mark this
>     packet with the CE codepoint, if they would like to use that as a
>     method of congestion notification. If the TCP connection does not
>     wish to use ECN notification for a particular packet, the sending TCP
>     sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
>     CE codepoint in the received packet.
>
> That's not stating that "an ECN receiver can choose for every received
> CE to ignore it" - rather (at least as I read the text), it's stating that
> for a small specific fraction of the packets ("particular") the sender and
> receiver somehow know or have agreed in advance that ECN is not to be used,
> and such packets are sent as not-ECT with CE ignored on receipt.
>
> This is all rather hypothetical, as I agree with Mirja's underlying sense
> that once ECN is enabled on a TCP connection, it SHOULD be used with every
> packet.  All in all, it may be better to pursue some sort of technical
> that as opposed to trying to clean up this text editorially.
>
> It is dangerous for receivers to ignore CE markings if there is any
> possibility that the CE marking was applied as a congestion indication,
> and the original sentence can be read as allowing that dangerous behavior.
> The simplest way out of this may be to re-classify the errata (which
> removes the text that refers to that receiver behavior) as technical,
> and proceed from there.

I will reject the errata no. 3636.

Either Mirja or David, or someone else, should submit a new errata. 
Process-wise it is the better way forward.

Thanks,

   Martin

>
> Thanks,
> --David
>
>> -----Original Message-----
>> From: Mirja Kuehlewind [mailto:mirja.kuehlewind@ikr.uni-stuttgart.de]
>> Sent: Tuesday, June 04, 2013 10:22 AM
>> To: Black, David
>> Cc: RFC Errata System; kk@teraoptic.com; floyd@aciri.org;
>> spencerdawkins.ietf@gmail.com; martin.stiemerling@neclab.eu;
>> gorry@erg.abdn.ac.uk; jmpolk@cisco.com; tsvwg@ietf.org
>> Subject: Re: [Editorial Errata Reported] RFC3168 (3636)
>>
>> That means the sentence should say that an ECN receiver can choose for every
>> received CE to ignore it? What's the sense behind this? (Maybe this is a
>> technical erratum then, as e.g. ECN Nonce was designed to prevent exactly
>> this behavior.)
>>
>> If not, the sentence is still not very clear. Maybe splitting it up into two
>> sentences would help:
>>
>> Original Text
>> -------------
>> If the TCP connection does not
>>     wish to use ECN notification for a particular packet, the sending TCP
>>     sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
>>     CE codepoint in the received packet.
>>
>> Corrected Text
>> --------------
>> If the sending TCP does not
>>     wish to use ECN notification for a particular packet, it
>>     sets the ECN codepoint to not-ECT. If the TCP receiver independently does
>>     not wish to use ECN notification for a particular packet, it ignores a CE
>>     codepoint in the received packet.
>>
>>
>> Mirja
>>
>>
>> On Tuesday 04 June 2013 15:49:48 Black, David wrote:
>>> While the basic facts are correct, specifically that the receiver cannot
>>> determine whether CE has been erroneously set on a not-ECT-marked packet,
>>> I don't think this errata should be applied.
>>>
>>> The original sentence was intended to refer to independent actions by
>>> the sender and receiver, and was not supposed to imply that the receiver
>>> would ignore a received CE marking as a consequence of a not-ECT being
>>> applied by the sender.
>>>
>>> The following editorial clarification would improve the text (add
>>> "endpoint", and insert the word "independently"):
>>>
>>> Original Text
>>> -------------
>>> If the TCP connection does not
>>>     wish to use ECN notification for a particular packet, the sending TCP
>>>     sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
>>>     CE codepoint in the received packet.
>>>
>>> Corrected Text
>>> --------------
>>> If either endpoint of the TCP connection does not
>>>     wish to use ECN notification for a particular packet, the sending TCP
>>>     sets the ECN codepoint to not-ECT, and the TCP receiver independently
>>>     ignores a CE codepoint in the received packet.
>>>
>>> Thanks,
>>> --David
>>>
>>>> -----Original Message-----
>>>> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
>>>> Sent: Tuesday, June 04, 2013 9:30 AM
>>>> To: kk@teraoptic.com; floyd@aciri.org; Black, David;
>>>> spencerdawkins.ietf@gmail.com; martin.stiemerling@neclab.eu;
>>>> gorry@erg.abdn.ac.uk; Black, David; jmpolk@cisco.com
>>>> Cc: mirja.kuehlewind@ikr.uni-stuttgart.de; tsvwg@ietf.org;
>>>> rfc-editor@rfc- editor.org
>>>> Subject: [Editorial Errata Reported] RFC3168 (3636)
>>>>
>>>> The following errata report has been submitted for RFC3168,
>>>> "The Addition of Explicit Congestion Notification (ECN) to IP".
>>>>
>>>> --------------------------------------
>>>> You may review the report below and at:
>>>> http://www.rfc-editor.org/errata_search.php?rfc=3168&eid=3636
>>>>
>>>> --------------------------------------
>>>> Type: Editorial
>>>> Reported by: Mirja Kühlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
>>>>
>>>> Section: 6.1.1.
>>>>
>>>> Original Text
>>>> -------------
>>>> If the TCP connection does not
>>>>     wish to use ECN notification for a particular packet, the sending TCP
>>>>     sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
>>>>     CE codepoint in the received packet.
>>>>
>>>> Corrected Text
>>>> --------------
>>>> If the TCP connection does not
>>>>     wish to use ECN notification for a particular packet, the sending TCP
>>>>     sets the ECN codepoint to not-ECT.
>>>>
>>>> Notes
>>>> -----
>>>> CE should not be set on not-ECT capable packets. If this happens anyway,
>>>> the CE codepoint would overwrite the ECT codepoint. Thus there is no way
>>>> for the receiver to know it should ignore the CE codepoint; the sentence
>>>> is therefore nonsensical.
>>>>
>>>> Instructions:
>>>> -------------
>>>> This errata is currently posted as "Reported". If necessary, please
>>>> use "Reply All" to discuss whether it should be verified or
>>>> rejected. When a decision is reached, the verifying party (IESG)
>>>> can log in to change the status and edit the report, if necessary.
>>>>
>>>> --------------------------------------
>>>> RFC3168 (draft-ietf-tsvwg-ecn-04)
>>>> --------------------------------------
>>>> Title               : The Addition of Explicit Congestion Notification
>>>> (ECN) to IP
>>>> Publication Date    : September 2001
>>>> Author(s)           : K. Ramakrishnan, S. Floyd, D. Black
>>>> Category            : PROPOSED STANDARD
>>>> Source              : Transport Area Working Group
>>>> Area                : Transport
>>>> Stream              : IETF
>>>> Verifying Party     : IESG
>>
>>
>>
>> --
>> -------------------------------------------------------------------
>> Dipl.-Ing. Mirja Kühlewind
>> Institute of Communication Networks and Computer Engineering (IKR)
>> University of Stuttgart, Germany
>> Pfaffenwaldring 47, D-70569 Stuttgart
>>
>> tel: +49(0)711/685-67973
>> email: mirja.kuehlewind@ikr.uni-stuttgart.de
>> web: www.ikr.uni-stuttgart.de
>> -------------------------------------------------------------------
>

-- 
martin.stiemerling@neclab.eu

NEC Laboratories Europe
NEC Europe Limited
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Athene, Odyssey Business Park, West End  Road, London, HA4 6QE, GB
Registered in England 2832014

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From: ken carlberg <carlberg@g11.org.uk>
In-Reply-To: <20130710180147.GA614@cisco.com>
Date: Thu, 11 Jul 2013 12:04:58 -0400
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To: Toerless Eckert (eckert) <eckert@cisco.com>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

some food for thought...

about 10 years ago, I put together a draft that added an optional TLV =
prioritization header to RTP to accomplish something similar in terms of =
having the app decide which was the better RTP packet to drop during =
times of congestion.  My aim was to have these decisions made at =
forwarding application-layer gateways that did not act as a RTP/RTCP =
endpoint.  The strength of the approach (in my opinion :-) was that it =
could span pockets of non-diff-serv regions and still carry significance =
downstream.  The weakness was the need for DPI if one wanted to make a =
more informed decision along the path about dropping a packet.

I wonder if such a dual-layer design might be helpful in your efforts.  =
In any case, I have an interest in the normalizer draft.

-ken


On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> =
wrote:

> When dealing with congestion for real-time datagram (UDP/RTP) based =
video flows, one idea
> that we think has shown to provide good value is to intelligently drop =
in the network
> preferentially the least important packets within a video flow, for =
example non-referenced
> P-frames, because their loss creates the lowest impact to video =
quality.
>=20
> One of the main concerns raised against such schemes was one of =
fairness. If flows=20
> would start to indicate different priorities for packets in them, how =
would we
> be able to avoid that flows would indicate all their packets to be of =
high priority
> to shift packet drops to other flows. Or even more importantly: how =
could we motivate
> flows to indicate that some of their packets are low priority without =
them having to
> fear that they would experience more loss than than flows that don't =
do this.
>=20
> So, there is one draft that we would like to present (presumably in =
TSVWG)  explaining
> how we think this problem can be solved: =
draft-lai-tsvwg-normalizer-00.txt
>=20
> (Actually, this is how we did solve the problem in our implementatin =
of intelligent dropping,
> so this is not theoretical).
>=20
> Now granted, the problem of fairness is but one element of building a =
complete ssytem
> around the idea of intelligent dropping, so if folks here on the lists =
would like to
> see for example=20
>=20
>  a) how well can it work in routers to shift packet drops to low-prio =
packets
>  b) whats the evidence that it's good for video quality
>  c) How should we do the marking best going forward
>=20
> Then please let us know. We'd be very interested to provide insight =
into the parts of
> those pieces that we have also worked out or discuss the ones we have =
not (primarily c ;-)
>=20
> Thanks!
>    Toerless
>=20
> In-Reply-To: =
<A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
>=20
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) =
wrote:
>> Hi all,
>>=20
>> We have submitted a new Internet draft to describe a Normalization =
Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The =
goal of NM deployment is to create a new incentive for video encoders to =
generate more discardable packets when using AF PHB in DIffServ. We are =
looking forward to your review comments.
>>=20
>> Best regards,
>> CJ
>>=20
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert =
(eckert); Fred Yip (fyip)
>> Subject: New Version Notification for =
draft-lai-tsvwg-normalizer-00.txt
>>=20
>>=20
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-lai-tsvwg-normalizer
>> Revision:	 00
>> Title:		 Normalization Marker for AF PHB Group in =
DiffServ
>> Creation date:	 2013-02-08
>> WG ID:		 Individual Submission
>> Number of pages: 15
>> URL:             =
http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
>> Status:          =
http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
>> Htmlized:        =
http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
>>=20
>>=20
>> Abstract:
>>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>   nodes to normalize the distribution of DS codepoint (DSCP) markings
>>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>   useful for traffic conditioning with Active Queue Management (AQM)
>>   when the AF PHB group is deployed for multimedia service classes =
that
>>   carry video packets with advanced codec technologies.  Using NM can
>>   offer an incentive that is however missing in today's ecosystem of
>>   practical deployment, i.e., when congestion occurs in the AF PHB
>>   group, the network should allow a video codec to benefit from its
>>   effort of generating finer layers of intra-flow precedence (IFP) =
with
>>   discardable packets in a more balanced distribution.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


From chelai@cisco.com  Thu Jul 11 15:12:53 2013
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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: ken carlberg <carlberg@g11.org.uk>, "Toerless Eckert (eckert)" <eckert@cisco.com>
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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References: <20130710180147.GA614@cisco.com> <34BE8BE0-E863-4F69-8D45-18F5B97DE392@g11.org.uk>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality	under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Ken,

Thanks very much for the comment and showing an interest in our draft.

Yes, I agree it is beneficial if the intra-flow priorities info can be carr=
ied in some bit field other than DSCP markings. I guess we need to better c=
larify that NM is not necessarily specific to AF PHB but potentially applic=
able to other service classes. (In fact, our implementation did try DPI in =
RTP and H.264 NAL unit header for that, with some limited overheads in pack=
et processing.)

We know there can be a lot of intelligent ways to carrying priorities in pa=
ckets (e.g. your TLV in RTP hdr) which are likely beyond our knowledge and =
expertise, so we just introduced the CB mode in Section 3.1, quoted as foll=
ows:

"When NM operates in "color-blind" (CB) mode, which is optionally
   supported, it reads certain bits field(s) other than the AF-markings
   in the packets to determine the actual drop precedence of that
   packet.  This implies that NM may need DPI in the packets, e.g.
   parsing into H.264 AVC header in each RTP packets, or alternatively
   use some method where the drop precedence is carried from the encoder
   in a customized bits field other than the AF-marking in each packet."

CB might not be the best name... sorry if confusing. Anyway, we hope this a=
llows NM to collaborate better with video endpoints.

Regards,
CJ


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of k=
en carlberg
Sent: Thursday, July 11, 2013 9:05 AM
To: Toerless Eckert (eckert)
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

Toerless,

some food for thought...

about 10 years ago, I put together a draft that added an optional TLV prior=
itization header to RTP to accomplish something similar in terms of having =
the app decide which was the better RTP packet to drop during times of cong=
estion.  My aim was to have these decisions made at forwarding application-=
layer gateways that did not act as a RTP/RTCP endpoint.  The strength of th=
e approach (in my opinion :-) was that it could span pockets of non-diff-se=
rv regions and still carry significance downstream.  The weakness was the n=
eed for DPI if one wanted to make a more informed decision along the path a=
bout dropping a packet.

I wonder if such a dual-layer design might be helpful in your efforts.  In =
any case, I have an interest in the normalizer draft.

-ken


On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> wr=
ote:

> When dealing with congestion for real-time datagram (UDP/RTP) based video=
 flows, one idea
> that we think has shown to provide good value is to intelligently drop in=
 the network
> preferentially the least important packets within a video flow, for examp=
le non-referenced
> P-frames, because their loss creates the lowest impact to video quality.
>=20
> One of the main concerns raised against such schemes was one of fairness.=
 If flows=20
> would start to indicate different priorities for packets in them, how wou=
ld we
> be able to avoid that flows would indicate all their packets to be of hig=
h priority
> to shift packet drops to other flows. Or even more importantly: how could=
 we motivate
> flows to indicate that some of their packets are low priority without the=
m having to
> fear that they would experience more loss than than flows that don't do t=
his.
>=20
> So, there is one draft that we would like to present (presumably in TSVWG=
)  explaining
> how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.tx=
t
>=20
> (Actually, this is how we did solve the problem in our implementatin of i=
ntelligent dropping,
> so this is not theoretical).
>=20
> Now granted, the problem of fairness is but one element of building a com=
plete ssytem
> around the idea of intelligent dropping, so if folks here on the lists wo=
uld like to
> see for example=20
>=20
>  a) how well can it work in routers to shift packet drops to low-prio pac=
kets
>  b) whats the evidence that it's good for video quality
>  c) How should we do the marking best going forward
>=20
> Then please let us know. We'd be very interested to provide insight into =
the parts of
> those pieces that we have also worked out or discuss the ones we have not=
 (primarily c ;-)
>=20
> Thanks!
>    Toerless
>=20
> In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.=
com>
>=20
> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
>> Hi all,
>>=20
>> We have submitted a new Internet draft to describe a Normalization Marke=
r (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of =
NM deployment is to create a new incentive for video encoders to generate m=
ore discardable packets when using AF PHB in DIffServ. We are looking forwa=
rd to your review comments.
>>=20
>> Best regards,
>> CJ
>>=20
>> -----Original Message-----
>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>> Sent: Friday, February 08, 2013 6:25 PM
>> To: Cheng-Jia Lai (chelai)
>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert=
); Fred Yip (fyip)
>> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
>>=20
>>=20
>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>> has been successfully submitted by Cheng-Jia Lai and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-lai-tsvwg-normalizer
>> Revision:	 00
>> Title:		 Normalization Marker for AF PHB Group in DiffServ
>> Creation date:	 2013-02-08
>> WG ID:		 Individual Submission
>> Number of pages: 15
>> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-nor=
malizer-00.txt
>> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normali=
zer
>> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-0=
0
>>=20
>>=20
>> Abstract:
>>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>   nodes to normalize the distribution of DS codepoint (DSCP) markings
>>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
>>   useful for traffic conditioning with Active Queue Management (AQM)
>>   when the AF PHB group is deployed for multimedia service classes that
>>   carry video packets with advanced codec technologies.  Using NM can
>>   offer an incentive that is however missing in today's ecosystem of
>>   practical deployment, i.e., when congestion occurs in the AF PHB
>>   group, the network should allow a video codec to benefit from its
>>   effort of generating finer layers of intra-flow precedence (IFP) with
>>   discardable packets in a more balanced distribution.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


From eckert@cisco.com  Thu Jul 11 17:30:11 2013
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Date: Thu, 11 Jul 2013 17:30:02 -0700
From: Toerless Eckert <eckert@cisco.com>
To: ken carlberg <carlberg@g11.org.uk>
Message-ID: <20130712003002.GA30036@cisco.com>
References: <20130710180147.GA614@cisco.com> <34BE8BE0-E863-4F69-8D45-18F5B97DE392@g11.org.uk>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Thanks, Ken

Indeed yes, that's exactly the line of thought that we would like to
see proceed as well. Any URL still existing for that draft ?

"application layer gateway" could specifically mean switching MCU using
intelligent dropping in conjunction with outbound congestion control.

I definitely would like to see RTP headers as the ultimatele marking
option even for routers to do intelligent dropping (or other packet
processing "beyond DSCP)). Intra-flow (per-packet) DSCP is just a big pain 
(aka: impossible) for applications. Many OSs don't have any APIs to support
this. And DSCP management in networks is already difficult enough as it
stands.

Would certainly like to see if there is interest to write up r present
more pieces of this puzzle, especially an RTP extension. We just felt that the
fairness in intelligent dropping was the biggest concern our video
application folks had, that's why we wanted to explain how that piece
could work first. 

Cheers
    Toerless


On Thu, Jul 11, 2013 at 12:04:58PM -0400, ken carlberg wrote:
> Toerless,
> 
> some food for thought...
> 
> about 10 years ago, I put together a draft that added an optional TLV prioritization header to RTP to accomplish something similar in terms of having the app decide which was the better RTP packet to drop during times of congestion.  My aim was to have these decisions made at forwarding application-layer gateways that did not act as a RTP/RTCP endpoint.  The strength of the approach (in my opinion :-) was that it could span pockets of non-diff-serv regions and still carry significance downstream.  The weakness was the need for DPI if one wanted to make a more informed decision along the path about dropping a packet.
> 
> I wonder if such a dual-layer design might be helpful in your efforts.  In any case, I have an interest in the normalizer draft.
> 
> -ken
> 
> 
> On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) <eckert@cisco.com> wrote:
> 
> > When dealing with congestion for real-time datagram (UDP/RTP) based video flows, one idea
> > that we think has shown to provide good value is to intelligently drop in the network
> > preferentially the least important packets within a video flow, for example non-referenced
> > P-frames, because their loss creates the lowest impact to video quality.
> > 
> > One of the main concerns raised against such schemes was one of fairness. If flows 
> > would start to indicate different priorities for packets in them, how would we
> > be able to avoid that flows would indicate all their packets to be of high priority
> > to shift packet drops to other flows. Or even more importantly: how could we motivate
> > flows to indicate that some of their packets are low priority without them having to
> > fear that they would experience more loss than than flows that don't do this.
> > 
> > So, there is one draft that we would like to present (presumably in TSVWG)  explaining
> > how we think this problem can be solved: draft-lai-tsvwg-normalizer-00.txt
> > 
> > (Actually, this is how we did solve the problem in our implementatin of intelligent dropping,
> > so this is not theoretical).
> > 
> > Now granted, the problem of fairness is but one element of building a complete ssytem
> > around the idea of intelligent dropping, so if folks here on the lists would like to
> > see for example 
> > 
> >  a) how well can it work in routers to shift packet drops to low-prio packets
> >  b) whats the evidence that it's good for video quality
> >  c) How should we do the marking best going forward
> > 
> > Then please let us know. We'd be very interested to provide insight into the parts of
> > those pieces that we have also worked out or discuss the ones we have not (primarily c ;-)
> > 
> > Thanks!
> >    Toerless
> > 
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
> > 
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >> 
> >> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> >> 
> >> Best regards,
> >> CJ
> >> 
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> >> 
> >> 
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >> 
> >> Filename:	 draft-lai-tsvwg-normalizer
> >> Revision:	 00
> >> Title:		 Normalization Marker for AF PHB Group in DiffServ
> >> Creation date:	 2013-02-08
> >> WG ID:		 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> >> 
> >> 
> >> Abstract:
> >>   This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>   nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>   for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>   useful for traffic conditioning with Active Queue Management (AQM)
> >>   when the AF PHB group is deployed for multimedia service classes that
> >>   carry video packets with advanced codec technologies.  Using NM can
> >>   offer an incentive that is however missing in today's ecosystem of
> >>   practical deployment, i.e., when congestion occurs in the AF PHB
> >>   group, the network should allow a video codec to benefit from its
> >>   effort of generating finer layers of intra-flow precedence (IFP) with
> >>   discardable packets in a more balanced distribution.
> >> 
> >> 
> >> 
> >> 
> >> The IETF Secretariat
> >> 
> > 
> > -- 
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> > 
> 

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


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Date: Thu, 11 Jul 2013 17:59:30 -0700
From: "Toerless Eckert (eckert)" <eckert@cisco.com>
To: Dave Taht <dave.taht@gmail.com>
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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On Wed, Jul 10, 2013 at 12:02:26PM -0700, Dave Taht wrote:
> I've begun to look at webrtc of late in the context of fq_codel.
> 
> One thing that might work would be ECN marking important frames and
> not ECN marking others.

True. But i think there are at least 3 layers of priority that would
really be great to have: really important packet (long term reference
frames), "normal" packets, and low-proirity-packets (eg: discardable P).
And ECN could only do two.

Most of the time when you would want intelligent dropping, your drop
rate should be fairly low, eg< 20% instantaneous within the queue. If
you can do discardable P-frames it's of course best to drop those first.
Only when you instantaneously have really big bursts would you want to
drop "normal" packets and refrain to protection really important packets.

This does work really well with different queue-length (or WRED profiles).
If you only had two priorities (via ECN) it's hard to say how you
would best assign these two priorities to at least those three priority
levels.

> Problem overall is that ecn can be gamed and is largely not
> enabled, and so on...

Well. The gaming could be inhibited exactly by this draft because it
would try to guarantee the same amount of drops on a per-flow or "class"
basis, so if you're trying to game the system, you would be put into
your own class and would still see the same amount of drops as other flows.

Well, the main issue is really to redesignate ECN for a different
semantic than what it has today. And indicating packet priority is just
not quite the same as congestion feedback.

> In the case of the fq_codel algorithm I can see a "delayed drop"
> mechanism happening,
> where upon deciding to drop, it could defer a drop up to a few packets
> based on the classification
> of the packet, say 3 packets at most. Since flows are already
> identified, this is kind of easy
> by adding another state variable...

Yes, this is all quite interesting. We also played around with droppping
during enqueuing vs. during dequeuing, so there are a bunch of things
that could optimize intelligent dropping if you can build a more
intelligent queue.

[...]

> 
> So I can see trying to utilize the AF markings as you describe in your
> draft in codel's
> drop decision. While this is also subject to being gamed, the
> disincentives are slight.

Lets separate the marking from the mechanism to inhibit gaming. This
draft really is about the latter, and the marking described with AF is
really just an example. Yes, it would be lovely to agree on markings
too.

> And what codel depends on is a definition of "TCP friendly". That if a
> stream is not going to behave in a tcp friendly manner, then an entirely
> different  strategy for such flows seems necessary, and glomming on some
> additional state isn't going to be the right thing.

Who stood up at RMCAT some time back and claimed to have worked for
> 10 years on TCP and declared TCP not self friendly to itself, so RMCAT
shouldn't bother ? ;-))

Kidding aside: Intelligent dropping IMHO makes perfect sense when being
applied to RT traffic sharing a queue friendly with TCP. And intelligent
dropping can then still improve the performance.

I am pretty positive though that RMCAT traffic in general will behave
better and get lower delay and more video-appropriate loss if it can have
its own queue. Primarily because it will just take decates to root out
all the bad TCP traffic. 

I don't think that the strategies if you do or if you don't share the queue
with TCP would be fundamentally be different. 

> a) how well can it work in routers to shift packet drops to low-prio packets
> 
> I guess where I fall apart here is how to shift the encoders to send thinner
> streams at  any level of packet drop.

Video codec coders are known to be able to do crazy cool things.
There are a bunch of known mechanisms and probably more proprietary ones.

> I'd be interested in specific codecs, example video streams, and test
> scripts to try and fiddle with this, as well as viable metrics.

Sure. Best done in person. If we're going to be at the WG for this, we can
talk on the side how we've been doing this. 

Cheers
    Toerless

> > Thanks!
> >     Toerless
> >
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
> >
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >>
> >> We have submitted a new Internet draft to describe a Normalization Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal of NM deployment is to create a new incentive for video encoders to generate more discardable packets when using AF PHB in DIffServ. We are looking forward to your review comments.
> >>
> >> Best regards,
> >> CJ
> >>
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (eckert); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.txt
> >>
> >>
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >>
> >> Filename:      draft-lai-tsvwg-normalizer
> >> Revision:      00
> >> Title:                 Normalization Marker for AF PHB Group in DiffServ
> >> Creation date:         2013-02-08
> >> WG ID:                 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
> >>
> >>
> >> Abstract:
> >>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>    nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>    useful for traffic conditioning with Active Queue Management (AQM)
> >>    when the AF PHB group is deployed for multimedia service classes that
> >>    carry video packets with advanced codec technologies.  Using NM can
> >>    offer an incentive that is however missing in today's ecosystem of
> >>    practical deployment, i.e., when congestion occurs in the AF PHB
> >>    group, the network should allow a video codec to benefit from its
> >>    effort of generating finer layers of intra-flow precedence (IFP) with
> >>    discardable packets in a more balanced distribution.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
> >>
> >
> > --
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> >
> 
> 
> 
> -- 
> Dave Täht
> 
> Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.html

-- 
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


From wwwrun@rfc-editor.org  Fri Jul 12 04:17:36 2013
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Subject: [tsvwg] [Technical Errata Reported] RFC3168 (3680)
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The following errata report has been submitted for RFC3168,
"The Addition of Explicit Congestion Notification (ECN) to IP".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3168&eid=3680

--------------------------------------
Type: Technical
Reported by: Mirja Kühlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>

Section: 6.1.1.

Original Text
-------------
If the TCP connection does not
wish to use ECN notification for a particular packet, the sending TCP
sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
CE codepoint in the received packet.

Corrected Text
--------------
If the TCP connection does not
wish to use ECN notification for a particular packet, the sending TCP
sets the ECN codepoint to not-ECT.

Notes
-----
The receiver should not ignore any CE codepoint.

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3168 (draft-ietf-tsvwg-ecn-04)
--------------------------------------
Title               : The Addition of Explicit Congestion Notification (ECN) to IP
Publication Date    : September 2001
Author(s)           : K. Ramakrishnan, S. Floyd, D. Black
Category            : PROPOSED STANDARD
Source              : Transport Area Working Group
Area                : Transport
Stream              : IETF
Verifying Party     : IESG

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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos
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Subject: [tsvwg] FW: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos
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(Re)sending to tsvwg@ietf.org ... instead of tsvwg@tools.ietf.org ... Micha=
el Ramalho

-----Original Message-----
From: Michael Ramalho (mramalho)=20
Sent: Friday, July 12, 2013 9:47 AM
To: 'gorry@erg.abdn.ac.uk'; rmcat@ietf.org
Cc: tsvwg@tools.ietf.org
Subject: RE: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-=
qos

Gorry,

Your recollection regarding the desire for a separate DSCP code point for "=
RMCAT flows" is correct.

To review for everyone else here, when a RMCAT flow competes in a bottlenec=
k queue where the dominant traffic flows into that queue possess congestion=
 control that adapts based on loss (e.g., long-lived TCP flows), the RMCAT =
flow is stuck with a delay of whatever the maximum queuing delay through th=
e queue is. This situation is not desirable because the interactive applica=
tions that desire RMCAT transport desire low delay (i.e., to minimize any p=
otential bufferbloat).

So, while it is a given that a RMCAT flow must live (survive) when it finds=
 itself in such a traffic aggregate, the RMCAT flow would "prefer" to exist=
 in a traffic aggregate that does not require queue overflow when the other=
 flows adapt. This is desirable because a lower delay is possible relative =
to the former.

It is for this reason that RMCAT desires a separate DSCP code point. And so=
me RMCAT participants voiced that desire at the Orlando IETF transport work=
ing group meeting.

Draft draft-dhesikan-tsvwg-rtcweb-qos is a draft whose intension was to map=
 the EXISTING DSCPs into something RTCWeb could use today (draft authors ma=
y correct me if I am wrong). Given the recent nature of the RMCAT request (=
for a separate DSCP codepoint) to the transport WG - that request has not b=
een incorporated into the Dhesikan draft yet.

I have offered to craft a few sentences for inclusion into the Dhesikan dra=
ft explicitly stating the desire for a new codepoint is recognized, but tha=
t the draft only deals with the EXISTING codepoints (i.e., those defined by=
 RFC 4594).

I think this is the correct approach, as I think it is wise to provide RTCW=
eb applications designers with some guidance prior to the completion of RMC=
AT work. Comments are welcome.

My $0.02,

Michael Ramalho

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of g=
orry@erg.abdn.ac.uk
Sent: Tuesday, July 09, 2013 4:18 AM
To: rmcat@ietf.org
Cc: tsvwg@tools.ietf.org
Subject: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos

I's like to solicit feedback on draft-dhesikan-tsvwg-rtcweb-qos, proposed a=
s a work item in tsvwg.
This was brought to TSVWG for consideration because it discusses transport =
mappings to QoS intended for RTCweb.

At a previous RMCAT WG meeting it was mentioned that this advocates use of =
separate QoS code points for different streaming components.  At that time =
it was suggested that use of several QoS code points for different streamin=
g components would result in flow-isolation between the different RTCWeb su=
bflows and that this could potentially conflict with design goals in RMCAT =
that seek to use coupled congestion state between subflows. It would be hel=
pful to know if this draft does impact this and what issues need to be cons=
idered.

Please *do* provide feedback to tsvwg on this topic. We plan to discuss thi=
s in Berlin.

Gorry
(TSVWG Co-Chair)




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From: Jamal Hadi Salim <hadi@mojatatu.com>
Date: Wed, 10 Jul 2013 05:48:37 -0400
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Subject: Re: [tsvwg] comments draft-tuexen-tsvwg-sctp-prpolicies-01
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--047d7b6d9de0b4ae4004e1252fa4
Content-Type: text/plain; charset=ISO-8859-1

Hi Michael,

On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen <tuexen@fh-muenster.de>wrote:

>
> Assume the sender sends a user message which is approximately of size
> 3*MTU.
> So SCTP will fragment it into 3 DATA chunk and send them. If the user
> has specified that the number of retransmissions is limited to 0 and
> the last fragment gets lost in the network, the sender will increment
> sprstat_abandoned_sent.
>

Ok,  so the last fragment essentially doesnt get acknowledged before we
abandon ship. And the other stat indicates we abandoned while we still had
some
sitting locally. Is the intersection of the two (some are sent but not
acked and
some are still sitting locally) implying both stats will be incremented?

Other than the perceived ambiguity -  why do i care to know about the
two types of abandoned messages in a user app? I  will have to sum
those two, so why not just give me one stat?

BTW: the doc will benefit from such a small description as you gave me
above.

What will happen at the receiver side if you assume that the first two
> fragments have been received. If partial delivery was started, you will
> get a SCTP_PARTIAL_DELIVERY_EVENT event indicating
> SCTP_PARTIAL_DELIVERY_ABORTED.
> Without partial delivery being started, the receiver could also drop the
> first part of the message.
>
>
You answered the essence of my question.
Silly follow up: I am assuming that event will come with an EOR?
To use your example (at receiver):
EPOLLIN, recvmsg chunk1, chunk2, EAGAIN
EPOLLIN, recvmsg SCTP_PARTIAL_DELIVERY_EVENT, EOR

cheers,
jamal

--047d7b6d9de0b4ae4004e1252fa4
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><br><div class=3D"gmail_extra">Hi Michael,<br><br><div cla=
ss=3D"gmail_quote">On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen <span di=
r=3D"ltr">&lt;<a href=3D"mailto:tuexen@fh-muenster.de" target=3D"_blank">tu=
exen@fh-muenster.de</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div><div><br></div></div>
Assume the sender sends a user message which is approximately of size 3*MTU=
.<br>
So SCTP will fragment it into 3 DATA chunk and send them. If the user<br>
has specified that the number of retransmissions is limited to 0 and<br>
the last fragment gets lost in the network, the sender will increment<br>
sprstat_abandoned_sent.<br></blockquote><div><br></div><div>Ok, =A0so the l=
ast fragment essentially doesnt get acknowledged before we=A0</div><div>aba=
ndon ship. And the other stat indicates we abandoned while we still had som=
e</div>

<div>sitting locally. Is the intersection of the two (some are sent but not=
 acked and</div><div>some are still sitting locally) implying both stats wi=
ll be incremented?</div><div><br></div>
<div>Other than the perceived ambiguity - =A0why do i care to know about th=
e</div><div>two types of abandoned messages in a user app? I =A0will have t=
o sum</div><div>those two, so why not just give me one stat?</div><div>=A0<=
/div>

<div>BTW: the doc will benefit from such a small description as you gave me=
</div><div>above.</div><div><br></div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(=
204,204,204);border-left-style:solid;padding-left:1ex">


What will happen at the receiver side if you assume that the first two<br>
fragments have been received. If partial delivery was started, you will<br>
get a SCTP_PARTIAL_DELIVERY_EVENT event indicating SCTP_PARTIAL_DELIVERY_AB=
ORTED.<br>
Without partial delivery being started, the receiver could also drop the<br=
>
first part of the message.<br>
<br></blockquote><div><br></div><div>You answered the essence of my questio=
n.</div><div>Silly follow up: I am assuming that event will come with an EO=
R?</div><div>To use your example (at receiver):</div><div>EPOLLIN, recvmsg =
chunk1, chunk2, EAGAIN<br>

</div><div>EPOLLIN, recvmsg=A0SCTP_PARTIAL_DELIVERY_EVENT, EOR</div><div><b=
r></div><div>cheers,</div><div>jamal</div></div></div></div>

--047d7b6d9de0b4ae4004e1252fa4--

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From: Michael Tuexen <tuexen@fh-muenster.de>
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Subject: Re: [tsvwg] comments draft-tuexen-tsvwg-sctp-prpolicies-01
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On Jul 10, 2013, at 11:48 AM, Jamal Hadi Salim <hadi@mojatatu.com> =
wrote:

>=20
> Hi Michael,
>=20
> On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen =
<tuexen@fh-muenster.de> wrote:
>=20
> Assume the sender sends a user message which is approximately of size =
3*MTU.
> So SCTP will fragment it into 3 DATA chunk and send them. If the user
> has specified that the number of retransmissions is limited to 0 and
> the last fragment gets lost in the network, the sender will increment
> sprstat_abandoned_sent.
>=20
> Ok,  so the last fragment essentially doesnt get acknowledged before =
we=20
> abandon ship. And the other stat indicates we abandoned while we still =
had some
> sitting locally. Is the intersection of the two (some are sent but not =
acked and
> some are still sitting locally) implying both stats will be =
incremented?
>=20
> Other than the perceived ambiguity -  why do i care to know about the
> two types of abandoned messages in a user app? I  will have to sum
> those two, so why not just give me one stat?
> =20
> BTW: the doc will benefit from such a small description as you gave me
> above.
The new revision contains some clarification:
http://tools.ietf.org/html/draft-tuexen-tsvwg-sctp-prpolicies-02

Best regards
Michael
>=20
> What will happen at the receiver side if you assume that the first two
> fragments have been received. If partial delivery was started, you =
will
> get a SCTP_PARTIAL_DELIVERY_EVENT event indicating =
SCTP_PARTIAL_DELIVERY_ABORTED.
> Without partial delivery being started, the receiver could also drop =
the
> first part of the message.
>=20
>=20
> You answered the essence of my question.
> Silly follow up: I am assuming that event will come with an EOR?
> To use your example (at receiver):
> EPOLLIN, recvmsg chunk1, chunk2, EAGAIN
> EPOLLIN, recvmsg SCTP_PARTIAL_DELIVERY_EVENT, EOR
>=20
> cheers,
> jamal


From mzanaty@cisco.com  Fri Jul 12 08:19:22 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos
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Subject: Re: [tsvwg] [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos
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Correcting tsvwg mailer.

-----Original Message-----
From: Mo Zanaty (mzanaty)=20
Sent: Friday, July 12, 2013 11:16 AM
To: Michael Ramalho (mramalho); gorry@erg.abdn.ac.uk; James Polk (jmpolk)
Cc: tsvwg@tools.ietf.org; rmcat@ietf.org
Subject: RE: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-=
qos

Folks should also be aware that RFC 4594 is being updated, and that is the =
best place to propose a separate DSCP for RMCAT traffic (or delay-adaptive =
traffic in general, although you could argue RMCAT and LEDBAT should also b=
e kept separate).

http://tools.ietf.org/html/draft-polk-tsvwg-rfc4594-update-03

Mo

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of M=
ichael Ramalho (mramalho)
Sent: Friday, July 12, 2013 9:47 AM
To: gorry@erg.abdn.ac.uk; rmcat@ietf.org
Cc: tsvwg@tools.ietf.org
Subject: Re: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-=
qos

Gorry,

Your recollection regarding the desire for a separate DSCP code point for "=
RMCAT flows" is correct.

To review for everyone else here, when a RMCAT flow competes in a bottlenec=
k queue where the dominant traffic flows into that queue possess congestion=
 control that adapts based on loss (e.g., long-lived TCP flows), the RMCAT =
flow is stuck with a delay of whatever the maximum queuing delay through th=
e queue is. This situation is not desirable because the interactive applica=
tions that desire RMCAT transport desire low delay (i.e., to minimize any p=
otential bufferbloat).

So, while it is a given that a RMCAT flow must live (survive) when it finds=
 itself in such a traffic aggregate, the RMCAT flow would "prefer" to exist=
 in a traffic aggregate that does not require queue overflow when the other=
 flows adapt. This is desirable because a lower delay is possible relative =
to the former.

It is for this reason that RMCAT desires a separate DSCP code point. And so=
me RMCAT participants voiced that desire at the Orlando IETF transport work=
ing group meeting.

Draft draft-dhesikan-tsvwg-rtcweb-qos is a draft whose intension was to map=
 the EXISTING DSCPs into something RTCWeb could use today (draft authors ma=
y correct me if I am wrong). Given the recent nature of the RMCAT request (=
for a separate DSCP codepoint) to the transport WG - that request has not b=
een incorporated into the Dhesikan draft yet.

I have offered to craft a few sentences for inclusion into the Dhesikan dra=
ft explicitly stating the desire for a new codepoint is recognized, but tha=
t the draft only deals with the EXISTING codepoints (i.e., those defined by=
 RFC 4594).

I think this is the correct approach, as I think it is wise to provide RTCW=
eb applications designers with some guidance prior to the completion of RMC=
AT work. Comments are welcome.

My $0.02,

Michael Ramalho

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of g=
orry@erg.abdn.ac.uk
Sent: Tuesday, July 09, 2013 4:18 AM
To: rmcat@ietf.org
Cc: tsvwg@tools.ietf.org
Subject: [rmcat] Shared Congestion state & draft-dhesikan-tsvwg-rtcweb-qos

I's like to solicit feedback on draft-dhesikan-tsvwg-rtcweb-qos, proposed a=
s a work item in tsvwg.
This was brought to TSVWG for consideration because it discusses transport =
mappings to QoS intended for RTCweb.

At a previous RMCAT WG meeting it was mentioned that this advocates use of =
separate QoS code points for different streaming components.  At that time =
it was suggested that use of several QoS code points for different streamin=
g components would result in flow-isolation between the different RTCWeb su=
bflows and that this could potentially conflict with design goals in RMCAT =
that seek to use coupled congestion state between subflows. It would be hel=
pful to know if this draft does impact this and what issues need to be cons=
idered.

Please *do* provide feedback to tsvwg on this topic. We plan to discuss thi=
s in Berlin.

Gorry
(TSVWG Co-Chair)




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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : Generic Aggregation of Resource ReSerVation Protocol (RS=
VP) for IPv4 And IPv6 Reservations over PCN domains
	Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
	Filename        : draft-ietf-tsvwg-rsvp-pcn-05.txt
	Pages           : 27
	Date            : 2013-07-13

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-05


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From bhargava.anurag@gmail.com  Sun Jul 14 05:06:38 2013
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From: Anurag Bhargava <bhargava.anurag@gmail.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>, anuragb@cisco.com,  Georgios Karagiannis <karagian@cs.utwente.nl>
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--047d7b41c7465fe00c04e177937e
Content-Type: text/plain; charset=ISO-8859-1

Hi All,
  Here is the new draft which incorporates comments

   A) incorporated Bob's comments: For ex: Added text to be more clearer

"Even though these procedures are out of the scope

of this document, the PCN-ingress-node can refer to a central

decision point which can respond to the PCN ingress as per

[RFC2753]."



B) incorporated David's comment on misconfiguration and added Section 3.16:

"3.16. Misconfiguration of PCN-node

In an event where a PCN-node is misconfigured within a PCN-domain,

the desired behavior is same as described in Section 3.9. Therefore,

the Aggregated Resv messages are simply forwarded as normal IP

datagrams."


C) Few editorial fixes and some cleanups

Thx
-Anurag


---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Sat, Jul 13, 2013 at 12:11 PM
Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-05.txt
To: i-d-announce@ietf.org
Cc: tsvwg@ietf.org



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

        Title           : Generic Aggregation of Resource ReSerVation
Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domains
        Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
        Filename        : draft-ietf-tsvwg-rsvp-pcn-05.txt
        Pages           : 27
        Date            : 2013-07-13

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rsvp-pcn-05


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/




-- 
BR,
-Anurag

--047d7b41c7465fe00c04e177937e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi All,<div>=A0 Here is the new draft which incorporates c=
omments</div><div><br></div><div>=A0 =A0A<span class=3D"" style=3D"border-c=
ollapse:collapse;font-family:Tahoma;font-size:13px">) incorporated Bob&#39;=
s comments: For ex: Added text to be more clearer</span></div>
<span class=3D"" style=3D"border-collapse:collapse;font-family:Tahoma;font-=
size:13px"><span lang=3D"N"><p>&quot;Even though these procedures are out o=
f the scope</p><p>of this document, the PCN-ingress-node can refer to a cen=
tral</p>
<p>decision point which can respond to the PCN ingress as per</p><p>[RFC275=
3].&quot;</p><p>=A0</p></span><p>B) incorporated David&#39;s comment on mis=
configuration and added Section 3.16:</p><span lang=3D"N"><p>&quot;3.16. Mi=
sconfiguration of PCN-node</p>
<p>In an event where a PCN-node is misconfigured within a PCN-domain,</p><p=
>the desired behavior is same as described in Section 3.9. Therefore,</p><p=
>the Aggregated Resv messages are simply forwarded as normal IP</p><p>datag=
rams.&quot;</p>
</span><p>=A0=A0=A0<br>C)=A0Few editorial fixes and some cleanups</p><div><=
br></div><div>Thx</div><div>-Anurag</div></span><div><br><br><div class=3D"=
gmail_quote">---------- Forwarded message ----------<br>From: <b class=3D"g=
mail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-draf=
ts@ietf.org">internet-drafts@ietf.org</a>&gt;</span><br>
Date: Sat, Jul 13, 2013 at 12:11 PM<br>Subject: [tsvwg] I-D Action: draft-i=
etf-tsvwg-rsvp-pcn-05.txt<br>To: <a href=3D"mailto:i-d-announce@ietf.org">i=
-d-announce@ietf.org</a><br>Cc: <a href=3D"mailto:tsvwg@ietf.org">tsvwg@iet=
f.org</a><br>
<br><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=A0This draft is a work item of the Transport Area Working Group Working Gr=
oup of the IETF.<br>
<br>
=A0 =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : Generic Aggregation of Resource=
 ReSerVation Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domain=
s<br>
=A0 =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Georgios Karagiannis<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Anurag Bhargava<br>
=A0 =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-tsvwg-rsvp-pcn-05.txt<=
br>
=A0 =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 27<br>
=A0 =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-07-13<br>
<br>
Abstract:<br>
=A0 =A0This document specifies extensions to Generic Aggregated RSVP<br>
=A0 =A0[RFC4860] for support of the PCN Controlled Load (CL) and Single<br>
=A0 =A0Marking (SM) edge behaviors over a Diffserv cloud using Pre-<br>
=A0 =A0Congestion Notification.<br>
<br>
<br>
<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn</a=
><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-05"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp=
-pcn-05</a><br>
<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
</div><br><br clear=3D"all"><div><br></div>-- <br>BR,<br>-Anurag
</div></div>

--047d7b41c7465fe00c04e177937e--

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From: Anurag Bhargava <bhargava.anurag@gmail.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>, anuragb@cisco.com,  Georgios Karagiannis <karagian@cs.utwente.nl>
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-05.txt
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Please note that this resolves all the issues which were brought up in the
last WG IETF meet in Orlando last March.

Hopefully we can now aim for the next step.

Br,
-A


On Sun, Jul 14, 2013 at 8:06 AM, Anurag Bhargava
<bhargava.anurag@gmail.com>wrote:

> Hi All,
>   Here is the new draft which incorporates comments
>
>    A) incorporated Bob's comments: For ex: Added text to be more clearer
>
> "Even though these procedures are out of the scope
>
> of this document, the PCN-ingress-node can refer to a central
>
> decision point which can respond to the PCN ingress as per
>
> [RFC2753]."
>
>
>
> B) incorporated David's comment on misconfiguration and added Section 3.16:
>
> "3.16. Misconfiguration of PCN-node
>
> In an event where a PCN-node is misconfigured within a PCN-domain,
>
> the desired behavior is same as described in Section 3.9. Therefore,
>
> the Aggregated Resv messages are simply forwarded as normal IP
>
> datagrams."
>
>
> C) Few editorial fixes and some cleanups
>
> Thx
> -Anurag
>
>
> ---------- Forwarded message ----------
> From: <internet-drafts@ietf.org>
> Date: Sat, Jul 13, 2013 at 12:11 PM
> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-05.txt
> To: i-d-announce@ietf.org
> Cc: tsvwg@ietf.org
>
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Transport Area Working Group Working
> Group of the IETF.
>
>         Title           : Generic Aggregation of Resource ReSerVation
> Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domains
>         Author(s)       : Georgios Karagiannis
>                           Anurag Bhargava
>         Filename        : draft-ietf-tsvwg-rsvp-pcn-05.txt
>         Pages           : 27
>         Date            : 2013-07-13
>
> Abstract:
>    This document specifies extensions to Generic Aggregated RSVP
>    [RFC4860] for support of the PCN Controlled Load (CL) and Single
>    Marking (SM) edge behaviors over a Diffserv cloud using Pre-
>    Congestion Notification.
>
>
>
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rsvp-pcn-05
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
>
>
>
> --
> BR,
> -Anurag
>



-- 
BR,
-Anurag

--001a11c20b9437939b04e1779b39
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<div dir=3D"ltr">Please note that this resolves all the issues which were b=
rought up in the last WG IETF meet in Orlando last March.<div><br></div><di=
v>Hopefully we can now aim for the next step.</div><div><br></div><div>Br,<=
/div>
<div>-A</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_q=
uote">On Sun, Jul 14, 2013 at 8:06 AM, Anurag Bhargava <span dir=3D"ltr">&l=
t;<a href=3D"mailto:bhargava.anurag@gmail.com" target=3D"_blank">bhargava.a=
nurag@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">Hi All,<div>=A0 Here is the=
 new draft which incorporates comments</div><div><br></div><div>=A0 =A0A<sp=
an style=3D"border-collapse:collapse;font-family:Tahoma;font-size:13px">) i=
ncorporated Bob&#39;s comments: For ex: Added text to be more clearer</span=
></div>

<span style=3D"border-collapse:collapse;font-family:Tahoma;font-size:13px">=
<span lang=3D"N"><p>&quot;Even though these procedures are out of the scope=
</p><p>of this document, the PCN-ingress-node can refer to a central</p>
<p>decision point which can respond to the PCN ingress as per</p><p>[RFC275=
3].&quot;</p><p>=A0</p></span><p>B) incorporated David&#39;s comment on mis=
configuration and added Section 3.16:</p><span lang=3D"N"><p>&quot;3.16. Mi=
sconfiguration of PCN-node</p>

<p>In an event where a PCN-node is misconfigured within a PCN-domain,</p><p=
>the desired behavior is same as described in Section 3.9. Therefore,</p><p=
>the Aggregated Resv messages are simply forwarded as normal IP</p><p>
datagrams.&quot;</p>
</span><p>=A0=A0=A0<br>C)=A0Few editorial fixes and some cleanups</p><div><=
br></div><div>Thx</div><div>-Anurag</div></span><div><div><div class=3D"h5"=
><br><br><div class=3D"gmail_quote">---------- Forwarded message ----------=
<br>From: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=
=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf=
.org</a>&gt;</span><br>

Date: Sat, Jul 13, 2013 at 12:11 PM<br>Subject: [tsvwg] I-D Action: draft-i=
etf-tsvwg-rsvp-pcn-05.txt<br>To: <a href=3D"mailto:i-d-announce@ietf.org" t=
arget=3D"_blank">i-d-announce@ietf.org</a><br>Cc: <a href=3D"mailto:tsvwg@i=
etf.org" target=3D"_blank">tsvwg@ietf.org</a><br>

<br><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=A0This draft is a work item of the Transport Area Working Group Working Gr=
oup of the IETF.<br>
<br>
=A0 =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : Generic Aggregation of Resource=
 ReSerVation Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domain=
s<br>
=A0 =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Georgios Karagiannis<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Anurag Bhargava<br>
=A0 =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-tsvwg-rsvp-pcn-05.txt<=
br>
=A0 =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 27<br>
=A0 =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-07-13<br>
<br>
Abstract:<br>
=A0 =A0This document specifies extensions to Generic Aggregated RSVP<br>
=A0 =A0[RFC4860] for support of the PCN Controlled Load (CL) and Single<br>
=A0 =A0Marking (SM) edge behaviors over a Diffserv cloud using Pre-<br>
=A0 =A0Congestion Notification.<br>
<br>
<br>
<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn</a=
><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-05</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-05"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp=
-pcn-05</a><br>
<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
</div><br><br clear=3D"all"><div><br></div></div></div><span class=3D"HOEnZ=
b"><font color=3D"#888888">-- <br>BR,<br>-Anurag
</font></span></div></div>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>BR,<br>-Anur=
ag
</div>

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From: "Subha Dhesikan (sdhesika)" <sdhesika@cisco.com>
To: Andrew McGregor <andrewmcgr@gmail.com>, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Andrew,

Thank you for reviewing the draft and pointing out the issue that will aris=
e with the deployed base.

Just so I am clear, you are proposing a couple of hardships with the curren=
t draft:

1.       Smaller set of values that will work with the below-mentioned impl=
ementations

2.       The ordering be different in terms of priority from what is propos=
ed not only in this draft but the work prior that my draft is leveraging. F=
or example AF2 > AF3 and at the same time > AF1.

Defining a smaller set of values may be alright except we lose the granular=
ity and then having a mechanism to restore the real value in the network be=
comes important. My draft alludes to such a service which will take policy =
into account and impact markings if needed. If we restrict the values to a =
very small set, such a service will become critical.

The other challenge I see is with the relative position of the values being=
 different from what is traditionally known for DSCP where EF> AF4x >AF3x>.=
.. How have you handled it so far in a non-web rtc environment?

If I have not understood your comments properly, please correct.

Regards,
Subha

.

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of A=
ndrew McGregor
Sent: Tuesday, July 09, 2013 2:48 AM
To: gorry@erg.abdn.ac.uk
Cc: tsvwg
Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos

So, there is a significant bug in the mappings proposed.  That is this: the=
re is a substantial deployed base of home routers and other network devices=
 that use the Linux kernel's default output queue pfifo_fast, which uses a =
mapping derived from RFC 1349 (documented in the linux man page tc_prio, se=
e for example http://linux.die.net/man/8/tc-prio)   If the proposed mapping=
s don't work with that queue discipline, they won't work on the deployed ba=
se.  pfifo_fast has then unfortunate properties of ignoring the top 3 bits =
of the tos byte, and sometimes paying attention to one of the ECN bits in c=
hoosing a queueing band.

In effect, since Linux has produced a de-facto standard codepoint assignmen=
t for the edge of the network, we had better recommend something compatible=
.  The proposed mapping uses EF for all audio, which will create a priority=
 inversion and have all the Linux devices on the edge classify that traffic=
 as 'bulk interactive', lower than the interactive video at AF4x in the 'in=
teractive' band.  This is probably not what we want.

I've filled out the linux band numbers in table 1 from the draft; the third=
 number in each table cell is the linux band, smaller number is higher prio=
rity:
+-----------------------+-------------+-------------+-------------+
| Media Type            |    Low      |   Medium    |    High     |
+-----------------------+-------------+-------------+-------------+
| Audio                 |  46 (EF)  1 |  46 (EF)  1 |  46 (EF)  1 |
| Interactive Video     | 38 (AF43) 1 | 36 (AF42) 0 | 34 (AF41) 2 |
| Non-Interactive Video | 26 (AF33) 1 | 28 (AF32) 0 | 30 (AF31) 2 |
| Data                  |  8 (CS1)  1 |   0 (BE)  1 | 10 (AF11) 2 |
+-----------------------+-------------+-------------+-------------+
      Table 1 - Media Type, Priority, and DSCP Mapping

As can readily be seen, there are a number of priority inversions there.  E=
ffectively, RFC 4594 doesn't work as designed at the edge of the network, a=
nd we have to go back to RFC 1349, or at least a restricted set of RFC 4594=
 codepoints.

Effectively, Linux has given us a priority series that reads: AFx2>AFx3>AFx=
1>(CS1 or BE)

So, for the purpose of working with both Linux routers and RFC 4594, the us=
eful series of codepoints are AF42>AF33>AF21>(CS1 or BE), and it becomes an=
 exercise of deciding where to place those codepoints in the table.  I'd ra=
ther use CS1 than BE because it at least states that the application is doi=
ng QoS marking.

Aside from fixing the actual mappings, though, the draft is a great idea an=
d really should be published as soon as reasonably possible.

On Tue, Jul 9, 2013 at 6:03 PM, Gorry Fairhurst <gorry@erg.abdn.ac.uk<mailt=
o:gorry@erg.abdn.ac.uk>> wrote:
On 09/07/2013 00:35, Cullen Jennings wrote:

The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when it=
 was an individual draft then WG draft in another WG and I don't think ther=
e is much more to say about it. This is one of the normative dependencies o=
f the webrtc work and that stuff is already getting implemented in browsers=
 so I want to move this along in an expedient fashion.

I think it is ready for WGLC. If there are any issues people think need to =
be discussed before WGLC, please let us know so we can resolve them in Berl=
in.

Thanks, Cullen

In the email above you ask for WGLC. The first step is for the group to con=
sider adoption. As I recall, this draft was brought to TSVWG for considerat=
ion because it discusses transport mappings to QoS.

I understand from the draft this proposes default mappings for an RTCWeb us=
e cases. As I see it, the present draft:

* Advocates use of separate QoS code points for different streaming compone=
nts

* Describes both endpoint handling of priority for the media and transport =
priority/precedence (aka DSCP).

* Proposes formalising a mapping between specific DSCPs and subnetwork QoS =
functions (LTE, WiFi - but not general Ethernet).

Please read this draft and comment on the list - we will table this for dis=
cussion at the next IETF meeting.

Gorry



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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Andrew,<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thank you for reviewing t=
he draft and pointing out the issue that will arise with the deployed base.=
 &nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Just so I am clear, you a=
re proposing a couple of hardships with the current draft:<o:p></o:p></span=
></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">1.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Smaller set of va=
lues that will work with the below-mentioned implementations<o:p></o:p></sp=
an></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">2.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The ordering be d=
ifferent in terms of priority from what is proposed not only in this draft =
but the work prior that my draft is leveraging. For example
 AF2 &gt; AF3 and at the same time &gt; AF1.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Defining a smaller set of=
 values may be alright except we lose the granularity and then having a mec=
hanism to restore the real value in the network becomes
 important. My draft alludes to such a service which will take policy into =
account and impact markings if needed. If we restrict the values to a very =
small set, such a service will become critical.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">The other challenge I see=
 is with the relative position of the values being different from what is t=
raditionally known for DSCP where EF&gt; AF4x &gt;AF3x&gt;&#8230; How have
 you handled it so far in a non-web rtc environment?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">If I have not understood =
your comments properly, please correct.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Subha<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> tsvwg-bo=
unces@ietf.org [mailto:tsvwg-bounces@ietf.org]
<b>On Behalf Of </b>Andrew McGregor<br>
<b>Sent:</b> Tuesday, July 09, 2013 2:48 AM<br>
<b>To:</b> gorry@erg.abdn.ac.uk<br>
<b>Cc:</b> tsvwg<br>
<b>Subject:</b> Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">So, there is a significant bug in the mappings propo=
sed. &nbsp;That is this: there is a substantial deployed base of home route=
rs and other network devices that use the Linux kernel's default output que=
ue pfifo_fast, which uses a mapping derived
 from RFC 1349&nbsp;(documented in the linux man page tc_prio, see for exam=
ple <a href=3D"http://linux.die.net/man/8/tc-prio">
http://linux.die.net/man/8/tc-prio</a>)&nbsp; &nbsp;If the proposed mapping=
s don't work with that queue discipline, they won't work on the deployed ba=
se. &nbsp;pfifo_fast has then unfortunate properties of ignoring the top 3 =
bits of the tos byte, and sometimes paying attention
 to one of the ECN bits in choosing a queueing band.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">In effect, since Linux has produced a de-facto stand=
ard codepoint assignment for the edge of the network, we had better recomme=
nd something compatible. &nbsp;The proposed mapping uses EF for all audio, =
which will create a priority inversion
 and have all the Linux devices on the edge classify that traffic as 'bulk =
interactive', lower than the interactive video at AF4x in the 'interactive'=
 band. &nbsp;This is probably not what we want.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I've filled out the linux band numbers in table 1 fr=
om the draft; the third number in each table cell is the linux band, smalle=
r number is higher priority:<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&#43;-----------------------&#43;-------------&#43;-=
------------&#43;-------------&#43;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">| Media Type &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;| &nbsp; &nbsp;Low &nbsp; &nbsp; &nbsp;| &nbsp; Medium &nbsp; &nbsp;| &nb=
sp; &nbsp;High &nbsp; &nbsp; |<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&#43;-----------------------&#43;-------------&#43;-=
------------&#43;-------------&#43;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">| Audio &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; | &nbsp;46 (EF) &nbsp;1 | &nbsp;46 (EF) &nbsp;1 | &nbsp;46 (EF)=
 &nbsp;1 |<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">| Interactive Video &nbsp; &nbsp; | 38 (AF43) 1 | 36=
 (AF42) 0 | 34 (AF41) 2 |<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">| Non-Interactive Video | 26 (AF33) 1 | 28 (AF32) 0 =
| 30 (AF31) 2 |<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">| Data &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp;| &nbsp;8 (CS1) &nbsp;1 | &nbsp; 0 (BE) &nbsp;1 | 10 (AF11=
) 2 |<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&#43;-----------------------&#43;-------------&#43;-=
------------&#43;-------------&#43;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; &nbsp; &nbsp; Table 1 - Media Type, Priority,=
 and DSCP Mapping<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">As can readily be seen, there are a number of priori=
ty inversions there. &nbsp;Effectively, RFC 4594 doesn't work as designed a=
t the edge of the network, and we have to go back to RFC 1349, or at least =
a restricted set of RFC 4594 codepoints.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Effectively, Linux has given us a priority series th=
at reads: AFx2&gt;AFx3&gt;AFx1&gt;(CS1 or BE)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">So, for the purpose of working with both Linux route=
rs and RFC 4594, the useful series of codepoints are AF42&gt;AF33&gt;AF21&g=
t;(CS1 or BE), and it becomes an exercise of deciding where to place those =
codepoints in the table. &nbsp;I'd rather use CS1
 than BE because it at least states that the application is doing QoS marki=
ng.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Aside from fixing the actual mappings, though, the d=
raft is a great idea and really should be published as soon as reasonably p=
ossible.<o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 9, 2013 at 6:03 PM, Gorry Fairhurst &lt;=
<a href=3D"mailto:gorry@erg.abdn.ac.uk" target=3D"_blank">gorry@erg.abdn.ac=
.uk</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On 09/07/2013 00:35, Cullen Jennings wrote:<o:p></o:=
p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when it=
 was an individual draft then WG draft in another WG and I don't think ther=
e is much more to say about it. This is one of the normative dependencies o=
f the webrtc work and that stuff
 is already getting implemented in browsers so I want to move this along in=
 an expedient fashion.<br>
<br>
I think it is ready for WGLC. If there are any issues people think need to =
be discussed before WGLC, please let us know so we can resolve them in Berl=
in.<br>
<br>
Thanks, Cullen<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">In the email above yo=
u ask for WGLC. The first step is for the group to consider adoption. As I =
recall, this draft was brought to TSVWG for consideration because it discus=
ses transport mappings to QoS.<br>
<br>
I understand from the draft this proposes default mappings for an RTCWeb us=
e cases. As I see it, the present draft:<br>
<br>
* Advocates use of separate QoS code points for different streaming compone=
nts<br>
<br>
* Describes both endpoint handling of priority for the media and transport =
priority/precedence (aka DSCP).<br>
<br>
* Proposes formalising a mapping between specific DSCPs and subnetwork QoS =
functions (LTE, WiFi - but not general Ethernet).<br>
<br>
Please read this draft and comment on the list - we will table this for dis=
cussion at the next IETF meeting.<span style=3D"color:#888888"><br>
<br>
<span class=3D"hoenzb">Gorry</span><br>
<br>
</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
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From: "Subha Dhesikan (sdhesika)" <sdhesika@cisco.com>
To: Wesley Eddy <wes@mti-systems.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Cc: DAN DRUTA <dd5826@att.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Wes,

In my view,  this draft important since it does provides guidance on a coup=
le of fronts such as what the browser needs to do and what the application =
Javascript needs to do.  Also, it helps to clarify what inputs will be rece=
ived from the application and what inputs are provided by the browser to th=
e media engine. That is the key value of the draft.=20

I understand that there will be an added complexity when the flow is no lon=
ger the traditional 5 tuple but a single flow can have multiple DSCP values=
. However, it is not a concept introduced by this draft. When applications =
have started allowing  multiplexing of different types of traffic into a si=
ngle flow, the need has already been created. Not only in the webrtc space =
but also in the VDI space it is common to see flow that have multiplexed co=
ntent. While we can restrict the QoS for multiplexed flows by saying only o=
ne DSCP allowed per flow, my experience has been that it is tremendously ch=
allenging for QoS to be relevant in such deployments. =20

Having said the above, my draft does not either mandate a single or multipl=
e DSCP for multiplexed flow.  One can always negotiate the lack of support =
for multiplexed flow in which case single flows will be used with one DSCP =
per flow.

Unfortunately, I will not be in Berlin. I am hoping that Cullen, Dan and yo=
u can chat some more if this continues to be an issue.

Regards,
Subha

-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of W=
esley Eddy
Sent: Tuesday, July 09, 2013 7:57 PM
To: Cullen Jennings
Cc: DAN DRUTA; tsvwg@ietf.org
Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos

On 7/8/2013 7:35 PM, Cullen Jennings wrote:
>=20
> The draft-dhesikan-tsvwg-rtcweb-qos draft was discussed extensively when =
it was an individual draft then WG draft in another WG and I don't think th=
ere is much more to say about it. This is one of the normative dependencies=
 of the webrtc work and that stuff is already getting implemented in browse=
rs so I want to move this along in an expedient fashion.=20


Without checking ... it makes no sense to me that there would be a normativ=
e dependency on this since it does not impact the interoperability or corre=
ctness of an implementation.

It's purely a performance enhancement, if/when it actually causes the inten=
ded things to happen in the network.

That said, I think the interesting facet of this document is that packets w=
ithin the same flow (defined by 5-tuple of address-port pairs and protocol =
number) are receiving different codepoints, and the implications of that fo=
r a CC that may be on top of them need to be delved into.  There isn't real=
ly text or recommendations going into that, which seems to be the important=
 part, as if you can figure that out, these mappings are the more obvious p=
art.

So, I think it might make sense to do this, but the current document isn't =
addressing the implications or how this works in a bigger system sense, and=
 I don't think the particular marking tables are the hard or important part=
.


--
Wes Eddy
MTI Systems

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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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On 15/07/2013 06:46, Subha Dhesikan (sdhesika) wrote:
...
> The other challenge I see is with the relative position of the values being different from what is traditionally known for DSCP where EF> AF4x >AF3x>... How have you handled it so far in a non-web rtc environment?

I don't track rtcweb and have no intention of doing so, but what on earth do you
mean by writing an inequality for DSCPs? DSCPs do not specify a priority
order; they identify (locally) a PHB, which is essentially a queuing disciplime.
Suggesting that EF is "greater" than an AF behaviour is just wrong.

   Brian

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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Nevertheless, specifying a priority order is exactly what tc_prio and
pfifo_fast do... I agree it's wrong, but that's what a very large number of
existing devices do right now, and ignoring that fact is likely to cause
problems.

Yes, this is unfortunately a mess, and I can't see a way to avoid that
completely.  The smaller set of codepoints I mention, AF42>AF33>AF21>(CS1
or BE), is not comprehensive, but is at least consistently ordered; the
tc_prio map unfortunately makes EF unusable because it ends up equivalent
to BE.

To clarify a little, I'm saying that AF42, AF32, AF22 and AF12 are all the
same so far as tc_prio is concerned, same for AF41, AF31, AF21 and AF11,
and the same for AF[1234]3.  So to get some consistency between the
schemes, I have chosen for example AF42, which is in the highest priority
band for both, AF33 for the second, AF21 for the third, and CS1 for the
fourth, although even that is not ideal because it puts AF21 in a
less-than-best-effort class while CS1 is equivalent to best-effort on the
linux-based devices.

There is considerable established practice within the game industry of
using QoS codepoints that do map to tc_prio bands.  Therefore there's a
commonly deployed base of both low-end routers and code that depend on that
behaviour, and attempting to standardise something that is inconsistent
with it seems like an exercise in futility.

I should point out that not every linux-based router does this, it is
certainly possible to get 4594 behaviour, or ignore the TOS bits entirely,
or do DPI to assign TOS, or remark, or pretty much any behaviour you might
imagine.  However, the majority of linux-based devices use very close to
the default configuration of the IP stack.

I believe rtcweb should be pragmatic about this, and accept that very
fine-grained marking is not going to fly in the majority of cases on the
edge of the internet, and take the benefits available from a restricted set
of codepoints that are at least widely available, reasonably consistent in
semantics, and commonly also result in correct behaviour from lower layers,
particularly 802.1p and 802.11e MAC QoS.

Andrew


On 15 July 2013 06:08, Brian E Carpenter <brian.e.carpenter@gmail.com>wrote:

> On 15/07/2013 06:46, Subha Dhesikan (sdhesika) wrote:
> ...
> > The other challenge I see is with the relative position of the values
> being different from what is traditionally known for DSCP where EF> AF4x
> >AF3x>... How have you handled it so far in a non-web rtc environment?
>
> I don't track rtcweb and have no intention of doing so, but what on earth
> do you
> mean by writing an inequality for DSCPs? DSCPs do not specify a priority
> order; they identify (locally) a PHB, which is essentially a queuing
> disciplime.
> Suggesting that EF is "greater" than an AF behaviour is just wrong.
>
>    Brian
>



-- 
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 8143 7128

--001a11c221088c83d604e17e8abc
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div>Nevertheless, specifying a priority order is exactly =
what tc_prio and pfifo_fast do... I agree it&#39;s wrong, but that&#39;s wh=
at a very large number of existing devices do right now, and ignoring that =
fact is likely to cause problems.</div>
<div><br></div><div>Yes, this is unfortunately a mess, and I can&#39;t see =
a way to avoid that completely. =A0The smaller set of codepoints I mention,=
 AF42&gt;AF33&gt;AF21&gt;(CS1 or BE), is not comprehensive, but is at least=
 consistently ordered; the tc_prio map unfortunately makes EF unusable beca=
use it ends up equivalent to BE.</div>
<div><br></div><div>To clarify a little, I&#39;m saying that AF42, AF32, AF=
22 and AF12 are all the same so far as tc_prio is concerned, same for AF41,=
 AF31, AF21 and AF11, and the same for AF[1234]3. =A0So to get some consist=
ency between the schemes, I have chosen for example AF42, which is in the h=
ighest priority band for both, AF33 for the second, AF21 for the third, and=
 CS1 for the fourth, although even that is not ideal because it puts AF21 i=
n a less-than-best-effort class while CS1 is equivalent to best-effort on t=
he linux-based devices.</div>
<div><br></div><div>There is considerable established practice within the g=
ame industry of using QoS codepoints that do map to tc_prio bands. =A0There=
fore there&#39;s a commonly deployed base of both low-end routers and code =
that depend on that behaviour, and attempting to standardise something that=
 is inconsistent with it seems like an exercise in futility.</div>
<div><br></div><div>I should point out that not every linux-based router do=
es this, it is certainly possible to get 4594 behaviour, or ignore the TOS =
bits entirely, or do DPI to assign TOS, or remark, or pretty much any behav=
iour you might imagine. =A0However, the majority of linux-based devices use=
 very close to the default configuration of the IP stack.</div>
<div><br></div><div style>I believe rtcweb should be pragmatic about this, =
and accept that very fine-grained marking is not going to fly in the majori=
ty of cases on the edge of the internet, and take the benefits available fr=
om a restricted set of codepoints that are at least widely available, reaso=
nably consistent in semantics, and commonly also result in correct behaviou=
r from lower layers, particularly 802.1p and 802.11e MAC QoS.</div>
<div><br></div><div style>Andrew</div></div><div class=3D"gmail_extra"><br>=
<br><div class=3D"gmail_quote">On 15 July 2013 06:08, Brian E Carpenter <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:brian.e.carpenter@gmail.com" target=3D=
"_blank">brian.e.carpenter@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On 15/07/2013 06:46, Subha Dhesikan (sdhesik=
a) wrote:<br>
...<br>
&gt; The other challenge I see is with the relative position of the values =
being different from what is traditionally known for DSCP where EF&gt; AF4x=
 &gt;AF3x&gt;... How have you handled it so far in a non-web rtc environmen=
t?<br>

<br>
I don&#39;t track rtcweb and have no intention of doing so, but what on ear=
th do you<br>
mean by writing an inequality for DSCPs? DSCPs do not specify a priority<br=
>
order; they identify (locally) a PHB, which is essentially a queuing discip=
lime.<br>
Suggesting that EF is &quot;greater&quot; than an AF behaviour is just wron=
g.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
=A0 =A0Brian<br>
</font></span></blockquote></div><br><br clear=3D"all"><div><br></div>-- <b=
r><div dir=3D"ltr"><span style=3D"color:rgb(85,85,85);font-family:sans-seri=
f;font-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:s=
olid;border-color:rgb(213,15,37);padding-top:2px;margin-top:2px">Andrew McG=
regor=A0|</span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;f=
ont-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:soli=
d;border-color:rgb(51,105,232);padding-top:2px;margin-top:2px">=A0SRE=A0|</=
span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:sm=
all;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-co=
lor:rgb(0,153,57);padding-top:2px;margin-top:2px">=A0<a href=3D"mailto:andr=
ewmcgr@google.com" target=3D"_blank">andrewmcgr@google.com</a>=A0|</span><s=
pan style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small;lin=
e-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-color:rgb=
(238,178,17);padding-top:2px;margin-top:2px">=A0+61 4 8143 7128</span><br>
</div>
</div>

--001a11c221088c83d604e17e8abc--

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Date: Mon, 15 Jul 2013 08:51:39 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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On 15/07/2013 08:25, Andrew Mcgregor wrote:
> Nevertheless, specifying a priority order is exactly what tc_prio and
> pfifo_fast do... I agree it's wrong, but that's what a very large number of
> existing devices do right now, and ignoring that fact is likely to cause
> problems.
> 
> Yes, this is unfortunately a mess, and I can't see a way to avoid that
> completely.  The smaller set of codepoints I mention, AF42>AF33>AF21>(CS1
> or BE), is not comprehensive, but is at least consistently ordered; the
> tc_prio map unfortunately makes EF unusable because it ends up equivalent
> to BE.
> 
> To clarify a little, I'm saying that AF42, AF32, AF22 and AF12 are all the
> same so far as tc_prio is concerned, same for AF41, AF31, AF21 and AF11,
> and the same for AF[1234]3.  So to get some consistency between the
> schemes, I have chosen for example AF42, which is in the highest priority
> band for both, AF33 for the second, AF21 for the third, and CS1 for the
> fourth, although even that is not ideal because it puts AF21 in a
> less-than-best-effort class while CS1 is equivalent to best-effort on the
> linux-based devices.

In that case, the error is using the PHB names in the first place. The
behaviours you're describing are *not* the PHBs whose names you are using.
They need new names that describe them correctly.

To reiterate, the binding between a behaviour such as EF or AFxy and a particular
DSCP value is locally defined. If the behaviour is in fact simple preemptive
priority, please don't call it something else. The fact that Linux gets it
wrong doesn't mean IETF documents should be wrong too.

   Brian

> 
> There is considerable established practice within the game industry of
> using QoS codepoints that do map to tc_prio bands.  Therefore there's a
> commonly deployed base of both low-end routers and code that depend on that
> behaviour, and attempting to standardise something that is inconsistent
> with it seems like an exercise in futility.
> 
> I should point out that not every linux-based router does this, it is
> certainly possible to get 4594 behaviour, or ignore the TOS bits entirely,
> or do DPI to assign TOS, or remark, or pretty much any behaviour you might
> imagine.  However, the majority of linux-based devices use very close to
> the default configuration of the IP stack.
> 
> I believe rtcweb should be pragmatic about this, and accept that very
> fine-grained marking is not going to fly in the majority of cases on the
> edge of the internet, and take the benefits available from a restricted set
> of codepoints that are at least widely available, reasonably consistent in
> semantics, and commonly also result in correct behaviour from lower layers,
> particularly 802.1p and 802.11e MAC QoS.
> 
> Andrew
> 
> 
> On 15 July 2013 06:08, Brian E Carpenter <brian.e.carpenter@gmail.com>wrote:
> 
>> On 15/07/2013 06:46, Subha Dhesikan (sdhesika) wrote:
>> ...
>>> The other challenge I see is with the relative position of the values
>> being different from what is traditionally known for DSCP where EF> AF4x
>>> AF3x>... How have you handled it so far in a non-web rtc environment?
>> I don't track rtcweb and have no intention of doing so, but what on earth
>> do you
>> mean by writing an inequality for DSCPs? DSCPs do not specify a priority
>> order; they identify (locally) a PHB, which is essentially a queuing
>> disciplime.
>> Suggesting that EF is "greater" than an AF behaviour is just wrong.
>>
>>    Brian
>>
> 
> 
> 

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From: "Subha Dhesikan (sdhesika)" <sdhesika@cisco.com>
To: Brian E Carpenter <brian.e.carpenter@gmail.com>
Thread-Topic: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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From: "Subha Dhesikan (sdhesika)" <sdhesika@cisco.com>
To: Brian E Carpenter <brian.e.carpenter@gmail.com>, Andrew Mcgregor <andrewmcgr@google.com>
Thread-Topic: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Date: Mon, 15 Jul 2013 10:15:56 +1000
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From: Andrew Mcgregor <andrewmcgr@google.com>
To: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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On 15 July 2013 06:51, Brian E Carpenter <brian.e.carpenter@gmail.com>wrote:

> On 15/07/2013 08:25, Andrew Mcgregor wrote:
> > Nevertheless, specifying a priority order is exactly what tc_prio and
> > pfifo_fast do... I agree it's wrong, but that's what a very large number
> of
> > existing devices do right now, and ignoring that fact is likely to cause
> > problems.
> >
> > Yes, this is unfortunately a mess, and I can't see a way to avoid that
> > completely.  The smaller set of codepoints I mention, AF42>AF33>AF21>(CS1
> > or BE), is not comprehensive, but is at least consistently ordered; the
> > tc_prio map unfortunately makes EF unusable because it ends up equivalent
> > to BE.
> >
> > To clarify a little, I'm saying that AF42, AF32, AF22 and AF12 are all
> the
> > same so far as tc_prio is concerned, same for AF41, AF31, AF21 and AF11,
> > and the same for AF[1234]3.  So to get some consistency between the
> > schemes, I have chosen for example AF42, which is in the highest priority
> > band for both, AF33 for the second, AF21 for the third, and CS1 for the
> > fourth, although even that is not ideal because it puts AF21 in a
> > less-than-best-effort class while CS1 is equivalent to best-effort on the
> > linux-based devices.
>
> In that case, the error is using the PHB names in the first place. The
> behaviours you're describing are *not* the PHBs whose names you are using.
> They need new names that describe them correctly.
>

Well, perhaps they do.  However, using the RFC 4594 names for the bit
patterns in the packets per figure 3 of that RFC, rather than the PHB
definitions, those particular bit patterns get vaguely compatible treatment
between some router behaving per RFC 4594 PHB definitions and something
running pfifo_fast.  That is my point.

Perhaps we need another description of what is going on here, that's a
separate question, but we have those five codepoints (and perhaps a few
more, I didn't do an exhaustive search) that produce sensible behaviour in
both RFC 4594 recommended behaviour and the more widely deployed pfifo_fast.

More sensible names might well be those originally from 802.11e or 802.1p,
whichever of those coined these, those being VO ('voice'), VI
('video/interactive'), BE ('best effort') and BK ('background'), in which
case we have from my earlier example:

RFC 4594 name -> 802 name (TOS bit pattern)
AF42 -> VO (100100)
AF33 -> VI (011110)
AF21 -> BK (010010)
BE -> BE (000000)
CS1 -> BE (001000) (possibly, the different code point in this case saying
'BE, and I really mean that, this is not just default')

Using this set of bit patterns, the queueing behaviour will be correct on
many, perhaps most, existing access networks AND the 802 layer queueing
will be correct as well on a subset of those.  This would be a win, in
general.  It will also cause something sensible to happen in a network
using the mapping from RFC 4594 figure 3.

To reiterate, the binding between a behaviour such as EF or AFxy and a
> particular
> DSCP value is locally defined. If the behaviour is in fact simple
> preemptive
> priority, please don't call it something else. The fact that Linux gets it
> wrong doesn't mean IETF documents should be wrong too.
>

I believe that that local definition of the binding is a catastrophic bug
in the diffserv specs, and should never have been published in that form.
 However, too late now.  There is absolutely no way to tell an application
correctly what it should do with those bits unless there is an agreed set
of default meanings for at least some of the codepoints; maybe not total
specification of the exact behaviour, that's impractical, but at least a
way of setting some expectations.

If rtcweb continues with the code point recommendations in the draft as it
stands, code written to those recommendations will be quite seriously
broken when deployed.  The predictable outcome will be 'so turn off the
marking', and yet again an opportunity to do something sensible and
pragmatic with this feature is missed.  Let's not allow that to happen
simply because of terminology.  By saying 5 out of 64 codepoints should
have reasonable default behaviours, we might actually get this deployed.


>
>    Brian
>
> >
> > There is considerable established practice within the game industry of
> > using QoS codepoints that do map to tc_prio bands.  Therefore there's a
> > commonly deployed base of both low-end routers and code that depend on
> that
> > behaviour, and attempting to standardise something that is inconsistent
> > with it seems like an exercise in futility.
> >
> > I should point out that not every linux-based router does this, it is
> > certainly possible to get 4594 behaviour, or ignore the TOS bits
> entirely,
> > or do DPI to assign TOS, or remark, or pretty much any behaviour you
> might
> > imagine.  However, the majority of linux-based devices use very close to
> > the default configuration of the IP stack.
> >
> > I believe rtcweb should be pragmatic about this, and accept that very
> > fine-grained marking is not going to fly in the majority of cases on the
> > edge of the internet, and take the benefits available from a restricted
> set
> > of codepoints that are at least widely available, reasonably consistent
> in
> > semantics, and commonly also result in correct behaviour from lower
> layers,
> > particularly 802.1p and 802.11e MAC QoS.
> >
> > Andrew
> >
> >
> > On 15 July 2013 06:08, Brian E Carpenter <brian.e.carpenter@gmail.com
> >wrote:
> >
> >> On 15/07/2013 06:46, Subha Dhesikan (sdhesika) wrote:
> >> ...
> >>> The other challenge I see is with the relative position of the values
> >> being different from what is traditionally known for DSCP where EF> AF4x
> >>> AF3x>... How have you handled it so far in a non-web rtc environment?
> >> I don't track rtcweb and have no intention of doing so, but what on
> earth
> >> do you
> >> mean by writing an inequality for DSCPs? DSCPs do not specify a priority
> >> order; they identify (locally) a PHB, which is essentially a queuing
> >> disciplime.
> >> Suggesting that EF is "greater" than an AF behaviour is just wrong.
> >>
> >>    Brian
> >>
> >
> >
> >
>



-- 
Andrew McGregor | SRE | andrewmcgr@google.com | +61 4 8143 7128

--20cf3074d2e4aee0df04e181c356
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On 15 July 2013 06:51, Brian E Carpenter <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:brian.e.carpenter@gmail.com" target=3D"_blank">brian.e.carpen=
ter@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"im">On 15/07/2013 08:25, Andrew Mcgregor wro=
te:<br>

&gt; Nevertheless, specifying a priority order is exactly what tc_prio and<=
br>
&gt; pfifo_fast do... I agree it&#39;s wrong, but that&#39;s what a very la=
rge number of<br>
&gt; existing devices do right now, and ignoring that fact is likely to cau=
se<br>
&gt; problems.<br>
&gt;<br>
&gt; Yes, this is unfortunately a mess, and I can&#39;t see a way to avoid =
that<br>
&gt; completely. =A0The smaller set of codepoints I mention, AF42&gt;AF33&g=
t;AF21&gt;(CS1<br>
&gt; or BE), is not comprehensive, but is at least consistently ordered; th=
e<br>
&gt; tc_prio map unfortunately makes EF unusable because it ends up equival=
ent<br>
&gt; to BE.<br>
&gt;<br>
&gt; To clarify a little, I&#39;m saying that AF42, AF32, AF22 and AF12 are=
 all the<br>
&gt; same so far as tc_prio is concerned, same for AF41, AF31, AF21 and AF1=
1,<br>
&gt; and the same for AF[1234]3. =A0So to get some consistency between the<=
br>
&gt; schemes, I have chosen for example AF42, which is in the highest prior=
ity<br>
&gt; band for both, AF33 for the second, AF21 for the third, and CS1 for th=
e<br>
&gt; fourth, although even that is not ideal because it puts AF21 in a<br>
&gt; less-than-best-effort class while CS1 is equivalent to best-effort on =
the<br>
&gt; linux-based devices.<br>
<br>
</div>In that case, the error is using the PHB names in the first place. Th=
e<br>
behaviours you&#39;re describing are *not* the PHBs whose names you are usi=
ng.<br>
They need new names that describe them correctly.<br></blockquote><div><br>=
</div><div style>Well, perhaps they do. =A0However, using the RFC 4594 name=
s for the bit patterns in the packets per figure 3 of that RFC, rather than=
 the PHB definitions, those particular bit patterns get vaguely compatible =
treatment between some router behaving per RFC 4594 PHB definitions and som=
ething running pfifo_fast. =A0That is my point.</div>
<div style><br></div><div style>Perhaps we need another description of what=
 is going on here, that&#39;s a separate question, but we have those five c=
odepoints (and perhaps a few more, I didn&#39;t do an exhaustive search) th=
at produce sensible behaviour in both RFC 4594 recommended behaviour and th=
e more widely deployed pfifo_fast.</div>
<div style><br></div><div style>More sensible names might well be those ori=
ginally from 802.11e or 802.1p, whichever of those coined these, those bein=
g VO (&#39;voice&#39;), VI (&#39;video/interactive&#39;), BE (&#39;best eff=
ort&#39;) and BK (&#39;background&#39;), in which case we have from my earl=
ier example:</div>
<div style><br></div><div style>RFC 4594 name -&gt; 802 name (TOS bit patte=
rn)</div><div style>AF42 -&gt; VO (<span style=3D"color:rgb(0,0,0);font-siz=
e:1em">100100)</span></div><div style><span style=3D"color:rgb(0,0,0);font-=
size:1em">AF33 -&gt; VI (</span><span style=3D"color:rgb(0,0,0);font-size:1=
em">011110)</span></div>
<div style><span style=3D"color:rgb(0,0,0);font-size:1em">AF21 -&gt; BK (</=
span><span style=3D"color:rgb(0,0,0);font-size:1em">010010)</span></div><di=
v style><span style=3D"color:rgb(0,0,0);font-size:1em">BE -&gt; BE (000000)=
</span></div>
<div style><font color=3D"#000000">CS1 -&gt; BE (001000) (possibly, the dif=
ferent code point in this case saying &#39;BE, and I really mean that, this=
 is not just default&#39;)</font></div><div style><font color=3D"#000000"><=
br>
</font></div><div style><font color=3D"#000000">Using this set of bit patte=
rns, the queueing behaviour will be correct on many, perhaps most, existing=
 access networks AND the 802 layer queueing will be correct as well on a su=
bset of those. =A0This would be a win, in general. =A0It will also cause so=
mething sensible to happen in a network using the mapping from RFC 4594 fig=
ure 3.</font></div>
<div style><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bord=
er-left-style:solid;padding-left:1ex">
To reiterate, the binding between a behaviour such as EF or AFxy and a part=
icular<br>
DSCP value is locally defined. If the behaviour is in fact simple preemptiv=
e<br>
priority, please don&#39;t call it something else. The fact that Linux gets=
 it<br>
wrong doesn&#39;t mean IETF documents should be wrong too.<br></blockquote>=
<div><br></div><div style>I believe that that local definition of the bindi=
ng is a catastrophic bug in the diffserv specs, and should never have been =
published in that form. =A0However, too late now. =A0There is absolutely no=
 way to tell an application correctly what it should do with those bits unl=
ess there is an agreed set of default meanings for at least some of the cod=
epoints; maybe not total specification of the exact behaviour, that&#39;s i=
mpractical, but at least a way of setting some expectations.</div>
<div style><br></div><div style>If rtcweb continues with the code point rec=
ommendations in the draft as it stands, code written to those recommendatio=
ns will be quite seriously broken when deployed. =A0The predictable outcome=
 will be &#39;so turn off the marking&#39;, and yet again an opportunity to=
 do something sensible and pragmatic with this feature is missed. =A0Let&#3=
9;s not allow that to happen simply because of terminology. =A0By saying 5 =
out of 64 codepoints should have reasonable default behaviours, we might ac=
tually get this deployed.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<span class=3D""><font color=3D"#888888"><br>
=A0 =A0Brian<br>
</font></span><div class=3D""><div class=3D"h5"><br>
&gt;<br>
&gt; There is considerable established practice within the game industry of=
<br>
&gt; using QoS codepoints that do map to tc_prio bands. =A0Therefore there&=
#39;s a<br>
&gt; commonly deployed base of both low-end routers and code that depend on=
 that<br>
&gt; behaviour, and attempting to standardise something that is inconsisten=
t<br>
&gt; with it seems like an exercise in futility.<br>
&gt;<br>
&gt; I should point out that not every linux-based router does this, it is<=
br>
&gt; certainly possible to get 4594 behaviour, or ignore the TOS bits entir=
ely,<br>
&gt; or do DPI to assign TOS, or remark, or pretty much any behaviour you m=
ight<br>
&gt; imagine. =A0However, the majority of linux-based devices use very clos=
e to<br>
&gt; the default configuration of the IP stack.<br>
&gt;<br>
&gt; I believe rtcweb should be pragmatic about this, and accept that very<=
br>
&gt; fine-grained marking is not going to fly in the majority of cases on t=
he<br>
&gt; edge of the internet, and take the benefits available from a restricte=
d set<br>
&gt; of codepoints that are at least widely available, reasonably consisten=
t in<br>
&gt; semantics, and commonly also result in correct behaviour from lower la=
yers,<br>
&gt; particularly 802.1p and 802.11e MAC QoS.<br>
&gt;<br>
&gt; Andrew<br>
&gt;<br>
&gt;<br>
&gt; On 15 July 2013 06:08, Brian E Carpenter &lt;<a href=3D"mailto:brian.e=
.carpenter@gmail.com">brian.e.carpenter@gmail.com</a>&gt;wrote:<br>
&gt;<br>
&gt;&gt; On 15/07/2013 06:46, Subha Dhesikan (sdhesika) wrote:<br>
&gt;&gt; ...<br>
&gt;&gt;&gt; The other challenge I see is with the relative position of the=
 values<br>
&gt;&gt; being different from what is traditionally known for DSCP where EF=
&gt; AF4x<br>
&gt;&gt;&gt; AF3x&gt;... How have you handled it so far in a non-web rtc en=
vironment?<br>
&gt;&gt; I don&#39;t track rtcweb and have no intention of doing so, but wh=
at on earth<br>
&gt;&gt; do you<br>
&gt;&gt; mean by writing an inequality for DSCPs? DSCPs do not specify a pr=
iority<br>
&gt;&gt; order; they identify (locally) a PHB, which is essentially a queui=
ng<br>
&gt;&gt; disciplime.<br>
&gt;&gt; Suggesting that EF is &quot;greater&quot; than an AF behaviour is =
just wrong.<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0Brian<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<div dir=3D"ltr"><span style=3D"color:rgb(85,85,85);font-family:sans-serif;=
font-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:sol=
id;border-color:rgb(213,15,37);padding-top:2px;margin-top:2px">Andrew McGre=
gor=A0|</span><span style=3D"color:rgb(85,85,85);font-family:sans-serif;fon=
t-size:small;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;=
border-color:rgb(51,105,232);padding-top:2px;margin-top:2px">=A0SRE=A0|</sp=
an><span style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:smal=
l;line-height:1.5em;border-width:2px 0px 0px;border-style:solid;border-colo=
r:rgb(0,153,57);padding-top:2px;margin-top:2px">=A0<a href=3D"mailto:andrew=
mcgr@google.com" target=3D"_blank">andrewmcgr@google.com</a>=A0|</span><spa=
n style=3D"color:rgb(85,85,85);font-family:sans-serif;font-size:small;line-=
height:1.5em;border-width:2px 0px 0px;border-style:solid;border-color:rgb(2=
38,178,17);padding-top:2px;margin-top:2px">=A0+61 4 8143 7128</span><br>
</div>
</div></div>

--20cf3074d2e4aee0df04e181c356--

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Date: Mon, 15 Jul 2013 17:02:11 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] draft-dhesikan-tsvwg-rtcweb-qos
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Andrew,

Just a few more points from me:

On 15/07/2013 12:15, Andrew Mcgregor wrote:
...
> Perhaps we need another description of what is going on here, that's a
> separate question, 

Indeed we do, and it's called PHB Identifiers, which are intended for
naming PHBs in a globally unique way independent of the 6 bit DSCP
(RFC 3140).

It's a shame nobody uses them.

> but we have those five codepoints (and perhaps a few
> more, I didn't do an exhaustive search) that produce sensible behaviour in
> both RFC 4594 recommended behaviour and the more widely deployed pfifo_fast.

Fair enough, but 4594 only works across provider boundaries if there is mutual
agreement.

...
> 
> I believe that that local definition of the binding is a catastrophic bug
> in the diffserv specs, and should never have been published in that form.

It was (at the time) an essentially non-negotiable requirement from
those in the diffserv WG who claimed to be operators who wanted
freedom to use the TOS field as they wished. If today's operators have
a different view, I don't see why this couldn't be revisited, but
that's off topic here and now.

>  However, too late now.  There is absolutely no way to tell an application
> correctly what it should do with those bits unless there is an agreed set
> of default meanings for at least some of the codepoints; maybe not total
> specification of the exact behaviour, that's impractical, but at least a
> way of setting some expectations.

I agree, and this applies to any cross-domain use of the codepoints.
However, there are two caveats:

1. Use phrasing that recognises that the standard formally requires local
bindings.

2. Deal with the fact that these bits can be forged in the IP header.

  Brian


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From: "Shitanshu Shah (svshah)" <svshah@cisco.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] Preliminary TSVWG Agenda
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Hi Gorry,

An updated draft of sla exchange, based on the feed-back from the last
IETF, has been posted to both idr and tsvwg.
http://tools.ietf.org/html/draft-ietf-idr-sla-exchange-01

You may want to update the reference in reading material.

No plan to present the draft at this IETF either in idr or tsvwg.
We can take any concerned discussion on the wg e-mail alias. Looking
forward for review/comments.

Regards,
Shitanshu



On 7/9/13 6:31 AM, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk> wrote:

>Here is a link to the draft agenda - this is preliminary, we have still to
>schedule some talks and we may yet adjust this agenda.
>
>http://tools.ietf.org/wg/tsvwg/agenda?item=3Dagenda-87-tsvwg.html
>
>Let us know of corrections, conflicts or omissions.
>
>Gorry, David and James
>(TSVWG Co-Chairs)
>


From carlberg@g11.org.uk  Mon Jul 15 06:41:33 2013
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Cc: rmcat@ietf.org, rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [tsvwg] [rmcat] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

sorry for the tardy reply.  I'll dig through my archives and send you a =
copy.  my draft proposed adding a "classifier" field, but I only did one =
rev (which was missing some better rationale after a long talk I had =
with Dave Oran) and i suspect that it would need a fair amount of =
scrubbing before re-issuing another version.  the first order of =
business, though, will be to gauge the feedback/comments to your draft.

-ken

ps, CJ thanks for your clarification in your previous response


On Jul 11, 2013, at 8:30 PM, Toerless Eckert <eckert@cisco.com> wrote:

> Thanks, Ken
>=20
> Indeed yes, that's exactly the line of thought that we would like to
> see proceed as well. Any URL still existing for that draft ?
>=20
> "application layer gateway" could specifically mean switching MCU =
using
> intelligent dropping in conjunction with outbound congestion control.
>=20
> I definitely would like to see RTP headers as the ultimatele marking
> option even for routers to do intelligent dropping (or other packet
> processing "beyond DSCP)). Intra-flow (per-packet) DSCP is just a big =
pain=20
> (aka: impossible) for applications. Many OSs don't have any APIs to =
support
> this. And DSCP management in networks is already difficult enough as =
it
> stands.
>=20
> Would certainly like to see if there is interest to write up r present
> more pieces of this puzzle, especially an RTP extension. We just felt =
that the
> fairness in intelligent dropping was the biggest concern our video
> application folks had, that's why we wanted to explain how that piece
> could work first.=20
>=20
> Cheers
>    Toerless
>=20
>=20
> On Thu, Jul 11, 2013 at 12:04:58PM -0400, ken carlberg wrote:
>> Toerless,
>>=20
>> some food for thought...
>>=20
>> about 10 years ago, I put together a draft that added an optional TLV =
prioritization header to RTP to accomplish something similar in terms of =
having the app decide which was the better RTP packet to drop during =
times of congestion.  My aim was to have these decisions made at =
forwarding application-layer gateways that did not act as a RTP/RTCP =
endpoint.  The strength of the approach (in my opinion :-) was that it =
could span pockets of non-diff-serv regions and still carry significance =
downstream.  The weakness was the need for DPI if one wanted to make a =
more informed decision along the path about dropping a packet.
>>=20
>> I wonder if such a dual-layer design might be helpful in your =
efforts.  In any case, I have an interest in the normalizer draft.
>>=20
>> -ken
>>=20
>>=20
>> On Jul 10, 2013, at 2:01 PM, Toerless Eckert (eckert) =
<eckert@cisco.com> wrote:
>>=20
>>> When dealing with congestion for real-time datagram (UDP/RTP) based =
video flows, one idea
>>> that we think has shown to provide good value is to intelligently =
drop in the network
>>> preferentially the least important packets within a video flow, for =
example non-referenced
>>> P-frames, because their loss creates the lowest impact to video =
quality.
>>>=20
>>> One of the main concerns raised against such schemes was one of =
fairness. If flows=20
>>> would start to indicate different priorities for packets in them, =
how would we
>>> be able to avoid that flows would indicate all their packets to be =
of high priority
>>> to shift packet drops to other flows. Or even more importantly: how =
could we motivate
>>> flows to indicate that some of their packets are low priority =
without them having to
>>> fear that they would experience more loss than than flows that don't =
do this.
>>>=20
>>> So, there is one draft that we would like to present (presumably in =
TSVWG)  explaining
>>> how we think this problem can be solved: =
draft-lai-tsvwg-normalizer-00.txt
>>>=20
>>> (Actually, this is how we did solve the problem in our implementatin =
of intelligent dropping,
>>> so this is not theoretical).
>>>=20
>>> Now granted, the problem of fairness is but one element of building =
a complete ssytem
>>> around the idea of intelligent dropping, so if folks here on the =
lists would like to
>>> see for example=20
>>>=20
>>> a) how well can it work in routers to shift packet drops to low-prio =
packets
>>> b) whats the evidence that it's good for video quality
>>> c) How should we do the marking best going forward
>>>=20
>>> Then please let us know. We'd be very interested to provide insight =
into the parts of
>>> those pieces that we have also worked out or discuss the ones we =
have not (primarily c ;-)
>>>=20
>>> Thanks!
>>>   Toerless
>>>=20
>>> In-Reply-To: =
<A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisco.com>
>>>=20
>>> On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) =
wrote:
>>>> Hi all,
>>>>=20
>>>> We have submitted a new Internet draft to describe a Normalization =
Marker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The =
goal of NM deployment is to create a new incentive for video encoders to =
generate more discardable packets when using AF PHB in DIffServ. We are =
looking forward to your review comments.
>>>>=20
>>>> Best regards,
>>>> CJ
>>>>=20
>>>> -----Original Message-----
>>>> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
>>>> Sent: Friday, February 08, 2013 6:25 PM
>>>> To: Cheng-Jia Lai (chelai)
>>>> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert =
(eckert); Fred Yip (fyip)
>>>> Subject: New Version Notification for =
draft-lai-tsvwg-normalizer-00.txt
>>>>=20
>>>>=20
>>>> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
>>>> has been successfully submitted by Cheng-Jia Lai and posted to the
>>>> IETF repository.
>>>>=20
>>>> Filename:	 draft-lai-tsvwg-normalizer
>>>> Revision:	 00
>>>> Title:		 Normalization Marker for AF PHB Group in =
DiffServ
>>>> Creation date:	 2013-02-08
>>>> WG ID:		 Individual Submission
>>>> Number of pages: 15
>>>> URL:             =
http://www.ietf.org/internet-drafts/draft-lai-tsvwg-normalizer-00.txt
>>>> Status:          =
http://datatracker.ietf.org/doc/draft-lai-tsvwg-normalizer
>>>> Htmlized:        =
http://tools.ietf.org/html/draft-lai-tsvwg-normalizer-00
>>>>=20
>>>>=20
>>>> Abstract:
>>>>  This memo describes a Normalization Marker (NM) at DiffServ (DS)
>>>>  nodes to normalize the distribution of DS codepoint (DSCP) =
markings
>>>>  for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM =
is
>>>>  useful for traffic conditioning with Active Queue Management (AQM)
>>>>  when the AF PHB group is deployed for multimedia service classes =
that
>>>>  carry video packets with advanced codec technologies.  Using NM =
can
>>>>  offer an incentive that is however missing in today's ecosystem of
>>>>  practical deployment, i.e., when congestion occurs in the AF PHB
>>>>  group, the network should allow a video codec to benefit from its
>>>>  effort of generating finer layers of intra-flow precedence (IFP) =
with
>>>>  discardable packets in a more balanced distribution.
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> The IETF Secretariat
>>>>=20
>>>=20
>>> --=20
>>> ---
>>> Toerless Eckert, eckert@cisco.com
>>> Cisco NSSTG Systems & Technology Architecture
>>> SDN: Let me play with the network, mommy!
>>>=20
>>=20
>=20
> --=20
> ---
> Toerless Eckert, eckert@cisco.com
> Cisco NSSTG Systems & Technology Architecture
> SDN: Let me play with the network, mommy!
>=20


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Subject: Re: [tsvwg] draft-ietf-tsvwg-sctp-failover: comments and suggestions
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--089e01493bf44406a804e18fa303
Content-Type: text/plain; charset=ISO-8859-1

Hi Karen,

Thanks for your comments. I also added my comments in line.

On Thu, Jul 4, 2013 at 8:38 AM, <Karen.Nielsen@tieto.com> wrote:

> Hi All,
>
> Please find some additional comments inline below.
>
> >
> >- Section 5.1: CWND Handling for Potentially Failed Destinations
> >
> <snip>
> >
> >I agree with you to stick to the behavior of RFC4960 as default behavior.
> We've
> >been exploring several possibilities for performance improvement and
> discussing
> >using 2 MTU might be an idea, although we think we'll need further
> discussions.
> >Preethi might have some more comments on this point.
> >
>
> Please accept the following addition comments:
>
> The motivation for the 2MTU, I now understand, is to aim to mitigate
> unwanted effects of
> path bouncing resulting from a spurious T3-timeout followed by failover
> and subsequent "immediate" switchback operation of RFC4960 once HB ACK
> is acknowledged within the back-off'ed RTO.
>
> While I think that one can very well say in this document that the CWND
> could be set to 2MTU
> in case the first HB sent on a PF path is successfully ack'ed, then I
> would like to note that
> this then is motivated by a desire, first and foremost, to be backward
> compatible with RC4960, more than it
> really necessarily addresses how to best mitigate throughput degradation
> effects of spurious T3-timeouts.
> Indeed for implementations that operate with different switchback modes
> than the immediate switchback operation of RFC4960,
>  (e.g. operate with certain permanent failover modes), then the set of the
> CWND to 2 MTU on the previous PF path
> is not of a big significance as it can be most optimal, from a performance
> perspective,
> to stay on the new data carrying path, which at this point in time may
> have a CWND of IW, IW+1MTU or even higher.
> I think that the motivation for setting the CWND to 2MTU possibly should
> be made conditionally on whether the SCTP indeed follows
> the immediate switchback operation of RFC4960.
>
> One further observation is that in a situation where it is only the 2dn or
> the 3rd (or 4th..) of the HBs sent on the PF path, that is successful, then
> setting the CWND to 2 MTU may actually constitute a decrease of the CWND.
> This is the case if the implementation
> follows RC4960 and raise the CWND to IW (4MTU) after an idle period of one
> RTO following the T3 timeout.
> When in the text it is said that the CWND should be set to 1MTU at the
> state transition from PF to ACTIVE does this then very explicitly
> addresses the fact that unsuccessful HBs, sent on a PF path, shall keep
> the CWND down at 1MTU as otherwise data failures would ?
> If so then I think that it could be good to add this clarification.
>

Ok. I see your point.
I think we should explicitly say that we should set CWND to 1MTU after
unsuccessful HBs.
I don't think using 4MTU after seeing HB timeout is good approach. Thanks
for pointing out.


> Final thoughts:
>
> We have in real deployment experienced fault situations where HBs
> consistently were able to get through on the primary path,
> but where the path broke due to congestion once it got loaded with data.
> There are different ways in which one can improve the
> protocol operation to best tackle such a situation. Our present solution,
> which is to deploy a permanent failover mode together with PF
> and looking for a way to add reliability to the T3 timeouts, may not be
> the ultimate way to solve this, but I would like to note that
> in such a situation then consistently returning to the primary path and
> setting CWND = 2MTU is not the right thing to do.
>
>
> >- Section 5.3: Permanent Failover
> >
> >We have good experiences from live networks with running SCTP in a "no
> >switchback" mode of operation with no automatic migration back to a
> preferred
> >primary path. Once the primary path is left due to it being potentially
> failed, then a
> >selected alternative path automatically becomes the new primary path.
> >
> >This mode of operation corresponds to a simplified variant of the
> permanent
> >failover operation proposed in [CAR002] with alpha = beta =
> >SCTP_PEER_ADDR:THLDS. The mode of operation is run with notifications of
> >change of primary path being provided to upper layers via RFC6458
> >SCTP_PEER_ADDR_CHANGE semantics (SCTP_ADDR_MADE_PRIM).
> >
> >The mode of operation does not suit all purposes, but in networks
> environments
> >where a number of equally preferred paths are available, then by this
> mode of
> >operation one minimizes the effects of unnecessary path bouncing and the
> thereby
> >associated effects of throughput degradation due to path changes and slow
> start
> >operation.
> >
> >Based on our experiences we would recommend and support that the Permanent
> >Failover mode of operation, at least in the simple variant described
> above, is
> >promoted as a valid alternative approach. For the purpose of activation
> of this
> >mode of operation a switchback (or failover) mode operation parameter
> could be
> >introduced as configurable by the introduction of an appropriate new
> read/write
> >socket option in the Socket API.
> >
> >
> >Yes. I agree that there will be a situation where Permanent Failover work
> effectively.
> >OTOH, supporting this prevents us from keeping RFC4960 behavior as RFC4960
> >doesn't have this mode.
> >But, I believe this is a very interesting point and we would like to get
> more feedback
> >in order to think about this more.
> >
>
> I understand to continue to support RFC4960 behaviour, but I suppose that
> one can add to RFC4960
> behaviour :-) We have the simple "no switchback mode" described above
> running in deployment.
>

Yes, I think that's doable. I think the question here is the degree of
recommendation for this.
The easiest one is just saying "An implementation MAY support no switchback
mode", but I'm gueesing some people might prefer more explicit expressions.



> >- Section 5.4: The impact of HB timeouts in association error counter
> >
> >The draft recommends for excluding HB timeouts from contributing to the
> >(association) error counter. We fully support this approach for all
> timeouts of HBs
> >that are send to probe the state of destinations addresses that are in PF
> state and
> >we would even propose for the draft to demand that generally timeouts
> stemming
> >from HBs, which are not sent on the Primary Path, will be excluded from
> the
> >association error counter.
> >
> >Thereby the association error counter and thus the association fate
> reflects the
> >forward progress of the data transmission to peer endpoint while at the
> same time it
> >will be appropriately isolated from timeouts of HBs sent on non-data
> carrying paths.
> >
> >Allowing timeout of HBs sent on the Primary Path to impact the
> association error
> >counter will serve to validate the aliveness of the peer endpoint when
> there is no
> >data in transit.
> >
> >This suggestion follows the thought expressed in
> http://www.ietf.org/mail-
> >archive/web/tsvwg/current/msg11294.html and it is thus believed to fulfil
> the original
> >intend of RFC4960.
> >
> >This is a bit difficult point for me. I don't know much about the
> original intention of
> >RFC4960, but if we follow what is written there, i think we need to
> increment
> >association error count. (Please correct me if I misunderstand this
> point) So, my
> >personal preference is to set bigger AMR to avoid early termination so
> that we can
> >keep the RFC4960 behavior.
> >However, we don't mind discussing to update the logic described in 4960
> for more
> >sophisticated solution.
> >
>
> Thanks !
>
> Yes, I concur that the text of section 8.1 of RFC4960 stipulates that all
> HB Acks and HB timeouts
> influence the association error counter. But from the referenced
> discussion thread,
>  http://www.ietf.org/mail-archive/web/tsvwg/current/msg11294.html (simply copying it in here to avoid repeating the arguments)
>  it also seems that by purpose no keywords were used when writing these
> paragraphs.
> Further as the Potentially Feature was not defined at the time, I am not
> sure that
> it would be valid to say that the text of RFC4960 apply (at least) to the
> HBs sent as part of this function.
>
> From our perspective what we would like to support and see happening is
> that all HBs, whether sent as part of the PF function or not, influence
> the association error counter
> if and only if they are transmitted on the Primary Path (and then with
> standard logic running making sure that all data transmissions
> prevent (or cancel in race conditions) any influence from HBs).
>
> Thereby the association error counter will be influenced by (and will thus
> reflect) the data transmission only.
> Except in idle situation where HBs on Primary Path will serve to update
> the association error counter in a similar manner as data would.
> Thereby one get a very clean and reliable implementation of the
> association error counter. By reliable, I mean that one can use
> the same AMR value regardless of the level of HB probing that one have
> activated and regardless of whether the Potentially Failed feature is
> activated or not.
> Such an implementation also makes the AMR value be independent from the
> number of paths supported in the association context.
> One can choose to use a value of AMR proportional with the number of paths
> in order to have all paths be tried a number of times before association
> closure, but
> with the choice suggested, one is not forced to "artificially" raise the
> AMR value to make room for the HBs that will fire as part of the
> potentially failed feature as well
> as for some fraction of the Path Monitoring HBs that can happen to fire
> during the same timeframe.
>
> An additional thought is that in the odd cases where HBs get through
> whereas data do not, then I think that it is contra-productive to have the
> association error counter be reset
> as long as data consistently fail.
>
> I, personally, does not so much like the suggestion to use a "maximal" or
> "bigger" AMR. I think that it will be complex to deduce exactly
> what failure time (aka number of data retransmission)  that a certain AMR
> value will guarantee you,
> whereas when only data transmission influences the AMR, things will be
> clean, simple and predictable.
>

I agree that the approach described in the draft is not the best approach.
But, since it doesn't violate RFC4960, we recommend it in the draft.
I think what you suggest sounds reasonable and prefer to update 4960 based
on this, although I guess we'll need more discussions.

 >
> >- Section 5.1 Thoughts on the destination address state machine
> >
> >In RFC4960 address state machine, notification of cease of use of the
> current
> >primary path as transmission path for new data is provided to upper layer
> via
> >(coupled with) the announcement of the destination as being UNREACHABLE.
> With
> >the introduction of the potentially failed state as a semi-hidden state
> (not resulting in
> >notified state transition to UL) the information on the fail-over to use
> a new data
> >carrying path is lost.
> >While there may be merits in suppressing notifications on potentially
> failed
> >addresses, then we believe that it is unfortunate that the notification
> of failover from
> >usage of the primary path for new data is lost.
> >This is especially problematic in path bouncing scenarios where the
> primary path is
> >unstable but still works some. E.g., HBs may go through, but the path
> breaks (data
> >transmission fails) as soon as it is more loaded. In such a situation no
> notifications
> >may be generated to UL, even if new data transmission consistently cycles
> to and
> >from the primary path. The solution to such path bouncing scenarios will
> have to
> >come from more advanced path failover mechanism, like [CAR002] (and would
> not
> >be solved just by notification), but it would be a great help if
> appropriate
> >notifications were provided.
> >
> >I think it will be OK to send notification as long as it's a new
> notification type. But
> >we should not send notifications defined in RFC6458.
>
> ok - I assume this is for backward compatible reasons.
>

Yes. As far as I can think that's the reason.

 >I guess having a new notification type won't cause problems, but I would
> like to get
> >more feedback on this.
>
> Ok - we can come back with a suggestion.
>

Thank you so much for many valuable feedback. We really appreciate!

Regards,
--
Yoshifumi

--089e01493bf44406a804e18fa303
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Karen,<div><br></div><div>Thanks for your comments. I a=
lso added my comments in line.<br><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Thu, Jul 4, 2013 at 8:38 AM,  <span dir=3D"ltr">&lt;<a =
href=3D"mailto:Karen.Nielsen@tieto.com" target=3D"_blank">Karen.Nielsen@tie=
to.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi All,<br>
<br>
Please find some additional comments inline below.<br>
<div><br>
&gt;<br>
&gt;- Section 5.1: CWND Handling for Potentially Failed Destinations<br>
&gt;<br>
</div>&lt;snip&gt;<br>
<div>&gt;<br>
&gt;I agree with you to stick to the behavior of RFC4960 as default behavio=
r. We&#39;ve<br>
&gt;been exploring several possibilities for performance improvement and di=
scussing<br>
&gt;using 2 MTU might be an idea, although we think we&#39;ll need further =
discussions.<br>
&gt;Preethi might have some more comments on this point.<br>
&gt;<br>
<br>
</div>Please accept the following addition comments:<br>
<br>
The motivation for the 2MTU, I now understand, is to aim to mitigate unwant=
ed effects of<br>
path bouncing resulting from a spurious T3-timeout followed by failover<br>
and subsequent &quot;immediate&quot; switchback operation of RFC4960 once H=
B ACK<br>
is acknowledged within the back-off&#39;ed RTO.<br>
<br>
While I think that one can very well say in this document that the CWND cou=
ld be set to 2MTU<br>
in case the first HB sent on a PF path is successfully ack&#39;ed, then I w=
ould like to note that<br>
this then is motivated by a desire, first and foremost, to be backward comp=
atible with RC4960, more than it<br>
really necessarily addresses how to best mitigate throughput degradation ef=
fects of spurious T3-timeouts.<br>
Indeed for implementations that operate with different switchback modes tha=
n the immediate switchback operation of RFC4960,<br>
=A0(e.g. operate with certain permanent failover modes), then the set of th=
e CWND to 2 MTU on the previous PF path<br>
is not of a big significance as it can be most optimal, from a performance =
perspective,<br>
to stay on the new data carrying path, which at this point in time may have=
 a CWND of IW, IW+1MTU or even higher.<br>
I think that the motivation for setting the CWND to 2MTU possibly should be=
 made conditionally on whether the SCTP indeed follows<br>
the immediate switchback operation of RFC4960.<br>
<br>
One further observation is that in a situation where it is only the 2dn or =
the 3rd (or 4th..) of the HBs sent on the PF path, that is successful, then=
<br>
setting the CWND to 2 MTU may actually constitute a decrease of the CWND. T=
his is the case if the implementation<br>
follows RC4960 and raise the CWND to IW (4MTU) after an idle period of one =
RTO following the T3 timeout.<br>
When in the text it is said that the CWND should be set to 1MTU at the stat=
e transition from PF to ACTIVE does this then very explicitly<br>
addresses the fact that unsuccessful HBs, sent on a PF path, shall keep the=
 CWND down at 1MTU as otherwise data failures would ?<br>
If so then I think that it could be good to add this clarification.<br></bl=
ockquote><div><br></div><div>Ok. I see your point.<br></div><div>I think we=
 should explicitly say that we should set CWND to 1MTU after unsuccessful H=
Bs. <br>
</div><div>I don&#39;t think using 4MTU after seeing HB timeout is good app=
roach. Thanks for pointing out.<br></div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<br>
Final thoughts:<br>
<br>
We have in real deployment experienced fault situations where HBs consisten=
tly were able to get through on the primary path,<br>
but where the path broke due to congestion once it got loaded with data. Th=
ere are different ways in which one can improve the<br>
protocol operation to best tackle such a situation. Our present solution, w=
hich is to deploy a permanent failover mode together with PF<br>
and looking for a way to add reliability to the T3 timeouts, may not be the=
 ultimate way to solve this, but I would like to note that<br>
in such a situation then consistently returning to the primary path and set=
ting CWND =3D 2MTU is not the right thing to do.<br>
<div><div><br>
<br>
&gt;- Section 5.3: Permanent Failover<br>
&gt;<br>
&gt;We have good experiences from live networks with running SCTP in a &quo=
t;no<br>
&gt;switchback&quot; mode of operation with no automatic migration back to =
a preferred<br>
&gt;primary path. Once the primary path is left due to it being potentially=
 failed, then a<br>
&gt;selected alternative path automatically becomes the new primary path.<b=
r>
&gt;<br>
&gt;This mode of operation corresponds to a simplified variant of the perma=
nent<br>
&gt;failover operation proposed in [CAR002] with alpha =3D beta =3D<br>
&gt;SCTP_PEER_ADDR:THLDS. The mode of operation is run with notifications o=
f<br>
&gt;change of primary path being provided to upper layers via RFC6458<br>
&gt;SCTP_PEER_ADDR_CHANGE semantics (SCTP_ADDR_MADE_PRIM).<br>
&gt;<br>
&gt;The mode of operation does not suit all purposes, but in networks envir=
onments<br>
&gt;where a number of equally preferred paths are available, then by this m=
ode of<br>
&gt;operation one minimizes the effects of unnecessary path bouncing and th=
e thereby<br>
&gt;associated effects of throughput degradation due to path changes and sl=
ow start<br>
&gt;operation.<br>
&gt;<br>
&gt;Based on our experiences we would recommend and support that the Perman=
ent<br>
&gt;Failover mode of operation, at least in the simple variant described ab=
ove, is<br>
&gt;promoted as a valid alternative approach. For the purpose of activation=
 of this<br>
&gt;mode of operation a switchback (or failover) mode operation parameter c=
ould be<br>
&gt;introduced as configurable by the introduction of an appropriate new re=
ad/write<br>
&gt;socket option in the Socket API.<br>
&gt;<br>
&gt;<br>
&gt;Yes. I agree that there will be a situation where Permanent Failover wo=
rk effectively.<br>
&gt;OTOH, supporting this=A0prevents us from keeping RFC4960 behavior as RF=
C4960<br>
&gt;doesn&#39;t have this mode.<br>
&gt;But, I believe this is a very interesting point and we would like to ge=
t more feedback<br>
&gt;in order to think about this more.<br>
&gt;<br>
<br>
</div></div>I understand to continue to support RFC4960 behaviour, but I su=
ppose that one can add to RFC4960<br>
behaviour :-) We have the simple &quot;no switchback mode&quot; described a=
bove running in deployment.<br></blockquote><div><br></div><div>Yes, I thin=
k that&#39;s doable. I think the question here is the degree of recommendat=
ion for this.</div>


<div>The easiest one is just saying &quot;An implementation MAY support no =
switchback mode&quot;, but I&#39;m gueesing some people might prefer more e=
xplicit expressions.=A0</div><div><br></div><div><br></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">


<div><br>
&gt;- Section 5.4: The impact of HB timeouts in association error counter<b=
r>
&gt;<br>
&gt;The draft recommends for excluding HB timeouts from contributing to the=
<br>
&gt;(association) error counter. We fully support this approach for all tim=
eouts of HBs<br>
&gt;that are send to probe the state of destinations addresses that are in =
PF state and<br>
&gt;we would even propose for the draft to demand that generally timeouts s=
temming<br>
&gt;from HBs, which are not sent on the Primary Path, will be excluded from=
 the<br>
&gt;association error counter.<br>
&gt;<br>
&gt;Thereby the association error counter and thus the association fate ref=
lects the<br>
&gt;forward progress of the data transmission to peer endpoint while at the=
 same time it<br>
&gt;will be appropriately isolated from timeouts of HBs sent on non-data ca=
rrying paths.<br>
&gt;<br>
&gt;Allowing timeout of HBs sent on the Primary Path to impact the associat=
ion error<br>
&gt;counter will serve to validate the aliveness of the peer endpoint when =
there is no<br>
&gt;data in transit.<br>
&gt;<br>
&gt;This suggestion follows the thought expressed in <a href=3D"http://www.=
ietf.org/mail-" target=3D"_blank">http://www.ietf.org/mail-</a><br>
&gt;archive/web/tsvwg/current/msg11294.html and it is thus believed to fulf=
il the original<br>
&gt;intend of RFC4960.<br>
&gt;<br>
&gt;This is a bit difficult point for me. I don&#39;t know much about the o=
riginal intention of<br>
&gt;RFC4960, but if we follow what is written there, i think we need to inc=
rement<br>
&gt;association error count. (Please correct me if I misunderstand this poi=
nt) So, my<br>
&gt;personal preference is to set bigger AMR to avoid early termination so =
that we can<br>
&gt;keep the RFC4960 behavior.<br>
&gt;However, we don&#39;t mind discussing to update the logic described in =
4960 for more<br>
&gt;sophisticated solution.<br>
&gt;<br>
<br>
</div>Thanks !<br>
<br>
Yes, I concur that the text of section 8.1 of RFC4960 stipulates that all H=
B Acks and HB timeouts<br>
influence the association error counter. But from the referenced discussion=
 thread,<br>
=A0<a href=3D"http://www.ietf.org/mail-archive/web/tsvwg/current/msg11294.h=
tml" target=3D"_blank">http://www.ietf.org/mail-archive/web/tsvwg/current/m=
sg11294.html</a> =A0(simply copying it in here to avoid repeating the argum=
ents)<br>



=A0it also seems that by purpose no keywords were used when writing these p=
aragraphs.<br>
Further as the Potentially Feature was not defined at the time, I am not su=
re that<br>
it would be valid to say that the text of RFC4960 apply (at least) to the H=
Bs sent as part of this function.<br>
<br>
>From our perspective what we would like to support and see happening is<br>
that all HBs, whether sent as part of the PF function or not, influence the=
 association error counter<br>
if and only if they are transmitted on the Primary Path (and then with stan=
dard logic running making sure that all data transmissions<br>
prevent (or cancel in race conditions) any influence from HBs).<br>
<br>
Thereby the association error counter will be influenced by (and will thus =
reflect) the data transmission only.<br>
Except in idle situation where HBs on Primary Path will serve to update the=
 association error counter in a similar manner as data would.<br>
Thereby one get a very clean and reliable implementation of the association=
 error counter. By reliable, I mean that one can use<br>
the same AMR value regardless of the level of HB probing that one have acti=
vated and regardless of whether the Potentially Failed feature is activated=
 or not.<br>
Such an implementation also makes the AMR value be independent from the num=
ber of paths supported in the association context.<br>
One can choose to use a value of AMR proportional with the number of paths =
in order to have all paths be tried a number of times before association cl=
osure, but<br>
with the choice suggested, one is not forced to &quot;artificially&quot; ra=
ise the AMR value to make room for the HBs that will fire as part of the po=
tentially failed feature as well<br>
as for some fraction of the Path Monitoring HBs that can happen to fire dur=
ing the same timeframe.<br>
<br>
An additional thought is that in the odd cases where HBs get through wherea=
s data do not, then I think that it is contra-productive to have the associ=
ation error counter be reset<br>
as long as data consistently fail.<br>
<br>
I, personally, does not so much like the suggestion to use a &quot;maximal&=
quot; or &quot;bigger&quot; AMR. I think that it will be complex to deduce =
exactly<br>
what failure time (aka number of data retransmission) =A0that a certain AMR=
 value will guarantee you,<br>
whereas when only data transmission influences the AMR, things will be clea=
n, simple and predictable.<br></blockquote><div><br></div><div>I agree that=
 the approach described in the draft is not the best approach. But, since i=
t doesn&#39;t violate RFC4960, we recommend it in the draft.</div>


<div>I think what you suggest sounds reasonable and prefer to update 4960 b=
ased on this, although I guess we&#39;ll need more discussions.<br><br></di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">


<div>
&gt;<br>
&gt;- Section 5.1 Thoughts on the destination address state machine<br>
&gt;<br>
&gt;In RFC4960 address state machine, notification of cease of use of the c=
urrent<br>
&gt;primary path as transmission path for new data is provided to upper lay=
er via<br>
&gt;(coupled with) the announcement of the destination as being UNREACHABLE=
. With<br>
&gt;the introduction of the potentially failed state as a semi-hidden state=
 (not resulting in<br>
&gt;notified state transition to UL) the information on the fail-over to us=
e a new data<br>
&gt;carrying path is lost.<br>
&gt;While there may be merits in suppressing notifications on potentially f=
ailed<br>
&gt;addresses, then we believe that it is unfortunate that the notification=
 of failover from<br>
&gt;usage of the primary path for new data is lost.<br>
&gt;This is especially problematic in path bouncing scenarios where the pri=
mary path is<br>
&gt;unstable but still works some. E.g., HBs may go through, but the path b=
reaks (data<br>
&gt;transmission fails) as soon as it is more loaded. In such a situation n=
o notifications<br>
&gt;may be generated to UL, even if new data transmission consistently cycl=
es to and<br>
&gt;from the primary path. The solution to such path bouncing scenarios wil=
l have to<br>
&gt;come from more advanced path failover mechanism, like [CAR002] (and wou=
ld not<br>
&gt;be solved just by notification), but it would be a great help if approp=
riate<br>
&gt;notifications were provided.<br>
&gt;<br>
&gt;I think it will be OK to send notification as long as it&#39;s a new no=
tification type. But<br>
&gt;we should not send notifications defined in RFC6458.<br>
<br>
</div>ok - I assume this is for backward compatible reasons.<br></blockquot=
e><div><br></div><div>Yes. As far as I can think that&#39;s the reason.</di=
v><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">


<div>
&gt;I guess having a new notification type won&#39;t cause problems, but I =
would like to get<br>
&gt;more feedback on this.<br>
<br>
</div>Ok - we can come back with a suggestion.<br></blockquote><div><br></d=
iv><div>Thank you so much for many valuable feedback. We really appreciate!=
</div><div><br></div><div>Regards,</div><div>--</div>
<div>Yoshifumi=A0</div></div><br></div></div></div>

--089e01493bf44406a804e18fa303--

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : DTLS Encapsulation of SCTP Packets
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Randall R. Stewart
                          Michael Tuexen
	Filename        : draft-ietf-tsvwg-sctp-dtls-encaps-01.txt
	Pages           : 7
	Date            : 2013-07-15

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol originally defined to run on top of the network protocols
   IPv4 or IPv6.  This document specifies how SCTP can be used on top of
   the Datagram Transport Layer Security (DTLS) protocol.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-dtls-encaps

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-01

A diff from the previous version is available at:
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Internet-Drafts are also available by anonymous FTP at:
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : Recommendations for Transport Port Uses
	Author(s)       : Joe Touch
	Filename        : draft-ietf-tsvwg-port-use-02.txt
	Pages           : 18
	Date            : 2013-07-15

Abstract:
   This document provides recommendations to application and service
   designers on how to use the transport protocol port number space to
   help in its preservation.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-port-use

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http://tools.ietf.org/html/draft-ietf-tsvwg-port-use-02

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From jmpolk@cisco.com  Mon Jul 15 23:08:05 2013
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Date: Tue, 16 Jul 2013 01:07:59 -0500
To: tsvwg@ietf.org
From: James Polk <jmpolk@cisco.com>
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Subject: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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(as an author)

Toerless and I put together a draft about legacy rate-adaptation 
based only on loss vs. what RMCAT is looking to (for RTCWEB), which 
is based on delay and loss.  Here's the URL.

http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt

It's more raw than we had in mind, but we believe this is necessary, 
based on implementation experience and what users and customers have 
in their networks, or are planning on having in their networks soon.

James & Toerless


From mramalho@cisco.com  Tue Jul 16 06:36:22 2013
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOggZ1ibmIYctWIUyl3FUXvgClMZlnRd2QgAAELjA=
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Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. loss-only	rate-adaptive for	DiffServ
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TSVWG Members,

I should have copied you on my email to the RMCAT mailer below.

I strongly support the creation of a new DSCP for transports in which their=
 congestion control can adapt based on delay.

The current ID in support of this (http://www.ietf.org/internet-drafts/draf=
t-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt ) is a work in progres=
s, but a step in the right direction.

It is a given that the ability of the eventual RMCAT adaptation mechanisms =
to achieve low delay will be function of the other dominant traffic it is c=
ompeting against. If the dominant competing traffic DEPENDS on loss, the RM=
CAT packets are held hostage to being delayed by the bottleneck queue delay=
 maximum.

Thus, whenever possible, it will be preferable for RMCAT flows to compete w=
ith other congestion control transports that adapt on delay. Even this may =
not get us to the desired low-delay goals when a portion of the traffic has=
 long RTTs (i.e., adaptation control loops that are long in time), or for l=
inks that have a highly-time varying capacity,  but it will help for a lot =
of common bottleneck topologies (e.g., slowly time-varying access bufferblo=
at).

It is my hope that this topic has some discussion time in the Berlin Transp=
ort WG (not the specific codepoint to be chosen, but rather the need for on=
e).

Off Soapbox,

Michael Ramalho, Ph.D.

-----Original Message-----
From: Michael Ramalho (mramalho)=20
Sent: Tuesday, July 16, 2013 9:06 AM
To: rmcat@ietf.org
Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only rate-=
adaptive for DiffServ

RMCAT Design Team,

The draft Lars references below is a formal request for a DSCP dedicated to=
 "RMCAT-only (or other nice delay-based cc) traffic".

It will take a while to become socialized ... and we can progress our RMCAT=
 work in the interim.

Michael Ramalho

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of E=
ggert, Lars
Sent: Tuesday, July 16, 2013 5:26 AM
To: WG WG
Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only rate-adap=
tive for DiffServ

Possibly of interest to RMCAT.

Begin forwarded message:

> From: James Polk <jmpolk@cisco.com>
> Subject: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for Di=
ffServ
> Date: July 16, 2013 8:07:59 GMT+02:00
> To: <tsvwg@ietf.org>
>=20
> (as an author)
>=20
> Toerless and I put together a draft about legacy rate-adaptation based on=
ly on loss vs. what RMCAT is looking to (for RTCWEB), which is based on del=
ay and loss.  Here's the URL.
>=20
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt
>=20
> It's more raw than we had in mind, but we believe this is necessary, base=
d on implementation experience and what users and customers have in their n=
etworks, or are planning on having in their networks soon.
>=20
> James & Toerless
>=20


From chelai@cisco.com  Tue Jul 16 11:36:41 2013
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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>, "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOgil6VGqxx9YuYkWplWeJnoICXplnn7LQ
Date: Tue, 16 Jul 2013 18:36:31 +0000
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com>
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Subject: Re: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.	loss-only	rate-adaptive for	DiffServ
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Same here... I feel it makes sense to create a new DSCP / CS4 for transport=
 of video flows that have rate adaptive behaviors and/or intra-flow prefere=
ntial drop priorities as indicated in the I-D. The service provided by the =
network may thus be considered new, i.e. different from what exists in AF4x=
, so IMHO, using a new or reusing CS4 service class adds clarity for the us=
er applications.

Regards,
CJ


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of M=
ichael Ramalho (mramalho)
Sent: Tuesday, July 16, 2013 6:36 AM
To: tsvwg@ietf.org
Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. loss-only rate-=
adaptive for DiffServ

TSVWG Members,

I should have copied you on my email to the RMCAT mailer below.

I strongly support the creation of a new DSCP for transports in which their=
 congestion control can adapt based on delay.

The current ID in support of this (http://www.ietf.org/internet-drafts/draf=
t-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt ) is a work in progres=
s, but a step in the right direction.

It is a given that the ability of the eventual RMCAT adaptation mechanisms =
to achieve low delay will be function of the other dominant traffic it is c=
ompeting against. If the dominant competing traffic DEPENDS on loss, the RM=
CAT packets are held hostage to being delayed by the bottleneck queue delay=
 maximum.

Thus, whenever possible, it will be preferable for RMCAT flows to compete w=
ith other congestion control transports that adapt on delay. Even this may =
not get us to the desired low-delay goals when a portion of the traffic has=
 long RTTs (i.e., adaptation control loops that are long in time), or for l=
inks that have a highly-time varying capacity,  but it will help for a lot =
of common bottleneck topologies (e.g., slowly time-varying access bufferblo=
at).

It is my hope that this topic has some discussion time in the Berlin Transp=
ort WG (not the specific codepoint to be chosen, but rather the need for on=
e).

Off Soapbox,

Michael Ramalho, Ph.D.

-----Original Message-----
From: Michael Ramalho (mramalho)=20
Sent: Tuesday, July 16, 2013 9:06 AM
To: rmcat@ietf.org
Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only rate-=
adaptive for DiffServ

RMCAT Design Team,

The draft Lars references below is a formal request for a DSCP dedicated to=
 "RMCAT-only (or other nice delay-based cc) traffic".

It will take a while to become socialized ... and we can progress our RMCAT=
 work in the interim.

Michael Ramalho

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of E=
ggert, Lars
Sent: Tuesday, July 16, 2013 5:26 AM
To: WG WG
Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only rate-adap=
tive for DiffServ

Possibly of interest to RMCAT.

Begin forwarded message:

> From: James Polk <jmpolk@cisco.com>
> Subject: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for Di=
ffServ
> Date: July 16, 2013 8:07:59 GMT+02:00
> To: <tsvwg@ietf.org>
>=20
> (as an author)
>=20
> Toerless and I put together a draft about legacy rate-adaptation based on=
ly on loss vs. what RMCAT is looking to (for RTCWEB), which is based on del=
ay and loss.  Here's the URL.
>=20
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt
>=20
> It's more raw than we had in mind, but we believe this is necessary, base=
d on implementation experience and what users and customers have in their n=
etworks, or are planning on having in their networks soon.
>=20
> James & Toerless
>=20


From eckelcu@cisco.com  Tue Jul 16 13:47:01 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
Thread-Index: AQHOgeraA2Us2T761Eqijb60sjBbhZlnwdGA
Date: Tue, 16 Jul 2013 20:46:54 +0000
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for	DiffServ
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I read the draft and thank you for addressing this need. It does a good job=
 of making the case for the need to differentiate the DSCP assignments of d=
elay-and-loss-based vs. just-loss-based rate-adaptive video applications. W=
hile I do not agree with some of the finer points in the draft, including t=
he statement in section 3 that AF4x traffic in general is inelastic/fixed b=
itrate, I completely agree that this traffic is largely loss sensitive and =
yet often relies on loss based adaption.
The assignment of a new service class and the proposal for CS4 and CS4-Disc=
ardable for rate-adaptive video makes sense to me. At the same time, it is =
important to re-designate AF4x as you describe in order to match the realit=
y of today's deployments and avoid future confusion.
I look forward to seeing this work progress as it offer an incentive for ap=
plications to embrace congestion control algorithms such as those being dev=
eloped in RMCAT without fearing they will be overrun by non-delay adaptive =
traffic.

Cheers,
Charles


> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> James Polk (jmpolk)
> Sent: Monday, July 15, 2013 11:08 PM
> To: tsvwg@ietf.org
> Subject: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for Di=
ffServ
>=20
> (as an author)
>=20
> Toerless and I put together a draft about legacy rate-adaptation
> based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> is based on delay and loss.  Here's the URL.
>=20
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-
> classes-00.txt
>=20
> It's more raw than we had in mind, but we believe this is necessary,
> based on implementation experience and what users and customers have
> in their networks, or are planning on having in their networks soon.
>=20
> James & Toerless


From jmpolk@cisco.com  Tue Jul 16 14:06:53 2013
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To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
From: James Polk <jmpolk@cisco.com>
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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At 03:46 PM 7/16/2013, Charles Eckel (eckelcu) wrote:
>I read the draft and thank you for addressing this need. It does a 
>good job of making the case for the need to differentiate the DSCP 
>assignments of delay-and-loss-based vs. just-loss-based 
>rate-adaptive video applications. While I do not agree with some of 
>the finer points in the draft, including the statement in section 3 
>that AF4x traffic in general is inelastic/fixed bitrate,

Charles

Thank you for the quick review!

If we authors gave the impression that all AF4X marked traffic is 
inelastic and uses the 'same' fixed bitrates, we apologize. However, 
each AF4X flow from a given codec - be it audio or video - in today's 
implementations, pretty much use the same bitrate per flow (not the 
same bitrate for each flow). Do you agree with this clarification, or 
would you like to offer substitute/replacement text in the offending section?

James

>  I completely agree that this traffic is largely loss sensitive and 
> yet often relies on loss based adaption.
>The assignment of a new service class and the proposal for CS4 and 
>CS4-Discardable for rate-adaptive video makes sense to me. At the 
>same time, it is important to re-designate AF4x as you describe in 
>order to match the reality of today's deployments and avoid future confusion.
>I look forward to seeing this work progress as it offer an incentive 
>for applications to embrace congestion control algorithms such as 
>those being developed in RMCAT without fearing they will be overrun 
>by non-delay adaptive traffic.
>
>Cheers,
>Charles
>
>
> > -----Original Message-----
> > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> > James Polk (jmpolk)
> > Sent: Monday, July 15, 2013 11:08 PM
> > To: tsvwg@ietf.org
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only 
> rate-adaptive for DiffServ
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation
> > based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> > is based on delay and loss.  Here's the URL.
> >
> > 
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-
> > classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is necessary,
> > based on implementation experience and what users and customers have
> > in their networks, or are planning on having in their networks soon.
> >
> > James & Toerless


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Subject: Re: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. loss-only	rate-adaptive for	DiffServ
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At 08:36 AM 7/16/2013, Michael Ramalho (mramalho) wrote:
>TSVWG Members,
>
>I should have copied you on my email to the RMCAT mailer below.
>
>I strongly support the creation of a new DSCP for transports in 
>which their congestion control can adapt based on delay.
>
>The current ID in support of this 
>(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt 
>) is a work in progress, but a step in the right direction.

Michael

Thank you for the quick review.

The draft isn't very long (under 8 pages currently), and it is a work 
in progress - but do you have text or just points that this draft 
needs to cover that it doesn't currently?

James


>It is a given that the ability of the eventual RMCAT adaptation 
>mechanisms to achieve low delay will be function of the other 
>dominant traffic it is competing against. If the dominant competing 
>traffic DEPENDS on loss, the RMCAT packets are held hostage to being 
>delayed by the bottleneck queue delay maximum.
>
>Thus, whenever possible, it will be preferable for RMCAT flows to 
>compete with other congestion control transports that adapt on 
>delay. Even this may not get us to the desired low-delay goals when 
>a portion of the traffic has long RTTs (i.e., adaptation control 
>loops that are long in time), or for links that have a highly-time 
>varying capacity,  but it will help for a lot of common bottleneck 
>topologies (e.g., slowly time-varying access bufferbloat).
>
>It is my hope that this topic has some discussion time in the Berlin 
>Transport WG (not the specific codepoint to be chosen, but rather 
>the need for one).
>
>Off Soapbox,
>
>Michael Ramalho, Ph.D.
>
>-----Original Message-----
>From: Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 9:06 AM
>To: rmcat@ietf.org
>Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. 
>loss-only rate-adaptive for DiffServ
>
>RMCAT Design Team,
>
>The draft Lars references below is a formal request for a DSCP 
>dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>
>It will take a while to become socialized ... and we can progress 
>our RMCAT work in the interim.
>
>Michael Ramalho
>
>-----Original Message-----
>From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On 
>Behalf Of Eggert, Lars
>Sent: Tuesday, July 16, 2013 5:26 AM
>To: WG WG
>Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only 
>rate-adaptive for DiffServ
>
>Possibly of interest to RMCAT.
>
>Begin forwarded message:
>
> > From: James Polk <jmpolk@cisco.com>
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only 
> rate-adaptive for DiffServ
> > Date: July 16, 2013 8:07:59 GMT+02:00
> > To: <tsvwg@ietf.org>
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation 
> based only on loss vs. what RMCAT is looking to (for RTCWEB), which 
> is based on delay and loss.  Here's the URL.
> >
> > 
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is 
> necessary, based on implementation experience and what users and 
> customers have in their networks, or are planning on having in 
> their networks soon.
> >
> > James & Toerless
> >


From jmpolk@cisco.com  Tue Jul 16 14:11:33 2013
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Date: Tue, 16 Jul 2013 16:11:27 -0500
To: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>, "James Polk (jmpolk)" <jmpolk@cisco.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>, "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>
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Subject: Re: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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CJ

Thank you for the quick review.

I'll ask you the same question I just asked Michael, the draft isn't 
very long (under 8 pages currently), and it is a work in progress - 
but do you have text or just points that this draft needs to cover 
that it doesn't currently?

James

At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>Same here... I feel it makes sense to create a new DSCP / CS4 for 
>transport of video flows that have rate adaptive behaviors and/or 
>intra-flow preferential drop priorities as indicated in the I-D. The 
>service provided by the network may thus be considered new, i.e. 
>different from what exists in AF4x, so IMHO, using a new or reusing 
>CS4 service class adds clarity for the user applications.
>
>Regards,
>CJ
>
>
>-----Original Message-----
>From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On 
>Behalf Of Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 6:36 AM
>To: tsvwg@ietf.org
>Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. 
>loss-only rate-adaptive for DiffServ
>
>TSVWG Members,
>
>I should have copied you on my email to the RMCAT mailer below.
>
>I strongly support the creation of a new DSCP for transports in 
>which their congestion control can adapt based on delay.
>
>The current ID in support of this 
>(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt 
>) is a work in progress, but a step in the right direction.
>
>It is a given that the ability of the eventual RMCAT adaptation 
>mechanisms to achieve low delay will be function of the other 
>dominant traffic it is competing against. If the dominant competing 
>traffic DEPENDS on loss, the RMCAT packets are held hostage to being 
>delayed by the bottleneck queue delay maximum.
>
>Thus, whenever possible, it will be preferable for RMCAT flows to 
>compete with other congestion control transports that adapt on 
>delay. Even this may not get us to the desired low-delay goals when 
>a portion of the traffic has long RTTs (i.e., adaptation control 
>loops that are long in time), or for links that have a highly-time 
>varying capacity,  but it will help for a lot of common bottleneck 
>topologies (e.g., slowly time-varying access bufferbloat).
>
>It is my hope that this topic has some discussion time in the Berlin 
>Transport WG (not the specific codepoint to be chosen, but rather 
>the need for one).
>
>Off Soapbox,
>
>Michael Ramalho, Ph.D.
>
>-----Original Message-----
>From: Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 9:06 AM
>To: rmcat@ietf.org
>Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. 
>loss-only rate-adaptive for DiffServ
>
>RMCAT Design Team,
>
>The draft Lars references below is a formal request for a DSCP 
>dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>
>It will take a while to become socialized ... and we can progress 
>our RMCAT work in the interim.
>
>Michael Ramalho
>
>-----Original Message-----
>From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On 
>Behalf Of Eggert, Lars
>Sent: Tuesday, July 16, 2013 5:26 AM
>To: WG WG
>Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only 
>rate-adaptive for DiffServ
>
>Possibly of interest to RMCAT.
>
>Begin forwarded message:
>
> > From: James Polk <jmpolk@cisco.com>
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only 
> rate-adaptive for DiffServ
> > Date: July 16, 2013 8:07:59 GMT+02:00
> > To: <tsvwg@ietf.org>
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation 
> based only on loss vs. what RMCAT is looking to (for RTCWEB), which 
> is based on delay and loss.  Here's the URL.
> >
> > 
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is 
> necessary, based on implementation experience and what users and 
> customers have in their networks, or are planning on having in 
> their networks soon.
> >
> > James & Toerless
> >


From eckelcu@cisco.com  Tue Jul 16 16:32:42 2013
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive  for DiffServ
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Date: Tue, 16 Jul 2013 23:32:31 +0000
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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> -----Original Message-----
> From: James Polk (jmpolk)
> Sent: Tuesday, July 16, 2013 2:07 PM
> To: Charles Eckel (eckelcu); tsvwg@ietf.org
> Cc: James Polk (jmpolk)
> Subject: RE: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive fo=
r
> DiffServ
>=20
> At 03:46 PM 7/16/2013, Charles Eckel (eckelcu) wrote:
> >I read the draft and thank you for addressing this need. It does a
> >good job of making the case for the need to differentiate the DSCP
> >assignments of delay-and-loss-based vs. just-loss-based
> >rate-adaptive video applications. While I do not agree with some of
> >the finer points in the draft, including the statement in section 3
> >that AF4x traffic in general is inelastic/fixed bitrate,
>=20
> Charles
>=20
> Thank you for the quick review!
>=20
> If we authors gave the impression that all AF4X marked traffic is
> inelastic and uses the 'same' fixed bitrates, we apologize. However,
> each AF4X flow from a given codec - be it audio or video - in today's
> implementations, pretty much use the same bitrate per flow (not the
> same bitrate for each flow). Do you agree with this clarification, or
> would you like to offer substitute/replacement text in the offending sect=
ion?

I think video codec implementations in the AF4X class today often have vari=
able bit rates; however, those variations are primarily due to the amount o=
f motion and sometimes as an adaptation to loss. The latter is what is impo=
rtant in this discussion. Video codecs will continue to have variable bit r=
ates in accordance with the amount of motion, but if aligned with the outpu=
t of RMCAT, they will adapt based on delay as well as loss.=20

Cheers,
Charles

=20
> James
>=20
> >  I completely agree that this traffic is largely loss sensitive and
> > yet often relies on loss based adaption.
> >The assignment of a new service class and the proposal for CS4 and
> >CS4-Discardable for rate-adaptive video makes sense to me. At the
> >same time, it is important to re-designate AF4x as you describe in
> >order to match the reality of today's deployments and avoid future confu=
sion.
> >I look forward to seeing this work progress as it offer an incentive
> >for applications to embrace congestion control algorithms such as
> >those being developed in RMCAT without fearing they will be overrun
> >by non-delay adaptive traffic.
> >
> >Cheers,
> >Charles
> >
> >
> > > -----Original Message-----
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behal=
f
> Of
> > > James Polk (jmpolk)
> > > Sent: Monday, July 15, 2013 11:08 PM
> > > To: tsvwg@ietf.org
> > > Subject: [tsvwg] Submitted ID on delay vs. loss-only
> > rate-adaptive for DiffServ
> > >
> > > (as an author)
> > >
> > > Toerless and I put together a draft about legacy rate-adaptation
> > > based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> > > is based on delay and loss.  Here's the URL.
> > >
> > >
> > http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-
> service-
> > > classes-00.txt
> > >
> > > It's more raw than we had in mind, but we believe this is necessary,
> > > based on implementation experience and what users and customers have
> > > in their networks, or are planning on having in their networks soon.
> > >
> > > James & Toerless


From jmpolk@cisco.com  Tue Jul 16 16:40:36 2013
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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At 06:32 PM 7/16/2013, Charles Eckel (eckelcu) wrote:
> > -----Original Message-----
> > From: James Polk (jmpolk)
> > Sent: Tuesday, July 16, 2013 2:07 PM
> > To: Charles Eckel (eckelcu); tsvwg@ietf.org
> > Cc: James Polk (jmpolk)
> > Subject: RE: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for
> > DiffServ
> >
> > At 03:46 PM 7/16/2013, Charles Eckel (eckelcu) wrote:
> > >I read the draft and thank you for addressing this need. It does a
> > >good job of making the case for the need to differentiate the DSCP
> > >assignments of delay-and-loss-based vs. just-loss-based
> > >rate-adaptive video applications. While I do not agree with some of
> > >the finer points in the draft, including the statement in section 3
> > >that AF4x traffic in general is inelastic/fixed bitrate,
> >
> > Charles
> >
> > Thank you for the quick review!
> >
> > If we authors gave the impression that all AF4X marked traffic is
> > inelastic and uses the 'same' fixed bitrates, we apologize. However,
> > each AF4X flow from a given codec - be it audio or video - in today's
> > implementations, pretty much use the same bitrate per flow (not the
> > same bitrate for each flow). Do you agree with this clarification, or
> > would you like to offer substitute/replacement text in the 
> offending section?
>
>I think video codec implementations in the AF4X class today often 
>have variable bit rates; however, those variations are primarily due 
>to the amount of motion and sometimes as an adaptation to loss. The 
>latter is what is important in this discussion. Video codecs will 
>continue to have variable bit rates in accordance with the amount of 
>motion, but if aligned with the output of RMCAT, they will adapt 
>based on delay as well as loss.

Thanks - we'll include those points in the next version.

James


>Cheers,
>Charles
>
>
> > James
> >
> > >  I completely agree that this traffic is largely loss sensitive and
> > > yet often relies on loss based adaption.
> > >The assignment of a new service class and the proposal for CS4 and
> > >CS4-Discardable for rate-adaptive video makes sense to me. At the
> > >same time, it is important to re-designate AF4x as you describe in
> > >order to match the reality of today's deployments and avoid 
> future confusion.
> > >I look forward to seeing this work progress as it offer an incentive
> > >for applications to embrace congestion control algorithms such as
> > >those being developed in RMCAT without fearing they will be overrun
> > >by non-delay adaptive traffic.
> > >
> > >Cheers,
> > >Charles
> > >
> > >
> > > > -----Original Message-----
> > > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf
> > Of
> > > > James Polk (jmpolk)
> > > > Sent: Monday, July 15, 2013 11:08 PM
> > > > To: tsvwg@ietf.org
> > > > Subject: [tsvwg] Submitted ID on delay vs. loss-only
> > > rate-adaptive for DiffServ
> > > >
> > > > (as an author)
> > > >
> > > > Toerless and I put together a draft about legacy rate-adaptation
> > > > based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> > > > is based on delay and loss.  Here's the URL.
> > > >
> > > >
> > > http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-
> > service-
> > > > classes-00.txt
> > > >
> > > > It's more raw than we had in mind, but we believe this is necessary,
> > > > based on implementation experience and what users and customers have
> > > > in their networks, or are planning on having in their networks soon.
> > > >
> > > > James & Toerless


From mzanaty@cisco.com  Tue Jul 16 20:23:19 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>, "Cheng-Jia Lai (chelai)" <chelai@cisco.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>, "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOgmkPCJii7CPYUk2ch+5QoO+R/JloLzYQ
Date: Wed, 17 Jul 2013 03:23:11 +0000
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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I support something like what this draft proposes, to segregate RMCAT (and =
potentially other delay-adaptive traffic) from other traffic which does not=
 adapt to delay trends (and therefore incurs maximum queue delay for all tr=
affic in its queue). Of course, this only helps when such queue segregation=
 is available in the network, but effective active queue management is not =
available (otherwise there would be no delay signals for delay-adaptive tra=
ffic to act on).

An open question in my mind, that I think this draft needs to address, is w=
hether all delay-adaptive traffic should share the same queue/DSCP, or whet=
her we need different queues/DSCPs for different delay-adaptive protocols/a=
pplications (for example, RMCAT vs. LEDBAT).

Mo

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of J=
ames Polk (jmpolk)
Sent: Tuesday, July 16, 2013 5:11 PM
To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert (eckert); =
Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ

CJ

Thank you for the quick review.

I'll ask you the same question I just asked Michael, the draft isn't=20
very long (under 8 pages currently), and it is a work in progress -=20
but do you have text or just points that this draft needs to cover=20
that it doesn't currently?

James

At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>Same here... I feel it makes sense to create a new DSCP / CS4 for=20
>transport of video flows that have rate adaptive behaviors and/or=20
>intra-flow preferential drop priorities as indicated in the I-D. The=20
>service provided by the network may thus be considered new, i.e.=20
>different from what exists in AF4x, so IMHO, using a new or reusing=20
>CS4 service class adds clarity for the user applications.
>
>Regards,
>CJ
>
>
>-----Original Message-----
>From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On=20
>Behalf Of Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 6:36 AM
>To: tsvwg@ietf.org
>Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.=20
>loss-only rate-adaptive for DiffServ
>
>TSVWG Members,
>
>I should have copied you on my email to the RMCAT mailer below.
>
>I strongly support the creation of a new DSCP for transports in=20
>which their congestion control can adapt based on delay.
>
>The current ID in support of this=20
>(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt=20
>) is a work in progress, but a step in the right direction.
>
>It is a given that the ability of the eventual RMCAT adaptation=20
>mechanisms to achieve low delay will be function of the other=20
>dominant traffic it is competing against. If the dominant competing=20
>traffic DEPENDS on loss, the RMCAT packets are held hostage to being=20
>delayed by the bottleneck queue delay maximum.
>
>Thus, whenever possible, it will be preferable for RMCAT flows to=20
>compete with other congestion control transports that adapt on=20
>delay. Even this may not get us to the desired low-delay goals when=20
>a portion of the traffic has long RTTs (i.e., adaptation control=20
>loops that are long in time), or for links that have a highly-time=20
>varying capacity,  but it will help for a lot of common bottleneck=20
>topologies (e.g., slowly time-varying access bufferbloat).
>
>It is my hope that this topic has some discussion time in the Berlin=20
>Transport WG (not the specific codepoint to be chosen, but rather=20
>the need for one).
>
>Off Soapbox,
>
>Michael Ramalho, Ph.D.
>
>-----Original Message-----
>From: Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 9:06 AM
>To: rmcat@ietf.org
>Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.=20
>loss-only rate-adaptive for DiffServ
>
>RMCAT Design Team,
>
>The draft Lars references below is a formal request for a DSCP=20
>dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>
>It will take a while to become socialized ... and we can progress=20
>our RMCAT work in the interim.
>
>Michael Ramalho
>
>-----Original Message-----
>From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On=20
>Behalf Of Eggert, Lars
>Sent: Tuesday, July 16, 2013 5:26 AM
>To: WG WG
>Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only=20
>rate-adaptive for DiffServ
>
>Possibly of interest to RMCAT.
>
>Begin forwarded message:
>
> > From: James Polk <jmpolk@cisco.com>
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only=20
> rate-adaptive for DiffServ
> > Date: July 16, 2013 8:07:59 GMT+02:00
> > To: <tsvwg@ietf.org>
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation=20
> based only on loss vs. what RMCAT is looking to (for RTCWEB), which=20
> is based on delay and loss.  Here's the URL.
> >
> >=20
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is=20
> necessary, based on implementation experience and what users and=20
> customers have in their networks, or are planning on having in=20
> their networks soon.
> >
> > James & Toerless
> >


From gorry@erg.abdn.ac.uk  Tue Jul 16 23:50:16 2013
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <3879D71E758A7E4AA99A35DD8D41D3D91D494900@xmb-rcd-x14.cisco.com>
Date: Wed, 17 Jul 2013 07:50:09 +0100
From: gorry@erg.abdn.ac.uk
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Cc: "tsvwg@ietf.org" <tsvwg@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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I think the question of sharing diffserv is important, and my current
suggestion is that RMCAT should use one or at most two DSCPs and design a
CC that works well within the class - however this does not absolve RMCAT
from also having to work in the Internet without this class. This will
likely be the main use-case!

Although the proposal could help these applications - the IETF can not
somehow mandate this is universally deployed - and even if the DSCP(s)
were universally recognised, it would still be aggregated within some
networks and on L2 technologies that support fewer classes than DSCP
markings.

It would be interesting to know if others think we should be considering
more DSCPs for this?

Gorry

> I support something like what this draft proposes, to segregate RMCAT (and
> potentially other delay-adaptive traffic) from other traffic which does
> not adapt to delay trends (and therefore incurs maximum queue delay for
> all traffic in its queue). Of course, this only helps when such queue
> segregation is available in the network, but effective active queue
> management is not available (otherwise there would be no delay signals for
> delay-adaptive traffic to act on).
>
> An open question in my mind, that I think this draft needs to address, is
> whether all delay-adaptive traffic should share the same queue/DSCP, or
> whether we need different queues/DSCPs for different delay-adaptive
> protocols/applications (for example, RMCAT vs. LEDBAT).
>
> Mo
>
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
> James Polk (jmpolk)
> Sent: Tuesday, July 16, 2013 5:11 PM
> To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert (eckert);
> Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only
> rate-adaptive for DiffServ
>
> CJ
>
> Thank you for the quick review.
>
> I'll ask you the same question I just asked Michael, the draft isn't
> very long (under 8 pages currently), and it is a work in progress -
> but do you have text or just points that this draft needs to cover
> that it doesn't currently?
>
> James
>
> At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>>Same here... I feel it makes sense to create a new DSCP / CS4 for
>>transport of video flows that have rate adaptive behaviors and/or
>>intra-flow preferential drop priorities as indicated in the I-D. The
>>service provided by the network may thus be considered new, i.e.
>>different from what exists in AF4x, so IMHO, using a new or reusing
>>CS4 service class adds clarity for the user applications.
>>
>>Regards,
>>CJ
>>
>>
>>-----Original Message-----
>>From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
>>Behalf Of Michael Ramalho (mramalho)
>>Sent: Tuesday, July 16, 2013 6:36 AM
>>To: tsvwg@ietf.org
>>Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.
>>loss-only rate-adaptive for DiffServ
>>
>>TSVWG Members,
>>
>>I should have copied you on my email to the RMCAT mailer below.
>>
>>I strongly support the creation of a new DSCP for transports in
>>which their congestion control can adapt based on delay.
>>
>>The current ID in support of this
>>(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>>) is a work in progress, but a step in the right direction.
>>
>>It is a given that the ability of the eventual RMCAT adaptation
>>mechanisms to achieve low delay will be function of the other
>>dominant traffic it is competing against. If the dominant competing
>>traffic DEPENDS on loss, the RMCAT packets are held hostage to being
>>delayed by the bottleneck queue delay maximum.
>>
>>Thus, whenever possible, it will be preferable for RMCAT flows to
>>compete with other congestion control transports that adapt on
>>delay. Even this may not get us to the desired low-delay goals when
>>a portion of the traffic has long RTTs (i.e., adaptation control
>>loops that are long in time), or for links that have a highly-time
>>varying capacity,  but it will help for a lot of common bottleneck
>>topologies (e.g., slowly time-varying access bufferbloat).
>>
>>It is my hope that this topic has some discussion time in the Berlin
>>Transport WG (not the specific codepoint to be chosen, but rather
>>the need for one).
>>
>>Off Soapbox,
>>
>>Michael Ramalho, Ph.D.
>>
>>-----Original Message-----
>>From: Michael Ramalho (mramalho)
>>Sent: Tuesday, July 16, 2013 9:06 AM
>>To: rmcat@ietf.org
>>Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.
>>loss-only rate-adaptive for DiffServ
>>
>>RMCAT Design Team,
>>
>>The draft Lars references below is a formal request for a DSCP
>>dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>>
>>It will take a while to become socialized ... and we can progress
>>our RMCAT work in the interim.
>>
>>Michael Ramalho
>>
>>-----Original Message-----
>>From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
>>Behalf Of Eggert, Lars
>>Sent: Tuesday, July 16, 2013 5:26 AM
>>To: WG WG
>>Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>>Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only
>>rate-adaptive for DiffServ
>>
>>Possibly of interest to RMCAT.
>>
>>Begin forwarded message:
>>
>> > From: James Polk <jmpolk@cisco.com>
>> > Subject: [tsvwg] Submitted ID on delay vs. loss-only
>> rate-adaptive for DiffServ
>> > Date: July 16, 2013 8:07:59 GMT+02:00
>> > To: <tsvwg@ietf.org>
>> >
>> > (as an author)
>> >
>> > Toerless and I put together a draft about legacy rate-adaptation
>> based only on loss vs. what RMCAT is looking to (for RTCWEB), which
>> is based on delay and loss.  Here's the URL.
>> >
>> >
>> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>> >
>> > It's more raw than we had in mind, but we believe this is
>> necessary, based on implementation experience and what users and
>> customers have in their networks, or are planning on having in
>> their networks soon.
>> >
>> > James & Toerless
>> >
>



From michawe@ifi.uio.no  Wed Jul 17 08:59:30 2013
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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+1

On Jul 17, 2013, at 5:23 AM, Mo Zanaty (mzanaty) <mzanaty@cisco.com> =
wrote:

> I support something like what this draft proposes, to segregate RMCAT =
(and potentially other delay-adaptive traffic) from other traffic which =
does not adapt to delay trends (and therefore incurs maximum queue delay =
for all traffic in its queue). Of course, this only helps when such =
queue segregation is available in the network, but effective active =
queue management is not available (otherwise there would be no delay =
signals for delay-adaptive traffic to act on).
>=20
> An open question in my mind, that I think this draft needs to address, =
is whether all delay-adaptive traffic should share the same queue/DSCP, =
or whether we need different queues/DSCPs for different delay-adaptive =
protocols/applications (for example, RMCAT vs. LEDBAT).
>=20
> Mo
>=20
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf =
Of James Polk (jmpolk)
> Sent: Tuesday, July 16, 2013 5:11 PM
> To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert =
(eckert); Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. =
loss-only rate-adaptive for DiffServ
>=20
> CJ
>=20
> Thank you for the quick review.
>=20
> I'll ask you the same question I just asked Michael, the draft isn't=20=

> very long (under 8 pages currently), and it is a work in progress -=20
> but do you have text or just points that this draft needs to cover=20
> that it doesn't currently?
>=20
> James
>=20
> At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>> Same here... I feel it makes sense to create a new DSCP / CS4 for=20
>> transport of video flows that have rate adaptive behaviors and/or=20
>> intra-flow preferential drop priorities as indicated in the I-D. The=20=

>> service provided by the network may thus be considered new, i.e.=20
>> different from what exists in AF4x, so IMHO, using a new or reusing=20=

>> CS4 service class adds clarity for the user applications.
>>=20
>> Regards,
>> CJ
>>=20
>>=20
>> -----Original Message-----
>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On=20
>> Behalf Of Michael Ramalho (mramalho)
>> Sent: Tuesday, July 16, 2013 6:36 AM
>> To: tsvwg@ietf.org
>> Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.=20
>> loss-only rate-adaptive for DiffServ
>>=20
>> TSVWG Members,
>>=20
>> I should have copied you on my email to the RMCAT mailer below.
>>=20
>> I strongly support the creation of a new DSCP for transports in=20
>> which their congestion control can adapt based on delay.
>>=20
>> The current ID in support of this=20
>> =
(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt=20
>> ) is a work in progress, but a step in the right direction.
>>=20
>> It is a given that the ability of the eventual RMCAT adaptation=20
>> mechanisms to achieve low delay will be function of the other=20
>> dominant traffic it is competing against. If the dominant competing=20=

>> traffic DEPENDS on loss, the RMCAT packets are held hostage to being=20=

>> delayed by the bottleneck queue delay maximum.
>>=20
>> Thus, whenever possible, it will be preferable for RMCAT flows to=20
>> compete with other congestion control transports that adapt on=20
>> delay. Even this may not get us to the desired low-delay goals when=20=

>> a portion of the traffic has long RTTs (i.e., adaptation control=20
>> loops that are long in time), or for links that have a highly-time=20
>> varying capacity,  but it will help for a lot of common bottleneck=20
>> topologies (e.g., slowly time-varying access bufferbloat).
>>=20
>> It is my hope that this topic has some discussion time in the Berlin=20=

>> Transport WG (not the specific codepoint to be chosen, but rather=20
>> the need for one).
>>=20
>> Off Soapbox,
>>=20
>> Michael Ramalho, Ph.D.
>>=20
>> -----Original Message-----
>> From: Michael Ramalho (mramalho)
>> Sent: Tuesday, July 16, 2013 9:06 AM
>> To: rmcat@ietf.org
>> Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.=20
>> loss-only rate-adaptive for DiffServ
>>=20
>> RMCAT Design Team,
>>=20
>> The draft Lars references below is a formal request for a DSCP=20
>> dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>>=20
>> It will take a while to become socialized ... and we can progress=20
>> our RMCAT work in the interim.
>>=20
>> Michael Ramalho
>>=20
>> -----Original Message-----
>> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On=20
>> Behalf Of Eggert, Lars
>> Sent: Tuesday, July 16, 2013 5:26 AM
>> To: WG WG
>> Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>> Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only=20
>> rate-adaptive for DiffServ
>>=20
>> Possibly of interest to RMCAT.
>>=20
>> Begin forwarded message:
>>=20
>>> From: James Polk <jmpolk@cisco.com>
>>> Subject: [tsvwg] Submitted ID on delay vs. loss-only=20
>> rate-adaptive for DiffServ
>>> Date: July 16, 2013 8:07:59 GMT+02:00
>>> To: <tsvwg@ietf.org>
>>>=20
>>> (as an author)
>>>=20
>>> Toerless and I put together a draft about legacy rate-adaptation=20
>> based only on loss vs. what RMCAT is looking to (for RTCWEB), which=20=

>> is based on delay and loss.  Here's the URL.
>>>=20
>>>=20
>> =
http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-serv=
ice-classes-00.txt
>>>=20
>>> It's more raw than we had in mind, but we believe this is=20
>> necessary, based on implementation experience and what users and=20
>> customers have in their networks, or are planning on having in=20
>> their networks soon.
>>>=20
>>> James & Toerless
>>>=20
>=20


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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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On 7/16/2013 11:23 PM, Mo Zanaty (mzanaty) wrote:
>
> An open question in my mind, that I think this draft needs to address, is whether all delay-adaptive traffic should share the same queue/DSCP, or whether we need different queues/DSCPs for different delay-adaptive protocols/applications (for example, RMCAT vs. LEDBAT).
> 

Given that everything in the same queue will measure the same
delays, it seems to me like separating flows into queues would
be based on the relative target delays of the flows (tunable
per-flow) and NOT the algorithm used (as you say, RMCAT vs LEDBAT).

It is not wise, in my opinion, to have algorithm-specific DSCPs.
The DSCPs need to indicate the desired forwarding behavior, and
not what version of code the end-host runs.

-- 
Wes Eddy
MTI Systems

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Wesley Eddy <wes@mti-systems.com>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Agreed. I should have clarified in my RMCAT vs. LEDBAT example that the key=
 difference (for queue dynamics) is the target delay of near-zero vs. 100 m=
s. (Near-zero is debatable to mean anything well below 100 ms. I've heard 5=
0, 25, and even 0 mentioned as potential targets in RMCAT.)

Mo


-----Original Message-----
From: Wesley Eddy [mailto:wes@mti-systems.com]=20
Sent: Wednesday, July 17, 2013 2:26 PM
To: Mo Zanaty (mzanaty)
Cc: James Polk (jmpolk); Cheng-Jia Lai (chelai); Toerless Eckert (eckert); =
Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ

On 7/16/2013 11:23 PM, Mo Zanaty (mzanaty) wrote:
>
> An open question in my mind, that I think this draft needs to address, is=
 whether all delay-adaptive traffic should share the same queue/DSCP, or wh=
ether we need different queues/DSCPs for different delay-adaptive protocols=
/applications (for example, RMCAT vs. LEDBAT).
>=20

Given that everything in the same queue will measure the same
delays, it seems to me like separating flows into queues would
be based on the relative target delays of the flows (tunable
per-flow) and NOT the algorithm used (as you say, RMCAT vs LEDBAT).

It is not wise, in my opinion, to have algorithm-specific DSCPs.
The DSCPs need to indicate the desired forwarding behavior, and
not what version of code the end-host runs.

--=20
Wes Eddy
MTI Systems

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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 17/07/2013 18:50, gorry@erg.abdn.ac.uk wrote:
> I think the question of sharing diffserv is important, and my current
> suggestion is that RMCAT should use one or at most two DSCPs and design a
> CC that works well within the class - however this does not absolve RMCAT
> from also having to work in the Internet without this class. This will
> likely be the main use-case!
> 
> Although the proposal could help these applications - the IETF can not
> somehow mandate this is universally deployed - and even if the DSCP(s)
> were universally recognised, it would still be aggregated within some
> networks and on L2 technologies that support fewer classes than DSCP
> markings.
> 
> It would be interesting to know if others think we should be considering
> more DSCPs for this?

If you mean "more" in the sense of defining new PHBs, I sincerely hope
not. Diffserv works best with a small set of highly differentiated
PHBs. Also, in order to get cross-domain agreements on DSCP assignments,
the fewer there are the better.

    Brian

> 
> Gorry
> 
>> I support something like what this draft proposes, to segregate RMCAT (and
>> potentially other delay-adaptive traffic) from other traffic which does
>> not adapt to delay trends (and therefore incurs maximum queue delay for
>> all traffic in its queue). Of course, this only helps when such queue
>> segregation is available in the network, but effective active queue
>> management is not available (otherwise there would be no delay signals for
>> delay-adaptive traffic to act on).
>>
>> An open question in my mind, that I think this draft needs to address, is
>> whether all delay-adaptive traffic should share the same queue/DSCP, or
>> whether we need different queues/DSCPs for different delay-adaptive
>> protocols/applications (for example, RMCAT vs. LEDBAT).
>>
>> Mo
>>
>> -----Original Message-----
>> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
>> James Polk (jmpolk)
>> Sent: Tuesday, July 16, 2013 5:11 PM
>> To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert (eckert);
>> Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
>> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only
>> rate-adaptive for DiffServ
>>
>> CJ
>>
>> Thank you for the quick review.
>>
>> I'll ask you the same question I just asked Michael, the draft isn't
>> very long (under 8 pages currently), and it is a work in progress -
>> but do you have text or just points that this draft needs to cover
>> that it doesn't currently?
>>
>> James
>>
>> At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>>> Same here... I feel it makes sense to create a new DSCP / CS4 for
>>> transport of video flows that have rate adaptive behaviors and/or
>>> intra-flow preferential drop priorities as indicated in the I-D. The
>>> service provided by the network may thus be considered new, i.e.
>>> different from what exists in AF4x, so IMHO, using a new or reusing
>>> CS4 service class adds clarity for the user applications.
>>>
>>> Regards,
>>> CJ
>>>
>>>
>>> -----Original Message-----
>>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
>>> Behalf Of Michael Ramalho (mramalho)
>>> Sent: Tuesday, July 16, 2013 6:36 AM
>>> To: tsvwg@ietf.org
>>> Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.
>>> loss-only rate-adaptive for DiffServ
>>>
>>> TSVWG Members,
>>>
>>> I should have copied you on my email to the RMCAT mailer below.
>>>
>>> I strongly support the creation of a new DSCP for transports in
>>> which their congestion control can adapt based on delay.
>>>
>>> The current ID in support of this
>>> (http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>>> ) is a work in progress, but a step in the right direction.
>>>
>>> It is a given that the ability of the eventual RMCAT adaptation
>>> mechanisms to achieve low delay will be function of the other
>>> dominant traffic it is competing against. If the dominant competing
>>> traffic DEPENDS on loss, the RMCAT packets are held hostage to being
>>> delayed by the bottleneck queue delay maximum.
>>>
>>> Thus, whenever possible, it will be preferable for RMCAT flows to
>>> compete with other congestion control transports that adapt on
>>> delay. Even this may not get us to the desired low-delay goals when
>>> a portion of the traffic has long RTTs (i.e., adaptation control
>>> loops that are long in time), or for links that have a highly-time
>>> varying capacity,  but it will help for a lot of common bottleneck
>>> topologies (e.g., slowly time-varying access bufferbloat).
>>>
>>> It is my hope that this topic has some discussion time in the Berlin
>>> Transport WG (not the specific codepoint to be chosen, but rather
>>> the need for one).
>>>
>>> Off Soapbox,
>>>
>>> Michael Ramalho, Ph.D.
>>>
>>> -----Original Message-----
>>> From: Michael Ramalho (mramalho)
>>> Sent: Tuesday, July 16, 2013 9:06 AM
>>> To: rmcat@ietf.org
>>> Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.
>>> loss-only rate-adaptive for DiffServ
>>>
>>> RMCAT Design Team,
>>>
>>> The draft Lars references below is a formal request for a DSCP
>>> dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>>>
>>> It will take a while to become socialized ... and we can progress
>>> our RMCAT work in the interim.
>>>
>>> Michael Ramalho
>>>
>>> -----Original Message-----
>>> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
>>> Behalf Of Eggert, Lars
>>> Sent: Tuesday, July 16, 2013 5:26 AM
>>> To: WG WG
>>> Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>>> Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only
>>> rate-adaptive for DiffServ
>>>
>>> Possibly of interest to RMCAT.
>>>
>>> Begin forwarded message:
>>>
>>>> From: James Polk <jmpolk@cisco.com>
>>>> Subject: [tsvwg] Submitted ID on delay vs. loss-only
>>> rate-adaptive for DiffServ
>>>> Date: July 16, 2013 8:07:59 GMT+02:00
>>>> To: <tsvwg@ietf.org>
>>>>
>>>> (as an author)
>>>>
>>>> Toerless and I put together a draft about legacy rate-adaptation
>>> based only on loss vs. what RMCAT is looking to (for RTCWEB), which
>>> is based on delay and loss.  Here's the URL.
>>>>
>>> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>>>> It's more raw than we had in mind, but we believe this is
>>> necessary, based on implementation experience and what users and
>>> customers have in their networks, or are planning on having in
>>> their networks soon.
>>>> James & Toerless
>>>>
> 
> 
> 

From chelai@cisco.com  Wed Jul 17 13:22:17 2013
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From: "Cheng-Jia Lai (chelai)" <chelai@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>, "Toerless Eckert (eckert)" <eckert@cisco.com>, "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay  vs.	loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOgmkJVhIAra2Os0yVivAo05Amo5lpST9g
Date: Wed, 17 Jul 2013 20:22:08 +0000
Message-ID: <A860EC86B79FA646BF3F89165A8862641533A4E0@xmb-aln-x11.cisco.com>
References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com>
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Subject: Re: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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I think it would be good if Section 5.1 can include some specification or r=
ecommendation how a video flow's packets should be classified into CS4 and =
CS4-Discardable and how they will be dropped preferentially at congestion. =
Although AF4x has already specified how to do this by different drop probab=
ilities, it might not be suitable since I-frame, IDR, or other important pa=
ckets (e.g. in-band signals or supplementary metadata) can still be dropped=
 by probability (smaller though) before all discardable P frames in the sam=
e video flow are dropped.

In addition, maybe it should cover some fairness requirements, i.e. how the=
 video rate adaptive behaviors can expect to see packet drops w.r.t. conten=
tion, which IMHO should encourage video endpoints to generate more discarda=
ble P-frame or other lower priority packets, even at congestion. If possibl=
e, this should also try to avoid global synchronization, i.e. all video-enc=
oders to generate I-frame/IDR at the same time (even after they reduce rate=
, they still need to recover the lost video) when CS4 / CS4-Discardable que=
ue is congested and drops packets from all of them.

Regards,
CJ


-----Original Message-----
From: James Polk (jmpolk)=20
Sent: Tuesday, July 16, 2013 2:11 PM
To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert (eckert); =
Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
Subject: RE: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ

CJ

Thank you for the quick review.

I'll ask you the same question I just asked Michael, the draft isn't=20
very long (under 8 pages currently), and it is a work in progress -=20
but do you have text or just points that this draft needs to cover=20
that it doesn't currently?

James

At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>Same here... I feel it makes sense to create a new DSCP / CS4 for=20
>transport of video flows that have rate adaptive behaviors and/or=20
>intra-flow preferential drop priorities as indicated in the I-D. The=20
>service provided by the network may thus be considered new, i.e.=20
>different from what exists in AF4x, so IMHO, using a new or reusing=20
>CS4 service class adds clarity for the user applications.
>
>Regards,
>CJ
>
>
>-----Original Message-----
>From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On=20
>Behalf Of Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 6:36 AM
>To: tsvwg@ietf.org
>Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.=20
>loss-only rate-adaptive for DiffServ
>
>TSVWG Members,
>
>I should have copied you on my email to the RMCAT mailer below.
>
>I strongly support the creation of a new DSCP for transports in=20
>which their congestion control can adapt based on delay.
>
>The current ID in support of this=20
>(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt=20
>) is a work in progress, but a step in the right direction.
>
>It is a given that the ability of the eventual RMCAT adaptation=20
>mechanisms to achieve low delay will be function of the other=20
>dominant traffic it is competing against. If the dominant competing=20
>traffic DEPENDS on loss, the RMCAT packets are held hostage to being=20
>delayed by the bottleneck queue delay maximum.
>
>Thus, whenever possible, it will be preferable for RMCAT flows to=20
>compete with other congestion control transports that adapt on=20
>delay. Even this may not get us to the desired low-delay goals when=20
>a portion of the traffic has long RTTs (i.e., adaptation control=20
>loops that are long in time), or for links that have a highly-time=20
>varying capacity,  but it will help for a lot of common bottleneck=20
>topologies (e.g., slowly time-varying access bufferbloat).
>
>It is my hope that this topic has some discussion time in the Berlin=20
>Transport WG (not the specific codepoint to be chosen, but rather=20
>the need for one).
>
>Off Soapbox,
>
>Michael Ramalho, Ph.D.
>
>-----Original Message-----
>From: Michael Ramalho (mramalho)
>Sent: Tuesday, July 16, 2013 9:06 AM
>To: rmcat@ietf.org
>Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.=20
>loss-only rate-adaptive for DiffServ
>
>RMCAT Design Team,
>
>The draft Lars references below is a formal request for a DSCP=20
>dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>
>It will take a while to become socialized ... and we can progress=20
>our RMCAT work in the interim.
>
>Michael Ramalho
>
>-----Original Message-----
>From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On=20
>Behalf Of Eggert, Lars
>Sent: Tuesday, July 16, 2013 5:26 AM
>To: WG WG
>Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only=20
>rate-adaptive for DiffServ
>
>Possibly of interest to RMCAT.
>
>Begin forwarded message:
>
> > From: James Polk <jmpolk@cisco.com>
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only=20
> rate-adaptive for DiffServ
> > Date: July 16, 2013 8:07:59 GMT+02:00
> > To: <tsvwg@ietf.org>
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation=20
> based only on loss vs. what RMCAT is looking to (for RTCWEB), which=20
> is based on delay and loss.  Here's the URL.
> >
> >=20
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-ser=
vice-classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is=20
> necessary, based on implementation experience and what users and=20
> customers have in their networks, or are planning on having in=20
> their networks soon.
> >
> > James & Toerless
> >


From jmpolk@cisco.com  Wed Jul 17 13:37:39 2013
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From: James Polk <jmpolk@cisco.com>
In-Reply-To: <A860EC86B79FA646BF3F89165A8862641533A4E0@xmb-aln-x11.cisco .com>
References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <A860EC86B79FA646BF3F89165A8862641533A4E0@xmb-aln-x11.cisco.com>
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Subject: Re: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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CJ

These are good suggestions for the next version. I thank you for 
offering them. I agree that s5.1 is underspecified, and will be 
expanded next version along the lines of your 1st paragraph below, as well.

James

At 03:22 PM 7/17/2013, Cheng-Jia Lai (chelai) wrote:
>I think it would be good if Section 5.1 can include some 
>specification or recommendation how a video flow's packets should be 
>classified into CS4 and CS4-Discardable and how they will be dropped 
>preferentially at congestion. Although AF4x has already specified 
>how to do this by different drop probabilities, it might not be 
>suitable since I-frame, IDR, or other important packets (e.g. 
>in-band signals or supplementary metadata) can still be dropped by 
>probability (smaller though) before all discardable P frames in the 
>same video flow are dropped.
>
>In addition, maybe it should cover some fairness requirements, i.e. 
>how the video rate adaptive behaviors can expect to see packet drops 
>w.r.t. contention, which IMHO should encourage video endpoints to 
>generate more discardable P-frame or other lower priority packets, 
>even at congestion. If possible, this should also try to avoid 
>global synchronization, i.e. all video-encoders to generate 
>I-frame/IDR at the same time (even after they reduce rate, they 
>still need to recover the lost video) when CS4 / CS4-Discardable 
>queue is congested and drops packets from all of them.
>
>Regards,
>CJ
>
>
>-----Original Message-----
>From: James Polk (jmpolk)
>Sent: Tuesday, July 16, 2013 2:11 PM
>To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert 
>(eckert); Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
>Subject: RE: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs. 
>loss-only rate-adaptive for DiffServ
>
>CJ
>
>Thank you for the quick review.
>
>I'll ask you the same question I just asked Michael, the draft isn't
>very long (under 8 pages currently), and it is a work in progress -
>but do you have text or just points that this draft needs to cover
>that it doesn't currently?
>
>James
>
>At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
> >Same here... I feel it makes sense to create a new DSCP / CS4 for
> >transport of video flows that have rate adaptive behaviors and/or
> >intra-flow preferential drop priorities as indicated in the I-D. The
> >service provided by the network may thus be considered new, i.e.
> >different from what exists in AF4x, so IMHO, using a new or reusing
> >CS4 service class adds clarity for the user applications.
> >
> >Regards,
> >CJ
> >
> >
> >-----Original Message-----
> >From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
> >Behalf Of Michael Ramalho (mramalho)
> >Sent: Tuesday, July 16, 2013 6:36 AM
> >To: tsvwg@ietf.org
> >Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.
> >loss-only rate-adaptive for DiffServ
> >
> >TSVWG Members,
> >
> >I should have copied you on my email to the RMCAT mailer below.
> >
> >I strongly support the creation of a new DSCP for transports in
> >which their congestion control can adapt based on delay.
> >
> >The current ID in support of this
> >(http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss 
> -ds-service-classes-00.txt
> >) is a work in progress, but a step in the right direction.
> >
> >It is a given that the ability of the eventual RMCAT adaptation
> >mechanisms to achieve low delay will be function of the other
> >dominant traffic it is competing against. If the dominant competing
> >traffic DEPENDS on loss, the RMCAT packets are held hostage to being
> >delayed by the bottleneck queue delay maximum.
> >
> >Thus, whenever possible, it will be preferable for RMCAT flows to
> >compete with other congestion control transports that adapt on
> >delay. Even this may not get us to the desired low-delay goals when
> >a portion of the traffic has long RTTs (i.e., adaptation control
> >loops that are long in time), or for links that have a highly-time
> >varying capacity,  but it will help for a lot of common bottleneck
> >topologies (e.g., slowly time-varying access bufferbloat).
> >
> >It is my hope that this topic has some discussion time in the Berlin
> >Transport WG (not the specific codepoint to be chosen, but rather
> >the need for one).
> >
> >Off Soapbox,
> >
> >Michael Ramalho, Ph.D.
> >
> >-----Original Message-----
> >From: Michael Ramalho (mramalho)
> >Sent: Tuesday, July 16, 2013 9:06 AM
> >To: rmcat@ietf.org
> >Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.
> >loss-only rate-adaptive for DiffServ
> >
> >RMCAT Design Team,
> >
> >The draft Lars references below is a formal request for a DSCP
> >dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
> >
> >It will take a while to become socialized ... and we can progress
> >our RMCAT work in the interim.
> >
> >Michael Ramalho
> >
> >-----Original Message-----
> >From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
> >Behalf Of Eggert, Lars
> >Sent: Tuesday, July 16, 2013 5:26 AM
> >To: WG WG
> >Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
> >Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only
> >rate-adaptive for DiffServ
> >
> >Possibly of interest to RMCAT.
> >
> >Begin forwarded message:
> >
> > > From: James Polk <jmpolk@cisco.com>
> > > Subject: [tsvwg] Submitted ID on delay vs. loss-only
> > rate-adaptive for DiffServ
> > > Date: July 16, 2013 8:07:59 GMT+02:00
> > > To: <tsvwg@ietf.org>
> > >
> > > (as an author)
> > >
> > > Toerless and I put together a draft about legacy rate-adaptation
> > based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> > is based on delay and loss.  Here's the URL.
> > >
> > >
> > 
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
> > >
> > > It's more raw than we had in mind, but we believe this is
> > necessary, based on implementation experience and what users and
> > customers have in their networks, or are planning on having in
> > their networks soon.
> > >
> > > James & Toerless
> > >


From jmpolk@cisco.com  Wed Jul 17 13:46:06 2013
Return-Path: <jmpolk@cisco.com>
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To: Brian E Carpenter <brian.e.carpenter@gmail.com>, <gorry@erg.abdn.ac.uk>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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At 02:55 PM 7/17/2013, Brian E Carpenter wrote:
>On 17/07/2013 18:50, gorry@erg.abdn.ac.uk wrote:
> > I think the question of sharing diffserv is important, and my current
> > suggestion is that RMCAT should use one or at most two DSCPs and design a
> > CC that works well within the class - however this does not absolve RMCAT
> > from also having to work in the Internet without this class. This will
> > likely be the main use-case!
> >
> > Although the proposal could help these applications - the IETF can not
> > somehow mandate this is universally deployed - and even if the DSCP(s)
> > were universally recognised, it would still be aggregated within some
> > networks and on L2 technologies that support fewer classes than DSCP
> > markings.
> >
> > It would be interesting to know if others think we should be considering
> > more DSCPs for this?
>
>If you mean "more" in the sense of defining new PHBs, I sincerely hope
>not. Diffserv works best with a small set of highly differentiated
>PHBs. Also, in order to get cross-domain agreements on DSCP assignments,
>the fewer there are the better.

Brian - while I agree less is more manageable, but less what? Isn't 
'the less' or "... a small set..." you're referring to actually about 
treatment aggregates, defined by RFC 5127 (where similar PHBs are 
aggregated into 'treatment aggregates' to all be treated the same 
where we have big pipes)? Treatment Aggregates would see all of the 
above video traffic in the same individual treatment aggregate, 
solving the issue you bring up that _not_ any more treatment 
aggregates are defined.

James


>     Brian
>
> >
> > Gorry
> >
> >> I support something like what this draft proposes, to segregate RMCAT (and
> >> potentially other delay-adaptive traffic) from other traffic which does
> >> not adapt to delay trends (and therefore incurs maximum queue delay for
> >> all traffic in its queue). Of course, this only helps when such queue
> >> segregation is available in the network, but effective active queue
> >> management is not available (otherwise there would be no delay signals for
> >> delay-adaptive traffic to act on).
> >>
> >> An open question in my mind, that I think this draft needs to address, is
> >> whether all delay-adaptive traffic should share the same queue/DSCP, or
> >> whether we need different queues/DSCPs for different delay-adaptive
> >> protocols/applications (for example, RMCAT vs. LEDBAT).
> >>
> >> Mo
> >>
> >> -----Original Message-----
> >> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
> >> James Polk (jmpolk)
> >> Sent: Tuesday, July 16, 2013 5:11 PM
> >> To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert (eckert);
> >> Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
> >> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only
> >> rate-adaptive for DiffServ
> >>
> >> CJ
> >>
> >> Thank you for the quick review.
> >>
> >> I'll ask you the same question I just asked Michael, the draft isn't
> >> very long (under 8 pages currently), and it is a work in progress -
> >> but do you have text or just points that this draft needs to cover
> >> that it doesn't currently?
> >>
> >> James
> >>
> >> At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
> >>> Same here... I feel it makes sense to create a new DSCP / CS4 for
> >>> transport of video flows that have rate adaptive behaviors and/or
> >>> intra-flow preferential drop priorities as indicated in the I-D. The
> >>> service provided by the network may thus be considered new, i.e.
> >>> different from what exists in AF4x, so IMHO, using a new or reusing
> >>> CS4 service class adds clarity for the user applications.
> >>>
> >>> Regards,
> >>> CJ
> >>>
> >>>
> >>> -----Original Message-----
> >>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
> >>> Behalf Of Michael Ramalho (mramalho)
> >>> Sent: Tuesday, July 16, 2013 6:36 AM
> >>> To: tsvwg@ietf.org
> >>> Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.
> >>> loss-only rate-adaptive for DiffServ
> >>>
> >>> TSVWG Members,
> >>>
> >>> I should have copied you on my email to the RMCAT mailer below.
> >>>
> >>> I strongly support the creation of a new DSCP for transports in
> >>> which their congestion control can adapt based on delay.
> >>>
> >>> The current ID in support of this
> >>> 
> (http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
> >>> ) is a work in progress, but a step in the right direction.
> >>>
> >>> It is a given that the ability of the eventual RMCAT adaptation
> >>> mechanisms to achieve low delay will be function of the other
> >>> dominant traffic it is competing against. If the dominant competing
> >>> traffic DEPENDS on loss, the RMCAT packets are held hostage to being
> >>> delayed by the bottleneck queue delay maximum.
> >>>
> >>> Thus, whenever possible, it will be preferable for RMCAT flows to
> >>> compete with other congestion control transports that adapt on
> >>> delay. Even this may not get us to the desired low-delay goals when
> >>> a portion of the traffic has long RTTs (i.e., adaptation control
> >>> loops that are long in time), or for links that have a highly-time
> >>> varying capacity,  but it will help for a lot of common bottleneck
> >>> topologies (e.g., slowly time-varying access bufferbloat).
> >>>
> >>> It is my hope that this topic has some discussion time in the Berlin
> >>> Transport WG (not the specific codepoint to be chosen, but rather
> >>> the need for one).
> >>>
> >>> Off Soapbox,
> >>>
> >>> Michael Ramalho, Ph.D.
> >>>
> >>> -----Original Message-----
> >>> From: Michael Ramalho (mramalho)
> >>> Sent: Tuesday, July 16, 2013 9:06 AM
> >>> To: rmcat@ietf.org
> >>> Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.
> >>> loss-only rate-adaptive for DiffServ
> >>>
> >>> RMCAT Design Team,
> >>>
> >>> The draft Lars references below is a formal request for a DSCP
> >>> dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
> >>>
> >>> It will take a while to become socialized ... and we can progress
> >>> our RMCAT work in the interim.
> >>>
> >>> Michael Ramalho
> >>>
> >>> -----Original Message-----
> >>> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
> >>> Behalf Of Eggert, Lars
> >>> Sent: Tuesday, July 16, 2013 5:26 AM
> >>> To: WG WG
> >>> Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
> >>> Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only
> >>> rate-adaptive for DiffServ
> >>>
> >>> Possibly of interest to RMCAT.
> >>>
> >>> Begin forwarded message:
> >>>
> >>>> From: James Polk <jmpolk@cisco.com>
> >>>> Subject: [tsvwg] Submitted ID on delay vs. loss-only
> >>> rate-adaptive for DiffServ
> >>>> Date: July 16, 2013 8:07:59 GMT+02:00
> >>>> To: <tsvwg@ietf.org>
> >>>>
> >>>> (as an author)
> >>>>
> >>>> Toerless and I put together a draft about legacy rate-adaptation
> >>> based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> >>> is based on delay and loss.  Here's the URL.
> >>>>
> >>> 
> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
> >>>> It's more raw than we had in mind, but we believe this is
> >>> necessary, based on implementation experience and what users and
> >>> customers have in their networks, or are planning on having in
> >>> their networks soon.
> >>>> James & Toerless
> >>>>
> >
> >
> >


From david.black@emc.com  Wed Jul 17 14:37:48 2013
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From: "Black, David" <david.black@emc.com>
To: tsvwg WG <tsvwg@ietf.org>
Date: Wed, 17 Jul 2013 17:37:21 -0400
Thread-Topic: Delay vs. Loss draft: Terminology
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Subject: [tsvwg] Delay vs. Loss draft: Terminology
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<WG chair hat off>

This note concerns draft-polk-tsvwg-delay-vs-loss-ds-service-classes

I believe I see some sloppy use of terminology in this draft and the
related discussion.

CS4 was originally defined as a DSCP value in RFC 2474 (Diffserv field
in IP headers) and is used only as a DSCP value in RFC 4594 (Diffserv
Service classes).

That said, RFC 2474 can be read to allow use of CS4 as a PHB name - it's
effectively shorthand for "the PHB assigned to the CS4 DSCP - as is done
in draft-polk-tsvwg-rfc4594-update does.  While that may not be the best
usage, I think it's ok.

OTOH, draft-polk-tsvwg-delay-vs-loss-ds-service-classes appears to be
using strings such as "CS4", "CS4-Discardable" and "CS4+" as service
class names (see Sections 5 and 6).  That inclusion of the DSCP name
in the service class name is (IMHO), a stunningly bad idea that is
going to lead to confusion, and moreover is at odds with one of the basic
principles of the Diffserv architecture, namely separation of traffic
behavior (PHB and subsequently, service class) from the DSCP used in
the IP header to request suitable treatment in a particular network.

As an author of both RFC 2474 and 2475, I really would like to
see draft-polk-tsvwg-delay-vs-loss-ds-service-classes revised to be
clear on DSCP vs. PHB vs. service class and in particular to cease
and desist from using CS4 (or any other DSCP name) as the name or part
of the name of a service class - that will inevitably lead to the bad
and wrong assumption that such a service class can only ever be
deployed with the CS4 DSCP - while that sort of relationship is
RECOMMENDED by RFC 4594, it is not REQUIRED (where "RECOMMENNDED"
and "REQUIRED" have their RFC 2119 meanings).

An example of the possible confusion that is that there will be
networks in which the CS4 DSCP will be deployed with only one
of the service classes currently called "CS4" and "CS4-Discardable" in
draft-polk-tsvwg-delay-vs-loss-ds-service-classes.  Is anyone prepared
to try to explain with a straight face why the "CS4" service class
isn't using the "CS4" DSCP?

Thanks,
--David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
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From david.black@emc.com  Wed Jul 17 15:57:50 2013
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From: "Black, David" <david.black@emc.com>
To: RFC Errata System <rfc-editor@rfc-editor.org>, "kk@teraoptic.com" <kk@teraoptic.com>, "floyd@aciri.org" <floyd@aciri.org>, "spencerdawkins.ietf@gmail.com" <spencerdawkins.ietf@gmail.com>, "martin.stiemerling@neclab.eu" <martin.stiemerling@neclab.eu>, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "jmpolk@cisco.com" <jmpolk@cisco.com>
Date: Wed, 17 Jul 2013 18:56:20 -0400
Thread-Topic: [Technical Errata Reported] RFC3168 (3680)
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Subject: Re: [tsvwg] [Technical Errata Reported] RFC3168 (3680)
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I believe this errata is correct.

Thanks,
--David (as RFC 3168 author and tsvwg WG co-chair)

> -----Original Message-----
> From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]
> Sent: Friday, July 12, 2013 7:15 AM
> To: kk@teraoptic.com; floyd@aciri.org; Black, David;
> spencerdawkins.ietf@gmail.com; martin.stiemerling@neclab.eu;
> gorry@erg.abdn.ac.uk; Black, David; jmpolk@cisco.com
> Cc: mirja.kuehlewind@ikr.uni-stuttgart.de; tsvwg@ietf.org; rfc-editor@rfc=
-
> editor.org
> Subject: [Technical Errata Reported] RFC3168 (3680)
>=20
> The following errata report has been submitted for RFC3168,
> "The Addition of Explicit Congestion Notification (ECN) to IP".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D3168&eid=3D3680
>=20
> --------------------------------------
> Type: Technical
> Reported by: Mirja K=FChlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
>=20
> Section: 6.1.1.
>=20
> Original Text
> -------------
> If the TCP connection does not
> wish to use ECN notification for a particular packet, the sending TCP
> sets the ECN codepoint to not-ECT, and the TCP receiver ignores the
> CE codepoint in the received packet.
>=20
> Corrected Text
> --------------
> If the TCP connection does not
> wish to use ECN notification for a particular packet, the sending TCP
> sets the ECN codepoint to not-ECT.
>=20
> Notes
> -----
> The receiver should not ignore any CE codepoint.
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>=20
> --------------------------------------
> RFC3168 (draft-ietf-tsvwg-ecn-04)
> --------------------------------------
> Title               : The Addition of Explicit Congestion Notification (E=
CN)
> to IP
> Publication Date    : September 2001
> Author(s)           : K. Ramakrishnan, S. Floyd, D. Black
> Category            : PROPOSED STANDARD
> Source              : Transport Area Working Group
> Area                : Transport
> Stream              : IETF
> Verifying Party     : IESG


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Date: Thu, 18 Jul 2013 11:17:24 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Cc: gorry@erg.abdn.ac.uk, "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 18/07/2013 08:46, James Polk wrote:
> At 02:55 PM 7/17/2013, Brian E Carpenter wrote:
>> On 17/07/2013 18:50, gorry@erg.abdn.ac.uk wrote:
>> > I think the question of sharing diffserv is important, and my current
>> > suggestion is that RMCAT should use one or at most two DSCPs and
>> design a
>> > CC that works well within the class - however this does not absolve
>> RMCAT
>> > from also having to work in the Internet without this class. This will
>> > likely be the main use-case!
>> >
>> > Although the proposal could help these applications - the IETF can not
>> > somehow mandate this is universally deployed - and even if the DSCP(s)
>> > were universally recognised, it would still be aggregated within some
>> > networks and on L2 technologies that support fewer classes than DSCP
>> > markings.
>> >
>> > It would be interesting to know if others think we should be
>> considering
>> > more DSCPs for this?
>>
>> If you mean "more" in the sense of defining new PHBs, I sincerely hope
>> not. Diffserv works best with a small set of highly differentiated
>> PHBs. Also, in order to get cross-domain agreements on DSCP assignments,
>> the fewer there are the better.
> 
> Brian - while I agree less is more manageable, but less what? Isn't 'the
> less' or "... a small set..." you're referring to actually about
> treatment aggregates, defined by RFC 5127 (where similar PHBs are
> aggregated into 'treatment aggregates' to all be treated the same where
> we have big pipes)? Treatment Aggregates would see all of the above
> video traffic in the same individual treatment aggregate, solving the
> issue you bring up that _not_ any more treatment aggregates are defined.

Yes, I agree entirely.

   Brian

> 
> James
> 
> 
>>     Brian
>>
>> >
>> > Gorry
>> >
>> >> I support something like what this draft proposes, to segregate
>> RMCAT (and
>> >> potentially other delay-adaptive traffic) from other traffic which
>> does
>> >> not adapt to delay trends (and therefore incurs maximum queue delay
>> for
>> >> all traffic in its queue). Of course, this only helps when such queue
>> >> segregation is available in the network, but effective active queue
>> >> management is not available (otherwise there would be no delay
>> signals for
>> >> delay-adaptive traffic to act on).
>> >>
>> >> An open question in my mind, that I think this draft needs to
>> address, is
>> >> whether all delay-adaptive traffic should share the same
>> queue/DSCP, or
>> >> whether we need different queues/DSCPs for different delay-adaptive
>> >> protocols/applications (for example, RMCAT vs. LEDBAT).
>> >>
>> >> Mo
>> >>
>> >> -----Original Message-----
>> >> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
>> Behalf Of
>> >> James Polk (jmpolk)
>> >> Sent: Tuesday, July 16, 2013 5:11 PM
>> >> To: Cheng-Jia Lai (chelai); James Polk (jmpolk); Toerless Eckert
>> (eckert);
>> >> Michael Ramalho (mramalho); tsvwg@ietf.org; rmcat@ietf.org
>> >> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.
>> loss-only
>> >> rate-adaptive for DiffServ
>> >>
>> >> CJ
>> >>
>> >> Thank you for the quick review.
>> >>
>> >> I'll ask you the same question I just asked Michael, the draft isn't
>> >> very long (under 8 pages currently), and it is a work in progress -
>> >> but do you have text or just points that this draft needs to cover
>> >> that it doesn't currently?
>> >>
>> >> James
>> >>
>> >> At 01:36 PM 7/16/2013, Cheng-Jia Lai (chelai) wrote:
>> >>> Same here... I feel it makes sense to create a new DSCP / CS4 for
>> >>> transport of video flows that have rate adaptive behaviors and/or
>> >>> intra-flow preferential drop priorities as indicated in the I-D. The
>> >>> service provided by the network may thus be considered new, i.e.
>> >>> different from what exists in AF4x, so IMHO, using a new or reusing
>> >>> CS4 service class adds clarity for the user applications.
>> >>>
>> >>> Regards,
>> >>> CJ
>> >>>
>> >>>
>> >>> -----Original Message-----
>> >>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
>> >>> Behalf Of Michael Ramalho (mramalho)
>> >>> Sent: Tuesday, July 16, 2013 6:36 AM
>> >>> To: tsvwg@ietf.org
>> >>> Subject: [tsvwg] FW: [rmcat] Fwd: Submitted ID on delay vs.
>> >>> loss-only rate-adaptive for DiffServ
>> >>>
>> >>> TSVWG Members,
>> >>>
>> >>> I should have copied you on my email to the RMCAT mailer below.
>> >>>
>> >>> I strongly support the creation of a new DSCP for transports in
>> >>> which their congestion control can adapt based on delay.
>> >>>
>> >>> The current ID in support of this
>> >>>
>> (http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>>
>> >>> ) is a work in progress, but a step in the right direction.
>> >>>
>> >>> It is a given that the ability of the eventual RMCAT adaptation
>> >>> mechanisms to achieve low delay will be function of the other
>> >>> dominant traffic it is competing against. If the dominant competing
>> >>> traffic DEPENDS on loss, the RMCAT packets are held hostage to being
>> >>> delayed by the bottleneck queue delay maximum.
>> >>>
>> >>> Thus, whenever possible, it will be preferable for RMCAT flows to
>> >>> compete with other congestion control transports that adapt on
>> >>> delay. Even this may not get us to the desired low-delay goals when
>> >>> a portion of the traffic has long RTTs (i.e., adaptation control
>> >>> loops that are long in time), or for links that have a highly-time
>> >>> varying capacity,  but it will help for a lot of common bottleneck
>> >>> topologies (e.g., slowly time-varying access bufferbloat).
>> >>>
>> >>> It is my hope that this topic has some discussion time in the Berlin
>> >>> Transport WG (not the specific codepoint to be chosen, but rather
>> >>> the need for one).
>> >>>
>> >>> Off Soapbox,
>> >>>
>> >>> Michael Ramalho, Ph.D.
>> >>>
>> >>> -----Original Message-----
>> >>> From: Michael Ramalho (mramalho)
>> >>> Sent: Tuesday, July 16, 2013 9:06 AM
>> >>> To: rmcat@ietf.org
>> >>> Subject: RE: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs.
>> >>> loss-only rate-adaptive for DiffServ
>> >>>
>> >>> RMCAT Design Team,
>> >>>
>> >>> The draft Lars references below is a formal request for a DSCP
>> >>> dedicated to "RMCAT-only (or other nice delay-based cc) traffic".
>> >>>
>> >>> It will take a while to become socialized ... and we can progress
>> >>> our RMCAT work in the interim.
>> >>>
>> >>> Michael Ramalho
>> >>>
>> >>> -----Original Message-----
>> >>> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On
>> >>> Behalf Of Eggert, Lars
>> >>> Sent: Tuesday, July 16, 2013 5:26 AM
>> >>> To: WG WG
>> >>> Cc: draft-polk-tsvwg-delay-vs-loss-ds-service-classes@tools.ietf.org
>> >>> Subject: [rmcat] Fwd: [tsvwg] Submitted ID on delay vs. loss-only
>> >>> rate-adaptive for DiffServ
>> >>>
>> >>> Possibly of interest to RMCAT.
>> >>>
>> >>> Begin forwarded message:
>> >>>
>> >>>> From: James Polk <jmpolk@cisco.com>
>> >>>> Subject: [tsvwg] Submitted ID on delay vs. loss-only
>> >>> rate-adaptive for DiffServ
>> >>>> Date: July 16, 2013 8:07:59 GMT+02:00
>> >>>> To: <tsvwg@ietf.org>
>> >>>>
>> >>>> (as an author)
>> >>>>
>> >>>> Toerless and I put together a draft about legacy rate-adaptation
>> >>> based only on loss vs. what RMCAT is looking to (for RTCWEB), which
>> >>> is based on delay and loss.  Here's the URL.
>> >>>>
>> >>>
>> http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-service-classes-00.txt
>>
>> >>>> It's more raw than we had in mind, but we believe this is
>> >>> necessary, based on implementation experience and what users and
>> >>> customers have in their networks, or are planning on having in
>> >>> their networks soon.
>> >>>> James & Toerless
>> >>>>
>> >
>> >
>> >
> 
> 

From brian.e.carpenter@gmail.com  Wed Jul 17 16:35:47 2013
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Subject: Re: [tsvwg] Delay vs. Loss draft: Terminology
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On 18/07/2013 09:37, Black, David wrote:
> <WG chair hat off>
> 
> This note concerns draft-polk-tsvwg-delay-vs-loss-ds-service-classes
> 
> I believe I see some sloppy use of terminology in this draft and the
> related discussion.
> 
> CS4 was originally defined as a DSCP value in RFC 2474 (Diffserv field
> in IP headers) and is used only as a DSCP value in RFC 4594 (Diffserv
> Service classes).
> 
> That said, RFC 2474 can be read to allow use of CS4 as a PHB name - it's
> effectively shorthand for "the PHB assigned to the CS4 DSCP - as is done
> in draft-polk-tsvwg-rfc4594-update does.  While that may not be the best
> usage, I think it's ok.
> 
> OTOH, draft-polk-tsvwg-delay-vs-loss-ds-service-classes appears to be
> using strings such as "CS4", "CS4-Discardable" and "CS4+" as service
> class names (see Sections 5 and 6).  That inclusion of the DSCP name
> in the service class name is (IMHO), a stunningly bad idea that is
> going to lead to confusion, and moreover is at odds with one of the basic
> principles of the Diffserv architecture, namely separation of traffic
> behavior (PHB and subsequently, service class) from the DSCP used in
> the IP header to request suitable treatment in a particular network.

Hear, hear! It's especially true because CSn was included explicitly
for backwards-compatibility with RFC 791, so attempting to name new
behaviours as derivatives of CSn is particularly misleading. If you're
looking for a PHB that's like CS4 but "more discardable" that would be
CS3, anyway.

The CSn, including CS0, are unique in that the assigned code
points are not *recommended* mappings - section 4.2.2.1 of
RFC 2474 reserves them unconditionally. No wiggle room here;
there's a 1:1 correspondence between DSCP CSn and PHB CSn.
But that applies nowhere else in diffserv.

    Brian

> As an author of both RFC 2474 and 2475, I really would like to
> see draft-polk-tsvwg-delay-vs-loss-ds-service-classes revised to be
> clear on DSCP vs. PHB vs. service class and in particular to cease
> and desist from using CS4 (or any other DSCP name) as the name or part
> of the name of a service class - that will inevitably lead to the bad
> and wrong assumption that such a service class can only ever be
> deployed with the CS4 DSCP - while that sort of relationship is
> RECOMMENDED by RFC 4594, it is not REQUIRED (where "RECOMMENNDED"
> and "REQUIRED" have their RFC 2119 meanings).
> 
> An example of the possible confusion that is that there will be
> networks in which the CS4 DSCP will be deployed with only one
> of the service classes currently called "CS4" and "CS4-Discardable" in
> draft-polk-tsvwg-delay-vs-loss-ds-service-classes.  Is anyone prepared
> to try to explain with a straight face why the "CS4" service class
> isn't using the "CS4" DSCP?
> 
> Thanks,
> --David
> ----------------------------------------------------
> David L. Black, Distinguished Engineer
> EMC Corporation, 176 South St., Hopkinton, MA  01748
> +1 (508) 293-7953             FAX: +1 (508) 293-7786
> david.black@emc.com        Mobile: +1 (978) 394-7754
> ----------------------------------------------------
> 
> 
> 

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Charles Eckel \(eckelcu\)'" <eckelcu@cisco.com>, "'James Polk \(jmpolk\)'" <jmpolk@cisco.com>, <tsvwg@ietf.org>
References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <92B7E61ADAC1BB4F941F943788C08828048B5077@xmb-aln-x08.cisco.com>
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Date: Thu, 18 Jul 2013 02:26:10 -0400
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive	for	DiffServ
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Charles,

The current use of AF4x is fairly inelastic.  Few video endpoints implement
algorithms to dynamically adjust the bit-rate in the face of potential loss.
Part of the reason is that to do that properly, one needs the work the rmcat
WG is doing.

So, I agree with the argument that we need something that is different than
AF4x for what AF4x was originally intended.  It simply a practical problem
trying to change existing equipment to do something different.

At any rate, it needs a good debate.

Paul

> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Charles Eckel (eckelcu)
> Sent: Tuesday, July 16, 2013 4:47 PM
> To: James Polk (jmpolk); tsvwg@ietf.org
> Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for
> DiffServ
> 
> I read the draft and thank you for addressing this need. It does a good
> job of making the case for the need to differentiate the DSCP assignments
> of delay-and-loss-based vs. just-loss-based rate-adaptive video
> applications. While I do not agree with some of the finer points in the
> draft, including the statement in section 3 that AF4x traffic in general
> is inelastic/fixed bitrate, I completely agree that this traffic is
> largely loss sensitive and yet often relies on loss based adaption.
> The assignment of a new service class and the proposal for CS4 and CS4-
> Discardable for rate-adaptive video makes sense to me. At the same time,
> it is important to re-designate AF4x as you describe in order to match the
> reality of today's deployments and avoid future confusion.
> I look forward to seeing this work progress as it offer an incentive for
> applications to embrace congestion control algorithms such as those being
> developed in RMCAT without fearing they will be overrun by non-delay
> adaptive traffic.
> 
> Cheers,
> Charles
> 
> 
> > -----Original Message-----
> > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf
> Of
> > James Polk (jmpolk)
> > Sent: Monday, July 15, 2013 11:08 PM
> > To: tsvwg@ietf.org
> > Subject: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for
> DiffServ
> >
> > (as an author)
> >
> > Toerless and I put together a draft about legacy rate-adaptation
> > based only on loss vs. what RMCAT is looking to (for RTCWEB), which
> > is based on delay and loss.  Here's the URL.
> >
> > http://www.ietf.org/internet-drafts/draft-polk-tsvwg-delay-vs-loss-ds-
> service-
> > classes-00.txt
> >
> > It's more raw than we had in mind, but we believe this is necessary,
> > based on implementation experience and what users and customers have
> > in their networks, or are planning on having in their networks soon.
> >
> > James & Toerless



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Charles Eckel \(eckelcu\)'" <eckelcu@cisco.com>, "'James Polk \(jmpolk\)'" <jmpolk@cisco.com>, <tsvwg@ietf.org>
References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com>	<92B7E61ADAC1BB4F941F943788C08828048B5077@xmb-aln-x08.cisco.com>	<201307162106.r6GL6khf029009@rcdn-core-2.cisco.com> <92B7E61ADAC1BB4F941F943788C08828048B52EC@xmb-aln-x08.cisco.com>
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Subject: Re: [tsvwg] Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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Charles,

> I think video codec implementations in the AF4X class today often have
> variable bit rates; however, those variations are primarily due to the
> amount of motion and sometimes as an adaptation to loss.

Right, that's the issue.  Trying to shove rmcat traffic into the same bucket
will be problematic, I think.

> The latter is
> what is important in this discussion. Video codecs will continue to have
> variable bit rates in accordance with the amount of motion, but if aligned
> with the output of RMCAT, they will adapt based on delay as well as loss.

That's the discussion that needs to be had.

Paul



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From: "Eggert, Lars" <lars@netapp.com>
To: Wesley Eddy <wes@mti-systems.com>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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On Jul 17, 2013, at 20:26, Wesley Eddy <wes@mti-systems.com> wrote:
> It is not wise, in my opinion, to have algorithm-specific DSCPs.
> The DSCPs need to indicate the desired forwarding behavior, and
> not what version of code the end-host runs.

+1

Lars

From michawe@ifi.uio.no  Thu Jul 18 04:59:07 2013
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Hi everyone,

I'm still working off my email backlog, and in doing so I got a bit =
confused about the various drafts and views in this space. Since the =
issue of interactions with congestion control was also mentioned, I'll =
try to give a general comment here:

First, about the stuff quoted below, it surprises me that we're even =
debating algorithm-specific DSCPs - it should be rather obvious that we =
don't want that?! I thought that the discussion is:
a) 1 DSCP for rtcweb
vs.
b) multiple DSCPs for rtcweb (in which case I was thinking of e.g. video =
/ audio / data rather than algorithms)

I have seen various emails that indicate agreement that we should have =
at least one DSCP for rtcweb in general. The reason that was given is =
that rmcat's congestion control will prefer to not be exposed to =
TCP-like non-delay-based traffic (even though it should be able to work =
reasonably well in the presence of such traffic too). As many others, I =
totally agree.

So let's move on to the next question, which is: should we have multiple =
DSCPs (for multiple traffic classes) or not. Well, there are several =
pro's and con's here - surely more than I list below, but these are the =
ones I'd like to lay out because of the coupled-cc stuff we're =
contributing to rmcat:

Con's of multiple DSCPs:
- 1 DSCP is simpler.
- Whatever the DSCPs achieve, for traffic from a single source going =
across the same shared bottleneck, coupled congestion control can =
probably do a better job  (*)  < =3D=3D=3D  but see this footnote!

Pro's of multiple DSCPs:
- coupled congestion control only applies to traffic from one host. In =
other words, protecting your latency-critical audio from your own data =
transfer doesn't help you against the rtcweb data transfer that a family =
member generates in your home, as (s)he sends his file across the same =
shared bottleneck.


(*) Footnote: coupled congestion control should really only be applied =
when we can be pretty sure that the traffic under control traverses the =
same shared bottleneck. One simple way to achieve that for rtcweb is to =
say that all traffic going from the same source to the same destination =
shares the same bottleneck, but this only applies if "source" and =
"destination" includes the port number, i.e. if traffic is multiplexed =
over the same 5-tuple ***AND IF IT SHARES THE SAME DSCP*** (so here we =
have the DSCP constraint). Due to the amount of traffic on their list I =
find it hard to track rtcweb, but it seems to me that it's not =
guaranteed that all rtcweb traffic will indeed be multiplexed in this =
fashion. Moreover, such shared bottleneck detection is rather limiting - =
it's also quite possible that traffic from the same source to multiple =
destinations traverses the same shared bottleneck (typically the =
source's access link), but this method would then only apply coupled =
congestion control across flows towards the same destination.

=46rom the above, it seems clear that we're better off using =
measurement-based shared bottleneck detection anyway, and we're working =
on that. Note that any measurement-based mechanism will have to look at =
correlations to determine if traffic shares the same queue, and =
therefore such a method should discover that flows which carry different =
DSCPs do *not* share the same bottleneck. Thus, it would automatically =
do the right thing.

To summarize, it seems that using multiple DSCPs for rtcweb can help, =
but creates a conflict with something that we shouldn't really rely on =
for coupled congestion control. Therefore, it probably isn't really a =
big problem from a congestion control point of view. I think that =
simplicity is perhaps a more important argument for only one DSCP.

My 2 cents.

Cheers,
Michael


On Jul 18, 2013, at 8:41 AM, "Eggert, Lars" <lars@netapp.com> wrote:

> On Jul 17, 2013, at 20:26, Wesley Eddy <wes@mti-systems.com> wrote:
>> It is not wise, in my opinion, to have algorithm-specific DSCPs.
>> The DSCPs need to indicate the desired forwarding behavior, and
>> not what version of code the end-host runs.
>=20
> +1
>=20
> Lars


From david.black@emc.com  Thu Jul 18 07:07:31 2013
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From: "Black, David" <david.black@emc.com>
To: Michael Welzl <michawe@ifi.uio.no>, "Eggert, Lars" <lars@netapp.com>
Date: Thu, 18 Jul 2013 10:03:20 -0400
Thread-Topic: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <3879D71E758A7E4AA99A35DD8D41D3D91D494900@xmb-rcd-x14.cisco.com> <51E6E1D5.1050006@mti-systems.com> <460B8D74-2700-4875-B970-16D5936F3F58@netapp.com> <2CB66184-A7BF-45CD-9C9F-7B252EAD4088@ifi.uio.no>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only	rate-adaptive for	DiffServ
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<WG chair hat off>

I have a couple of requests to help focus the discussion:

(1) Please focus on service classes and PHBs.  An important reason
 is that the proposal in the draft that started this discussion is
to repurpose an existing DSCP, namely CS4.  Asking for a "new DSCP"
is premature, and this is yet more rationale for why "CS4" should
never have been used as a service class name in that draft.

(2) Can we generalize beyond rtcweb/rmcat to look at the traffic
behavior involved?  This is not the first time that adaptive transport
techniques that strive for low queue occupancy have arisen - Data
Center TCP is another example, which is based on enhanced ECN signaling:

	http://simula.stanford.edu/~alizade/Site/DCTCP.html

I assume that people like Michael and Lars are already very familiar
with DCTCP, so that link's included for everyone else :-).

If we can figure out what's going on here beyond "rtcweb/rmcat needs
its own queue", the draft I would really like to see for Vancouver is
the PHB specification for this.  That might be a variant of EF that
allows or requires adaptive response by endpoints, or it might be
something completely new.  While I recall at least talk about writing
a PHB spec for less-than-best-effort adaptive traffic (e.g., ledbat),
I don't recall that as having gone anywhere.

Thanks,
--David

> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Michael Welzl
> Sent: Thursday, July 18, 2013 7:59 AM
> To: Eggert, Lars
> Cc: tsvwg@ietf.org; Mo Zanaty (mzanaty); rmcat@ietf.org; Michael Ramalho
> (mramalho); James Polk (jmpolk)
> Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only
> rate-adaptive for DiffServ
>=20
> Hi everyone,
>=20
> I'm still working off my email backlog, and in doing so I got a bit confu=
sed
> about the various drafts and views in this space. Since the issue of
> interactions with congestion control was also mentioned, I'll try to give=
 a
> general comment here:
>=20
> First, about the stuff quoted below, it surprises me that we're even deba=
ting
> algorithm-specific DSCPs - it should be rather obvious that we don't want
> that?! I thought that the discussion is:
> a) 1 DSCP for rtcweb
> vs.
> b) multiple DSCPs for rtcweb (in which case I was thinking of e.g. video =
/
> audio / data rather than algorithms)
>=20
> I have seen various emails that indicate agreement that we should have at
> least one DSCP for rtcweb in general. The reason that was given is that
> rmcat's congestion control will prefer to not be exposed to TCP-like non-
> delay-based traffic (even though it should be able to work reasonably wel=
l in
> the presence of such traffic too). As many others, I totally agree.
>=20
> So let's move on to the next question, which is: should we have multiple =
DSCPs
> (for multiple traffic classes) or not. Well, there are several pro's and =
con's
> here - surely more than I list below, but these are the ones I'd like to =
lay
> out because of the coupled-cc stuff we're contributing to rmcat:
>=20
> Con's of multiple DSCPs:
> - 1 DSCP is simpler.
> - Whatever the DSCPs achieve, for traffic from a single source going acro=
ss
> the same shared bottleneck, coupled congestion control can probably do a
> better job  (*)  < =3D=3D=3D  but see this footnote!
>=20
> Pro's of multiple DSCPs:
> - coupled congestion control only applies to traffic from one host. In ot=
her
> words, protecting your latency-critical audio from your own data transfer
> doesn't help you against the rtcweb data transfer that a family member
> generates in your home, as (s)he sends his file across the same shared
> bottleneck.
>=20
>=20
> (*) Footnote: coupled congestion control should really only be applied wh=
en we
> can be pretty sure that the traffic under control traverses the same shar=
ed
> bottleneck. One simple way to achieve that for rtcweb is to say that all
> traffic going from the same source to the same destination shares the sam=
e
> bottleneck, but this only applies if "source" and "destination" includes =
the
> port number, i.e. if traffic is multiplexed over the same 5-tuple ***AND =
IF IT
> SHARES THE SAME DSCP*** (so here we have the DSCP constraint). Due to the
> amount of traffic on their list I find it hard to track rtcweb, but it se=
ems
> to me that it's not guaranteed that all rtcweb traffic will indeed be
> multiplexed in this fashion. Moreover, such shared bottleneck detection i=
s
> rather limiting - it's also quite possible that traffic from the same sou=
rce
> to multiple destinations traverses the same shared bottleneck (typically =
the
> source's access link), but this method would then only apply coupled
> congestion control across flows towards the same destination.
>=20
> From the above, it seems clear that we're better off using measurement-ba=
sed
> shared bottleneck detection anyway, and we're working on that. Note that =
any
> measurement-based mechanism will have to look at correlations to determin=
e if
> traffic shares the same queue, and therefore such a method should discove=
r
> that flows which carry different DSCPs do *not* share the same bottleneck=
.
> Thus, it would automatically do the right thing.
>=20
> To summarize, it seems that using multiple DSCPs for rtcweb can help, but
> creates a conflict with something that we shouldn't really rely on for co=
upled
> congestion control. Therefore, it probably isn't really a big problem fro=
m a
> congestion control point of view. I think that simplicity is perhaps a mo=
re
> important argument for only one DSCP.
>=20
> My 2 cents.
>=20
> Cheers,
> Michael
>=20
>=20
> On Jul 18, 2013, at 8:41 AM, "Eggert, Lars" <lars@netapp.com> wrote:
>=20
> > On Jul 17, 2013, at 20:26, Wesley Eddy <wes@mti-systems.com> wrote:
> >> It is not wise, in my opinion, to have algorithm-specific DSCPs.
> >> The DSCPs need to indicate the desired forwarding behavior, and
> >> not what version of code the end-host runs.
> >
> > +1
> >
> > Lars
>=20


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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>, "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOgpz4+m0uun89XEWA3y5OIxdfmJlphOaAgADNWgCAAFigAP//z5dg
Date: Thu, 18 Jul 2013 14:52:02 +0000
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <3879D71E758A7E4AA99A35DD8D41D3D91D494900@xmb-rcd-x14.cisco.com> <51E6E1D5.1050006@mti-systems.com> <460B8D74-2700-4875-B970-16D5936F3F58@netapp.com> <2CB66184-A7BF-45CD-9C9F-7B252EAD4088@ifi.uio.no>
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Cc: "tsvwg@ietf.org" <tsvwg@ietf.org>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "rmcat@ietf.org" <rmcat@ietf.org>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Michael (Welzl),

I think you have summarized the discussion well except, perhaps, for an imp=
licit assumption on how rmcat flows will (or can) be used within a singular=
 WebRTC session (we really need to standardize what we mean by stream, flow=
, session, association and other transport protocol naming conventions).

Before I add commentary to your excellent email summary (with "MAR:" below)=
, let me state a few points.

Point 1: I have presumed that the rmcat congestion control algorithm (hopef=
ully codified into an actual protocol specification someday) would be appli=
ed to the MEDIA stream(s) within the WebRTC session. This is because non-st=
reams WITHIN a WebRTC session can have different (and sometimes conflicting=
) requirements or characteristics.

Point 2: Your "cross-flow" congestion control point is a very good one, but=
 one that is out-of-scope for RMCAT as I read the charter (Lars and Mirja c=
an comment). Existing experience with conferencing sessions that have both =
real-time, semi-real-time (e.g., "screen share" where a display of  scrolli=
ng screen can be delayed for a few 100 milliseconds) and non-real time elas=
tic (e.g., background transfer) within them OVER A SINGULAR BOTTLENECK have=
 shown that it is necessary under congestion to PACE information in the non=
-real time components of those streams in order for that the real-time stre=
ams not be impaired. So your point is well taken, but out-of-scope. More be=
low.

Michael (Ramalho)

-----Original Message-----
From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
Sent: Thursday, July 18, 2013 7:59 AM
To: Eggert, Lars
Cc: Wesley Eddy; Toerless Eckert (eckert); tsvwg@ietf.org; Mo Zanaty (mzana=
ty); rmcat@ietf.org; Michael Ramalho (mramalho); James Polk (jmpolk); Cheng=
-Jia Lai (chelai)
Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ

Hi everyone,

I'm still working off my email backlog, and in doing so I got a bit confuse=
d about the various drafts and views in this space. Since the issue of inte=
ractions with congestion control was also mentioned, I'll try to give a gen=
eral comment here:

First, about the stuff quoted below, it surprises me that we're even debati=
ng algorithm-specific DSCPs - it should be rather obvious that we don't wan=
t that?! I thought that the discussion is:
a) 1 DSCP for rtcweb
vs.
b) multiple DSCPs for rtcweb (in which case I was thinking of e.g. video / =
audio / data rather than algorithms)

I have seen various emails that indicate agreement that we should have at l=
east one DSCP for rtcweb in general. The reason that was given is that rmca=
t's congestion control will prefer to not be exposed to TCP-like non-delay-=
based traffic (even though it should be able to work reasonably well in the=
 presence of such traffic too). As many others, I totally agree.

MAR: I think that is the point that started the thread. It is my only (indi=
vidual, not WG) request to tsvwg.

So let's move on to the next question, which is: should we have multiple DS=
CPs (for multiple traffic classes) or not. Well, there are several pro's an=
d con's here - surely more than I list below, but these are the ones I'd li=
ke to lay out because of the coupled-cc stuff we're contributing to rmcat:

Con's of multiple DSCPs:
- 1 DSCP is simpler.
- Whatever the DSCPs achieve, for traffic from a single source going across=
 the same shared bottleneck, coupled congestion control can probably do a b=
etter job  (*)  < =3D=3D=3D  but see this footnote!

Pro's of multiple DSCPs:
- coupled congestion control only applies to traffic from one host. In othe=
r words, protecting your latency-critical audio from your own data transfer=
 doesn't help you against the rtcweb data transfer that a family member gen=
erates in your home, as (s)he sends his file across the same shared bottlen=
eck.

MAR: This is not within scope of RMCAT or TSVWG as I see it. It is likely w=
ithin scope of WebRTC if they choose to look at it - and they can punt to T=
SVWG or RMCAT to include in their charters if they desire.

(*) Footnote: coupled congestion control should really only be applied when=
 we can be pretty sure that the traffic under control traverses the same sh=
ared bottleneck. One simple way to achieve that for rtcweb is to say that a=
ll traffic going from the same source to the same destination shares the sa=
me bottleneck, but this only applies if "source" and "destination" includes=
 the port number, i.e. if traffic is multiplexed over the same 5-tuple ***A=
ND IF IT SHARES THE SAME DSCP*** (so here we have the DSCP constraint). Due=
 to the amount of traffic on their list I find it hard to track rtcweb, but=
 it seems to me that it's not guaranteed that all rtcweb traffic will indee=
d be multiplexed in this fashion. Moreover, such shared bottleneck detectio=
n is rather limiting - it's also quite possible that traffic from the same =
source to multiple destinations traverses the same shared bottleneck (typic=
ally the source's access link), but this method would then only apply coupl=
ed congestion control across flows towards the same destination.

>From the above, it seems clear that we're better off using measurement-base=
d shared bottleneck detection anyway, and we're working on that. Note that =
any measurement-based mechanism will have to look at correlations to determ=
ine if traffic shares the same queue, and therefore such a method should di=
scover that flows which carry different DSCPs do *not* share the same bottl=
eneck. Thus, it would automatically do the right thing.

MAR: Thankfully, for the shared bottleneck problem you are considering here=
, the WebRTC (sub) stream which is causing the undue congestion relative to=
 the total WebRTC stream (sometimes called "self-induced" congestion) can s=
ense it quickly. So it is not clear to me that having a congestion measurem=
ent apparatus shared by all flows in a WebRTC session (which would appear t=
o be the optimal solution if information flows between the flows were well =
specified and instantaneous) is actually better in practice.

To summarize, it seems that using multiple DSCPs for rtcweb can help, but c=
reates a conflict with something that we shouldn't really rely on for coupl=
ed congestion control. Therefore, it probably isn't really a big problem fr=
om a congestion control point of view. I think that simplicity is perhaps a=
 more important argument for only one DSCP.

MAR: Independent of your conclusion (and thank you for making it so clear),=
 I believe it to be out of scope for RMCAT. I would suggest that these issu=
es be brought to WebRTC.

My 2 cents.

Cheers,
Michael


On Jul 18, 2013, at 8:41 AM, "Eggert, Lars" <lars@netapp.com> wrote:

> On Jul 17, 2013, at 20:26, Wesley Eddy <wes@mti-systems.com> wrote:
>> It is not wise, in my opinion, to have algorithm-specific DSCPs.
>> The DSCPs need to indicate the desired forwarding behavior, and not=20
>> what version of code the end-host runs.
>=20
> +1
>=20
> Lars


From michawe@ifi.uio.no  Fri Jul 19 02:37:30 2013
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From: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Hi,

See in line below -


On Jul 18, 2013, at 4:52 PM, Michael Ramalho (mramalho) =
<mramalho@cisco.com> wrote:

> Michael (Welzl),
>=20
> I think you have summarized the discussion well except, perhaps, for =
an implicit assumption on how rmcat flows will (or can) be used within a =
singular WebRTC session (we really need to standardize what we mean by =
stream, flow, session, association and other transport protocol naming =
conventions).
>=20
> Before I add commentary to your excellent email summary (with "MAR:" =
below), let me state a few points.
>=20
> Point 1: I have presumed that the rmcat congestion control algorithm =
(hopefully codified into an actual protocol specification someday) would =
be applied to the MEDIA stream(s) within the WebRTC session. This is =
because non-streams WITHIN a WebRTC session can have different (and =
sometimes conflicting) requirements or characteristics.

I assume that you mean "non-media-streams" with "non-streams"?

Anyway: non-media streams are out of scope for rmcat, but as a technical =
eventual decision, it would be bad to not apply rmcat congestion control =
to *all* the traffic that shares a bottleneck. I know there are more, =
but I think we all agree that one key requirement for RMCAT congestion =
control is low delay. We're not going to get low delay if the browser =
we're dealing with for rtcweb uses e.g. a loss-based congestion control =
(for data) in addition to a delay-based congestion control (for media) =
across the same bottleneck - that's exactly the situation we'd like to =
avoid. Even if we have multiple individual delay-minimizing rtcweb =
flows, they will compete for bandwidth on the bottleneck that they =
share, which inevitably causes more queue growth than necessary. 2 flows =
of whatever mechanism you can come up with create more queue growth than =
1, so we should strive to avoid that sort of competition.

I'm arguing that this would be best design-wise, but I'd also say we =
don't have to worry about rtcweb data streams for now, they are indeed =
out of scope for RMCAT. Yet, even if we want to minimize the delay for =
media streams only, ignoring the possibility of a data channel, then =
we're still better off if we combine the media streams' congestion =
controllers whenever we can.

I might have been sloppy in talking about coupled congestion control =
being applied to both data and media when, at this point, in the RMCAT =
context, we should probably ignore data - sorry!


> Point 2: Your "cross-flow" congestion control point is a very good =
one, but one that is out-of-scope for RMCAT as I read the charter (Lars =
and Mirja can comment). Existing experience with conferencing

I'm sure it's not. We're really re-iterating a pre-BOF discussion here; =
there was uniform agreement that we'd need such congestion control =
coupling, culminating in this text bits in the charter:

"The working group will:  (..) - Develop a mechanism for identifying =
shared bottlenecks between groups of
flows, and means to flexibly allocate their rates within the aggregate =
hitting
the shared bottleneck."

"Deliverables: (..) - Identifying and controlling groups of flows as a =
Proposed Standard RFC"

"Milestones:  (..)=20
Jan 2014=09
Adopt first WG draft on identifying and controlling groups of flows

Jul 2014=09
Submit identifying and controlling groups of flows to IESG for Standards =
Track publication"


> sessions that have both real-time, semi-real-time (e.g., "screen =
share" where a display of  scrolling screen can be delayed for a few 100 =
milliseconds) and non-real time elastic (e.g., background transfer) =
within them OVER A SINGULAR BOTTLENECK have shown that it is necessary =
under congestion to PACE information in the non-real time components of =
those streams in order for that the real-time streams not be impaired. =
So your point is well taken, but out-of-scope. More below.

Pacing is surely one possible way of doing it. Anyway I don't see a =
conflict here.
Also more below=85.


> Michael (Ramalho)
>=20
> -----Original Message-----
> From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
> Sent: Thursday, July 18, 2013 7:59 AM
> To: Eggert, Lars
> Cc: Wesley Eddy; Toerless Eckert (eckert); tsvwg@ietf.org; Mo Zanaty =
(mzanaty); rmcat@ietf.org; Michael Ramalho (mramalho); James Polk =
(jmpolk); Cheng-Jia Lai (chelai)
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. =
loss-only rate-adaptive for DiffServ
>=20
> Hi everyone,
>=20
> I'm still working off my email backlog, and in doing so I got a bit =
confused about the various drafts and views in this space. Since the =
issue of interactions with congestion control was also mentioned, I'll =
try to give a general comment here:
>=20
> First, about the stuff quoted below, it surprises me that we're even =
debating algorithm-specific DSCPs - it should be rather obvious that we =
don't want that?! I thought that the discussion is:
> a) 1 DSCP for rtcweb
> vs.
> b) multiple DSCPs for rtcweb (in which case I was thinking of e.g. =
video / audio / data rather than algorithms)
>=20
> I have seen various emails that indicate agreement that we should have =
at least one DSCP for rtcweb in general. The reason that was given is =
that rmcat's congestion control will prefer to not be exposed to =
TCP-like non-delay-based traffic (even though it should be able to work =
reasonably well in the presence of such traffic too). As many others, I =
totally agree.
>=20
> MAR: I think that is the point that started the thread. It is my only =
(individual, not WG) request to tsvwg.
>=20
> So let's move on to the next question, which is: should we have =
multiple DSCPs (for multiple traffic classes) or not. Well, there are =
several pro's and con's here - surely more than I list below, but these =
are the ones I'd like to lay out because of the coupled-cc stuff we're =
contributing to rmcat:
>=20
> Con's of multiple DSCPs:
> - 1 DSCP is simpler.
> - Whatever the DSCPs achieve, for traffic from a single source going =
across the same shared bottleneck, coupled congestion control can =
probably do a better job  (*)  < =3D=3D=3D  but see this footnote!
>=20
> Pro's of multiple DSCPs:
> - coupled congestion control only applies to traffic from one host. In =
other words, protecting your latency-critical audio from your own data =
transfer doesn't help you against the rtcweb data transfer that a family =
member generates in your home, as (s)he sends his file across the same =
shared bottleneck.
>=20
> MAR: This is not within scope of RMCAT or TSVWG as I see it. It is =
likely within scope of WebRTC if they choose to look at it - and they =
can punt to TSVWG or RMCAT to include in their charters if they desire.

The "data transfer" bit isn't in scope, but coupled congestion control =
is. So my example wasn't quite ideal, all you need to do is to replace =
"latency-critical audio" with e.g. "high priority audio" and "data =
transfer" with e.g. "low priority video" in my "Pro's of multiple DSCPs" =
sentence above.


> (*) Footnote: coupled congestion control should really only be applied =
when we can be pretty sure that the traffic under control traverses the =
same shared bottleneck. One simple way to achieve that for rtcweb is to =
say that all traffic going from the same source to the same destination =
shares the same bottleneck, but this only applies if "source" and =
"destination" includes the port number, i.e. if traffic is multiplexed =
over the same 5-tuple ***AND IF IT SHARES THE SAME DSCP*** (so here we =
have the DSCP constraint). Due to the amount of traffic on their list I =
find it hard to track rtcweb, but it seems to me that it's not =
guaranteed that all rtcweb traffic will indeed be multiplexed in this =
fashion. Moreover, such shared bottleneck detection is rather limiting - =
it's also quite possible that traffic from the same source to multiple =
destinations traverses the same shared bottleneck (typically the =
source's access link), but this method would then only apply coupled =
congestion control across flows towards the same destination.
>=20
> =46rom the above, it seems clear that we're better off using =
measurement-based shared bottleneck detection anyway, and we're working =
on that. Note that any measurement-based mechanism will have to look at =
correlations to determine if traffic shares the same queue, and =
therefore such a method should discover that flows which carry different =
DSCPs do *not* share the same bottleneck. Thus, it would automatically =
do the right thing.
>=20
> MAR: Thankfully, for the shared bottleneck problem you are considering =
here, the WebRTC (sub) stream which is causing the undue congestion =
relative to the total WebRTC stream (sometimes called "self-induced" =
congestion) can sense it quickly. So it is not clear to me that having a =
congestion measurement apparatus shared by all flows in a WebRTC session =
(which would appear to be the optimal solution if information flows =
between the flows were well specified and instantaneous) is actually =
better in practice.

We'll hopefully be able to show some results in Berlin that will =
convince you that this is indeed better in practice.
As for the information flow between flows being well specified, we're =
trying to address this here:
http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-01
and this document needs some discussion, so please take a look and =
comment!

As for the information flow between flows being instantaneous, at least =
for now, we're working with the notion that a flow only needs to talk to =
our coupling entity (which we call "Flow State Exchange (FSE)") whenever =
it updates it sending rate as a result of its own congestion control, =
and the FSE does *not* need to immediately notify all other flows. So, =
the constraint regarding "instantaneous" information flow is quite =
relaxed - we did this because a design goal of ours was to minimize the =
changes required to applications already implementing one or another =
form of congestion control.


> To summarize, it seems that using multiple DSCPs for rtcweb can help, =
but creates a conflict with something that we shouldn't really rely on =
for coupled congestion control. Therefore, it probably isn't really a =
big problem from a congestion control point of view. I think that =
simplicity is perhaps a more important argument for only one DSCP.
>=20
> MAR: Independent of your conclusion (and thank you for making it so =
clear), I believe it to be out of scope for RMCAT. I would suggest that =
these issues be brought to WebRTC.

I do hope that I have now convinced you that shared bottleneck detection =
and congestion control coupling are in scope, and only my mentioning of =
"data" was out of scope (even though I do think that  including the data =
channel, at some point in the future, would lead to better overall =
behavior anyway).

Cheers,
Michael


From brian.e.carpenter@gmail.com  Fri Jul 19 13:08:45 2013
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Date: Sat, 20 Jul 2013 08:08:46 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Cc: "rmcat@ietf.org" <rmcat@ietf.org>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 19/07/2013 21:37, Michael Welzl wrote:
> Hi,
> 
> See in line below -
> 
> 
> On Jul 18, 2013, at 4:52 PM, Michael Ramalho (mramalho)
> <mramalho@cisco.com> wrote:
> 
>> Michael (Welzl),
>> 
>> I think you have summarized the discussion well except,
>> perhaps, for an implicit assumption on how rmcat flows will
>> (or can) be used within a singular WebRTC session (we
>> really need to standardize what we mean by stream, flow,
>> session, association and other transport protocol naming
>> conventions).
>> 
>> Before I add commentary to your excellent email summary
>> (with "MAR:" below), let me state a few points.
>> 
>> Point 1: I have presumed that the rmcat congestion control
>> algorithm (hopefully codified into an actual protocol
>> specification someday) would be applied to the MEDIA
>> stream(s) within the WebRTC session. This is because
>> non-streams WITHIN a WebRTC session can have different (and
>> sometimes conflicting) requirements or characteristics.
> 
> I assume that you mean "non-media-streams" with
> "non-streams"?
> 
> Anyway: non-media streams are out of scope for rmcat, but as
> a technical eventual decision, it would be bad to not apply
> rmcat congestion control to *all* the traffic that shares a
> bottleneck. I know there are more, but I think we all agree
> that one key requirement for RMCAT congestion control is low
> delay. We're not going to get low delay if the browser we're
> dealing with for rtcweb uses e.g. a loss-based congestion
> control (for data) in addition to a delay-based congestion
> control (for media) across the same bottleneck - that's
> exactly the situation we'd like to avoid. 

I'm rather confused. As I understand things, this is by
definition a situation you can't avoid in the Internet, since
the mix of traffic at any point, including at a bottleneck, is
completely arbitrary. What you have just described is *exactly*
why our scaleable QoS model (diffserv) defines qualitatively
different per-hop behaviours, so that a delay-based PHB can
coexist with a loss-based PHB at the same interface. Needless to
say, there is no free lunch, which is why the default PHB (for
all unclassified traffic) gets no guarantees at all.

Wishing that it was different cannot make it different - delay-
sensitive traffic competes with loss-sensitive traffic, because
that's how the world works.

> Even if we have
> multiple individual delay-minimizing rtcweb flows, they will
> compete for bandwidth on the bottleneck that they share,
> which inevitably causes more queue growth than necessary. 2
> flows of whatever mechanism you can come up with create more
> queue growth than 1, so we should strive to avoid that sort
> of competition.
> 
> I'm arguing that this would be best design-wise, but I'd also
> say we don't have to worry about rtcweb data streams for now,
> they are indeed out of scope for RMCAT. Yet, even if we want
> to minimize the delay for media streams only, ignoring the
> possibility of a data channel, then we're still better off if
> we combine the media streams' congestion controllers whenever
> we can.

That is definitely true - if you have N streams that need the
same PHB, running them as a single behaviour aggregate is always
better.

> 
> I might have been sloppy in talking about coupled congestion
> control being applied to both data and media when, at this
> point, in the RMCAT context, we should probably ignore data -
> sorry!

Do you mean ignore it, or simply regard it as background traffic
that will soak up all resources that media traffic doesn't use?
I think those are two rather different things.

    Brian

From carlberg@g11.org.uk  Fri Jul 19 13:39:39 2013
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From: ken carlberg <carlberg@g11.org.uk>
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Cc: "tsvwg@ietf.org" <tsvwg@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On Jul 19, 2013, at 4:08 PM, Brian E Carpenter =
<brian.e.carpenter@gmail.com> wrote:

> I'm rather confused. As I understand things, this is by
> definition a situation you can't avoid in the Internet, since
> the mix of traffic at any point, including at a bottleneck, is
> completely arbitrary. What you have just described is *exactly*
> why our scaleable QoS model (diffserv) defines qualitatively
> different per-hop behaviours, so that a delay-based PHB can
> coexist with a loss-based PHB at the same interface. Needless to
> say, there is no free lunch, which is why the default PHB (for
> all unclassified traffic) gets no guarantees at all.
>=20
> Wishing that it was different cannot make it different - delay-
> sensitive traffic competes with loss-sensitive traffic, because
> that's how the world works.

there is actually some middle ground.  Traffic engineering exists and is =
used to do some measure of segmentation of flows -- particularly in =
access networks that are more prone to congestion than transit =
backbones.  And TE tends to get even more pronounced in "walled =
gardens"*.  The issues to keep in mind is that your mileage may very =
depending on the providers, and you can't count on it existing =
throughout the Internet -- its a jungle out there and one has to =
consider/accept the worst case scenario of an arbitrary traffic mix.=20

-ken

* a fair amount of folks don't consider "walled gardens" as part of the =
Internet, or of concern to the IETF.  I'll acknowledge the viewpoint, =
and respectfully disagree by keeping in mind the original phrase "an =
internetwork of networks".=

From mramalho@cisco.com  Fri Jul 19 14:32:59 2013
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
Thread-Index: AQHOgpz4+m0uun89XEWA3y5OIxdfmJlphOaAgADNWgCAAFigAP//z5dggAGbMAD///d2gA==
Date: Fri, 19 Jul 2013 21:32:51 +0000
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Hi Michael,

A few more comments below, identified with "MAR2:".

Before I begin, let's say we are also re-visiting an email thread of a few =
weeks ago. I should have been more clear about the "cross-stream" part. We =
agreed in my July 5 email (and your reply to it) that all applications ON A=
 GIVEN HOST could know about each other (what we called Case 1 - same sourc=
e address with potentially different destination addresses). However the ca=
se of a shared bottleneck created by applications on different hosts (what =
we called Case 2 - having different source addresses) you agreed was out-of=
-scope.

Now apparently you are calling "Case 2" back in scope when referring to "*a=
ll* the traffic that shares a bottleneck" (your quote and your emphasis). T=
hat is the part I disagree with.

Michael Ramalho

-----Original Message-----
From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
Sent: Friday, July 19, 2013 5:37 AM
To: Michael Ramalho (mramalho)
Cc: Eggert, Lars; Wesley Eddy; Toerless Eckert (eckert); tsvwg@ietf.org; Mo=
 Zanaty (mzanaty); rmcat@ietf.org; James Polk (jmpolk); Cheng-Jia Lai (chel=
ai)
Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ

Hi,

See in line below -


On Jul 18, 2013, at 4:52 PM, Michael Ramalho (mramalho) <mramalho@cisco.com=
> wrote:

> Michael (Welzl),
>=20
> I think you have summarized the discussion well except, perhaps, for an i=
mplicit assumption on how rmcat flows will (or can) be used within a singul=
ar WebRTC session (we really need to standardize what we mean by stream, fl=
ow, session, association and other transport protocol naming conventions).
>=20
> Before I add commentary to your excellent email summary (with "MAR:" belo=
w), let me state a few points.
>=20
> Point 1: I have presumed that the rmcat congestion control algorithm (hop=
efully codified into an actual protocol specification someday) would be app=
lied to the MEDIA stream(s) within the WebRTC session. This is because non-=
streams WITHIN a WebRTC session can have different (and sometimes conflicti=
ng) requirements or characteristics.

I assume that you mean "non-media-streams" with "non-streams"?

MAR2: Duh. Yeah. Sorry.

Anyway: non-media streams are out of scope for rmcat, but as a technical ev=
entual decision, it would be bad to not apply rmcat congestion control to *=
all* the traffic that shares a bottleneck.

MAR2: But "*all* traffic that shares a bottleneck"  includes the traffic fr=
om other hosts too! You previously agreed (on July 5) all those other flows=
 are out of scope!

MAR2: If you meant "all the traffic IN A SPECIFIC WebRTC session" (i.e., bo=
th media and non-media streams), then I agree.

MAR2: Let's please try to differentiate the cases (or equivalently, precise=
ly specify the subset of the bottleneck traffic you are considering).

I know there are more, but I think we all agree that one key requirement fo=
r RMCAT congestion control is low delay. We're not going to get low delay i=
f the browser we're dealing with for rtcweb uses e.g. a loss-based congesti=
on control (for data) in addition to a delay-based congestion control (for =
media) across the same bottleneck - that's exactly the situation we'd like =
to avoid. Even if we have multiple individual delay-minimizing rtcweb flows=
, they will compete for bandwidth on the bottleneck that they share, which =
inevitably causes more queue growth than necessary.

MAR2: Yes, two independent control loops not knowing of each other could co=
ntribute to a doubling of the queue that one alone could do with bad timing=
. Join the design team meetings to specify the experiments you want to see.

2 flows of whatever mechanism you can come up with create more queue growth=
 than 1, so we should strive to avoid that sort of competition.

I'm arguing that this would be best design-wise, but I'd also say we don't =
have to worry about rtcweb data streams for now, they are indeed out of sco=
pe for RMCAT. Yet, even if we want to minimize the delay for media streams =
only, ignoring the possibility of a data channel, then we're still better o=
ff if we combine the media streams' congestion controllers whenever we can.

I might have been sloppy in talking about coupled congestion control being =
applied to both data and media when, at this point, in the RMCAT context, w=
e should probably ignore data - sorry!

> Point 2: Your "cross-flow" congestion control point is a very good=20
> one, but one that is out-of-scope for RMCAT as I read the charter=20
> (Lars and Mirja can comment). Existing experience with conferencing

I'm sure it's not. We're really re-iterating a pre-BOF discussion here; the=
re was uniform agreement that we'd need such congestion control coupling, c=
ulminating in this text bits in the charter:

"The working group will:  (..) - Develop a mechanism for identifying shared=
 bottlenecks between groups of flows, and means to flexibly allocate their =
rates within the aggregate hitting the shared bottleneck."

"Deliverables: (..) - Identifying and controlling groups of flows as a Prop=
osed Standard RFC"

"Milestones:  (..)=20
Jan 2014=09
Adopt first WG draft on identifying and controlling groups of flows

Jul 2014=09
Submit identifying and controlling groups of flows to IESG for Standards Tr=
ack publication"

MAR2: We have been down this road before (on July 5). My comment was in res=
ponse to your "all traffic" comment. By my reading, your draft AND the RMCA=
T charter exclusively deals with flows emanating from a given host, and the=
refore limited in scope to ONLY THAT SUBSET of "all traffic hitting the sha=
red bottleneck" that came from a given host. Q.E.D.

MAR2: I stand behind my statement that it is outside the scope of RMCAT to =
consider "cross-flow" information from flows from a multiplicity of hosts t=
hat could have contributed traffic to the bottleneck.

MAR2: Perhaps we need to tighten the words in the charter so that they mean=
 the same thing to everyone ;-).

> sessions that have both real-time, semi-real-time (e.g., "screen share" w=
here a display of  scrolling screen can be delayed for a few 100 millisecon=
ds) and non-real time elastic (e.g., background transfer) within them OVER =
A SINGULAR BOTTLENECK have shown that it is necessary under congestion to P=
ACE information in the non-real time components of those streams in order f=
or that the real-time streams not be impaired. So your point is well taken,=
 but out-of-scope. More below.

Pacing is surely one possible way of doing it. Anyway I don't see a conflic=
t here.
Also more below....


> Michael (Ramalho)
>=20
> -----Original Message-----
> From: Michael Welzl [mailto:michawe@ifi.uio.no]
> Sent: Thursday, July 18, 2013 7:59 AM
> To: Eggert, Lars
> Cc: Wesley Eddy; Toerless Eckert (eckert); tsvwg@ietf.org; Mo Zanaty=20
> (mzanaty); rmcat@ietf.org; Michael Ramalho (mramalho); James Polk=20
> (jmpolk); Cheng-Jia Lai (chelai)
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.=20
> loss-only rate-adaptive for DiffServ
>=20
> Hi everyone,
>=20
> I'm still working off my email backlog, and in doing so I got a bit confu=
sed about the various drafts and views in this space. Since the issue of in=
teractions with congestion control was also mentioned, I'll try to give a g=
eneral comment here:
>=20
> First, about the stuff quoted below, it surprises me that we're even deba=
ting algorithm-specific DSCPs - it should be rather obvious that we don't w=
ant that?! I thought that the discussion is:
> a) 1 DSCP for rtcweb
> vs.
> b) multiple DSCPs for rtcweb (in which case I was thinking of e.g.=20
> video / audio / data rather than algorithms)
>=20
> I have seen various emails that indicate agreement that we should have at=
 least one DSCP for rtcweb in general. The reason that was given is that rm=
cat's congestion control will prefer to not be exposed to TCP-like non-dela=
y-based traffic (even though it should be able to work reasonably well in t=
he presence of such traffic too). As many others, I totally agree.
>=20
> MAR: I think that is the point that started the thread. It is my only (in=
dividual, not WG) request to tsvwg.
>=20
> So let's move on to the next question, which is: should we have multiple =
DSCPs (for multiple traffic classes) or not. Well, there are several pro's =
and con's here - surely more than I list below, but these are the ones I'd =
like to lay out because of the coupled-cc stuff we're contributing to rmcat=
:
>=20
> Con's of multiple DSCPs:
> - 1 DSCP is simpler.
> - Whatever the DSCPs achieve, for traffic from a single source going acro=
ss the same shared bottleneck, coupled congestion control can probably do a=
 better job  (*)  < =3D=3D=3D  but see this footnote!
>=20
> Pro's of multiple DSCPs:
> - coupled congestion control only applies to traffic from one host. In ot=
her words, protecting your latency-critical audio from your own data transf=
er doesn't help you against the rtcweb data transfer that a family member g=
enerates in your home, as (s)he sends his file across the same shared bottl=
eneck.
>=20
> MAR: This is not within scope of RMCAT or TSVWG as I see it. It is likely=
 within scope of WebRTC if they choose to look at it - and they can punt to=
 TSVWG or RMCAT to include in their charters if they desire.

The "data transfer" bit isn't in scope, but coupled congestion control is. =
So my example wasn't quite ideal, all you need to do is to replace "latency=
-critical audio" with e.g. "high priority audio" and "data transfer" with e=
.g. "low priority video" in my "Pro's of multiple DSCPs" sentence above.


> (*) Footnote: coupled congestion control should really only be applied wh=
en we can be pretty sure that the traffic under control traverses the same =
shared bottleneck. One simple way to achieve that for rtcweb is to say that=
 all traffic going from the same source to the same destination shares the =
same bottleneck, but this only applies if "source" and "destination" includ=
es the port number, i.e. if traffic is multiplexed over the same 5-tuple **=
*AND IF IT SHARES THE SAME DSCP*** (so here we have the DSCP constraint). D=
ue to the amount of traffic on their list I find it hard to track rtcweb, b=
ut it seems to me that it's not guaranteed that all rtcweb traffic will ind=
eed be multiplexed in this fashion. Moreover, such shared bottleneck detect=
ion is rather limiting - it's also quite possible that traffic from the sam=
e source to multiple destinations traverses the same shared bottleneck (typ=
ically the source's access link), but this method would then only apply cou=
pled congestion control across flows towards the same destination.
>=20
> From the above, it seems clear that we're better off using measurement-ba=
sed shared bottleneck detection anyway, and we're working on that. Note tha=
t any measurement-based mechanism will have to look at correlations to dete=
rmine if traffic shares the same queue, and therefore such a method should =
discover that flows which carry different DSCPs do *not* share the same bot=
tleneck. Thus, it would automatically do the right thing.
>=20
> MAR: Thankfully, for the shared bottleneck problem you are considering he=
re, the WebRTC (sub) stream which is causing the undue congestion relative =
to the total WebRTC stream (sometimes called "self-induced" congestion) can=
 sense it quickly. So it is not clear to me that having a congestion measur=
ement apparatus shared by all flows in a WebRTC session (which would appear=
 to be the optimal solution if information flows between the flows were wel=
l specified and instantaneous) is actually better in practice.

We'll hopefully be able to show some results in Berlin that will convince y=
ou that this is indeed better in practice.
As for the information flow between flows being well specified, we're tryin=
g to address this here:
http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-01
and this document needs some discussion, so please take a look and comment!

As for the information flow between flows being instantaneous, at least for=
 now, we're working with the notion that a flow only needs to talk to our c=
oupling entity (which we call "Flow State Exchange (FSE)") whenever it upda=
tes it sending rate as a result of its own congestion control, and the FSE =
does *not* need to immediately notify all other flows. So, the constraint r=
egarding "instantaneous" information flow is quite relaxed - we did this be=
cause a design goal of ours was to minimize the changes required to applica=
tions already implementing one or another form of congestion control.


> To summarize, it seems that using multiple DSCPs for rtcweb can help, but=
 creates a conflict with something that we shouldn't really rely on for cou=
pled congestion control. Therefore, it probably isn't really a big problem =
from a congestion control point of view. I think that simplicity is perhaps=
 a more important argument for only one DSCP.
>=20
> MAR: Independent of your conclusion (and thank you for making it so clear=
), I believe it to be out of scope for RMCAT. I would suggest that these is=
sues be brought to WebRTC.

I do hope that I have now convinced you that shared bottleneck detection an=
d congestion control coupling are in scope, and only my mentioning of "data=
" was out of scope (even though I do think that  including the data channel=
, at some point in the future, would lead to better overall behavior anyway=
).

MAR2: You haven't convinced me that coupled congestion control across diffe=
rent hosts are in scope yet - because your own document at its present vers=
ion number does not address the entire scope of the problem. Reasonable peo=
ple can disagree. See you in Berlin.

Cheers,
Michael


From brian.e.carpenter@gmail.com  Fri Jul 19 16:12:13 2013
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Date: Sat, 20 Jul 2013 11:12:20 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 20/07/2013 08:39, ken carlberg wrote:
> On Jul 19, 2013, at 4:08 PM, Brian E Carpenter <brian.e.carpenter@gmail.com> wrote:
> 
>> I'm rather confused. As I understand things, this is by
>> definition a situation you can't avoid in the Internet, since
>> the mix of traffic at any point, including at a bottleneck, is
>> completely arbitrary. What you have just described is *exactly*
>> why our scaleable QoS model (diffserv) defines qualitatively
>> different per-hop behaviours, so that a delay-based PHB can
>> coexist with a loss-based PHB at the same interface. Needless to
>> say, there is no free lunch, which is why the default PHB (for
>> all unclassified traffic) gets no guarantees at all.
>>
>> Wishing that it was different cannot make it different - delay-
>> sensitive traffic competes with loss-sensitive traffic, because
>> that's how the world works.
> 
> there is actually some middle ground.  Traffic engineering exists and is used to do some measure of segmentation of flows -- particularly in access networks that are more prone to congestion than transit backbones.

Absolutely, but of course to do any kind of dynamic TE you need
to be able to classify traffic somehow.

>  And TE tends to get even more pronounced in "walled gardens"*.  The issues to keep in mind is that your mileage may very depending on the providers, and you can't count on it existing throughout the Internet -- its a jungle out there and one has to consider/accept the worst case scenario of an arbitrary traffic mix. 

Exactly. Whatever we do to improve matters inside a walled
garden must not make things worse in that jungle.

> -ken
> 
> * a fair amount of folks don't consider "walled gardens" as part of the Internet, or of concern to the IETF.  I'll acknowledge the viewpoint, and respectfully disagree by keeping in mind the original phrase "an internetwork of networks".

I hesitate to keep banging on about the diffserv model, but I
think it's a realistic paradigm, and it definitely assumes
walled gardens as far as QoS is concerned (called "diffserv
domains"). You can slice and dice the problem as much as you
want, but I think we're stuck with the inter-domain model
boiling down to best effort, in the absence of pairwise SLAs
between competing providers.

   Brian

From michawe@ifi.uio.no  Sat Jul 20 00:18:54 2013
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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Hi,

My apologies to everyone whose time I have wasted by not being clear =
enough: this all seems to be a big misunderstanding.
I am *not* calling "Case 2" back in scope and I really only meant =
coupled congestion control for traffic from a single host.

Note this bit from my first email in this discussion:
**
Pro's of multiple DSCPs:
- coupled congestion control only applies to traffic from one host. In =
other words, protecting your latency-critical audio from your own data =
transfer doesn't help you against the rtcweb data transfer that a family =
member generates in your home, as (s)he sends his file across the same =
shared bottleneck.
**

A few things are somewhat off here: as we already discussed, I shouldn't =
have mentioned "data transfer"; probably worse, the notion of =
"protecting" is misleading, because what I really meant is that you =
cannot use coupled congestion control to protect one flow against =
another when that other doesn't originate from the same host.

(then again, the first sentence was quite clear?!  "coupled congestion =
control only applies to traffic from one host.")
Anyway, sorry again to everyone I have confused.

As for joining the design team discussions, the usual time of the telco =
doesn't work for me, but I plan to be at the design team meeting in =
Berlin.

Cheers,
Michael


On Jul 19, 2013, at 11:32 PM, Michael Ramalho (mramalho) =
<mramalho@cisco.com> wrote:

> Hi Michael,
>=20
> A few more comments below, identified with "MAR2:".
>=20
> Before I begin, let's say we are also re-visiting an email thread of a =
few weeks ago. I should have been more clear about the "cross-stream" =
part. We agreed in my July 5 email (and your reply to it) that all =
applications ON A GIVEN HOST could know about each other (what we called =
Case 1 - same source address with potentially different destination =
addresses). However the case of a shared bottleneck created by =
applications on different hosts (what we called Case 2 - having =
different source addresses) you agreed was out-of-scope.
>=20
> Now apparently you are calling "Case 2" back in scope when referring =
to "*all* the traffic that shares a bottleneck" (your quote and your =
emphasis). That is the part I disagree with.
>=20
> Michael Ramalho
>=20
> -----Original Message-----
> From: Michael Welzl [mailto:michawe@ifi.uio.no]=20
> Sent: Friday, July 19, 2013 5:37 AM
> To: Michael Ramalho (mramalho)
> Cc: Eggert, Lars; Wesley Eddy; Toerless Eckert (eckert); =
tsvwg@ietf.org; Mo Zanaty (mzanaty); rmcat@ietf.org; James Polk =
(jmpolk); Cheng-Jia Lai (chelai)
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. =
loss-only rate-adaptive for DiffServ
>=20
> Hi,
>=20
> See in line below -
>=20
>=20
> On Jul 18, 2013, at 4:52 PM, Michael Ramalho (mramalho) =
<mramalho@cisco.com> wrote:
>=20
>> Michael (Welzl),
>>=20
>> I think you have summarized the discussion well except, perhaps, for =
an implicit assumption on how rmcat flows will (or can) be used within a =
singular WebRTC session (we really need to standardize what we mean by =
stream, flow, session, association and other transport protocol naming =
conventions).
>>=20
>> Before I add commentary to your excellent email summary (with "MAR:" =
below), let me state a few points.
>>=20
>> Point 1: I have presumed that the rmcat congestion control algorithm =
(hopefully codified into an actual protocol specification someday) would =
be applied to the MEDIA stream(s) within the WebRTC session. This is =
because non-streams WITHIN a WebRTC session can have different (and =
sometimes conflicting) requirements or characteristics.
>=20
> I assume that you mean "non-media-streams" with "non-streams"?
>=20
> MAR2: Duh. Yeah. Sorry.
>=20
> Anyway: non-media streams are out of scope for rmcat, but as a =
technical eventual decision, it would be bad to not apply rmcat =
congestion control to *all* the traffic that shares a bottleneck.
>=20
> MAR2: But "*all* traffic that shares a bottleneck"  includes the =
traffic from other hosts too! You previously agreed (on July 5) all =
those other flows are out of scope!
>=20
> MAR2: If you meant "all the traffic IN A SPECIFIC WebRTC session" =
(i.e., both media and non-media streams), then I agree.
>=20
> MAR2: Let's please try to differentiate the cases (or equivalently, =
precisely specify the subset of the bottleneck traffic you are =
considering).
>=20
> I know there are more, but I think we all agree that one key =
requirement for RMCAT congestion control is low delay. We're not going =
to get low delay if the browser we're dealing with for rtcweb uses e.g. =
a loss-based congestion control (for data) in addition to a delay-based =
congestion control (for media) across the same bottleneck - that's =
exactly the situation we'd like to avoid. Even if we have multiple =
individual delay-minimizing rtcweb flows, they will compete for =
bandwidth on the bottleneck that they share, which inevitably causes =
more queue growth than necessary.
>=20
> MAR2: Yes, two independent control loops not knowing of each other =
could contribute to a doubling of the queue that one alone could do with =
bad timing. Join the design team meetings to specify the experiments you =
want to see.
>=20
> 2 flows of whatever mechanism you can come up with create more queue =
growth than 1, so we should strive to avoid that sort of competition.
>=20
> I'm arguing that this would be best design-wise, but I'd also say we =
don't have to worry about rtcweb data streams for now, they are indeed =
out of scope for RMCAT. Yet, even if we want to minimize the delay for =
media streams only, ignoring the possibility of a data channel, then =
we're still better off if we combine the media streams' congestion =
controllers whenever we can.
>=20
> I might have been sloppy in talking about coupled congestion control =
being applied to both data and media when, at this point, in the RMCAT =
context, we should probably ignore data - sorry!
>=20
>> Point 2: Your "cross-flow" congestion control point is a very good=20
>> one, but one that is out-of-scope for RMCAT as I read the charter=20
>> (Lars and Mirja can comment). Existing experience with conferencing
>=20
> I'm sure it's not. We're really re-iterating a pre-BOF discussion =
here; there was uniform agreement that we'd need such congestion control =
coupling, culminating in this text bits in the charter:
>=20
> "The working group will:  (..) - Develop a mechanism for identifying =
shared bottlenecks between groups of flows, and means to flexibly =
allocate their rates within the aggregate hitting the shared =
bottleneck."
>=20
> "Deliverables: (..) - Identifying and controlling groups of flows as a =
Proposed Standard RFC"
>=20
> "Milestones:  (..)=20
> Jan 2014=09
> Adopt first WG draft on identifying and controlling groups of flows
>=20
> Jul 2014=09
> Submit identifying and controlling groups of flows to IESG for =
Standards Track publication"
>=20
> MAR2: We have been down this road before (on July 5). My comment was =
in response to your "all traffic" comment. By my reading, your draft AND =
the RMCAT charter exclusively deals with flows emanating from a given =
host, and therefore limited in scope to ONLY THAT SUBSET of "all traffic =
hitting the shared bottleneck" that came from a given host. Q.E.D.
>=20
> MAR2: I stand behind my statement that it is outside the scope of =
RMCAT to consider "cross-flow" information from flows from a =
multiplicity of hosts that could have contributed traffic to the =
bottleneck.
>=20
> MAR2: Perhaps we need to tighten the words in the charter so that they =
mean the same thing to everyone ;-).
>=20
>> sessions that have both real-time, semi-real-time (e.g., "screen =
share" where a display of  scrolling screen can be delayed for a few 100 =
milliseconds) and non-real time elastic (e.g., background transfer) =
within them OVER A SINGULAR BOTTLENECK have shown that it is necessary =
under congestion to PACE information in the non-real time components of =
those streams in order for that the real-time streams not be impaired. =
So your point is well taken, but out-of-scope. More below.
>=20
> Pacing is surely one possible way of doing it. Anyway I don't see a =
conflict here.
> Also more below....
>=20
>=20
>> Michael (Ramalho)
>>=20
>> -----Original Message-----
>> From: Michael Welzl [mailto:michawe@ifi.uio.no]
>> Sent: Thursday, July 18, 2013 7:59 AM
>> To: Eggert, Lars
>> Cc: Wesley Eddy; Toerless Eckert (eckert); tsvwg@ietf.org; Mo Zanaty=20=

>> (mzanaty); rmcat@ietf.org; Michael Ramalho (mramalho); James Polk=20
>> (jmpolk); Cheng-Jia Lai (chelai)
>> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.=20
>> loss-only rate-adaptive for DiffServ
>>=20
>> Hi everyone,
>>=20
>> I'm still working off my email backlog, and in doing so I got a bit =
confused about the various drafts and views in this space. Since the =
issue of interactions with congestion control was also mentioned, I'll =
try to give a general comment here:
>>=20
>> First, about the stuff quoted below, it surprises me that we're even =
debating algorithm-specific DSCPs - it should be rather obvious that we =
don't want that?! I thought that the discussion is:
>> a) 1 DSCP for rtcweb
>> vs.
>> b) multiple DSCPs for rtcweb (in which case I was thinking of e.g.=20
>> video / audio / data rather than algorithms)
>>=20
>> I have seen various emails that indicate agreement that we should =
have at least one DSCP for rtcweb in general. The reason that was given =
is that rmcat's congestion control will prefer to not be exposed to =
TCP-like non-delay-based traffic (even though it should be able to work =
reasonably well in the presence of such traffic too). As many others, I =
totally agree.
>>=20
>> MAR: I think that is the point that started the thread. It is my only =
(individual, not WG) request to tsvwg.
>>=20
>> So let's move on to the next question, which is: should we have =
multiple DSCPs (for multiple traffic classes) or not. Well, there are =
several pro's and con's here - surely more than I list below, but these =
are the ones I'd like to lay out because of the coupled-cc stuff we're =
contributing to rmcat:
>>=20
>> Con's of multiple DSCPs:
>> - 1 DSCP is simpler.
>> - Whatever the DSCPs achieve, for traffic from a single source going =
across the same shared bottleneck, coupled congestion control can =
probably do a better job  (*)  < =3D=3D=3D  but see this footnote!
>>=20
>> Pro's of multiple DSCPs:
>> - coupled congestion control only applies to traffic from one host. =
In other words, protecting your latency-critical audio from your own =
data transfer doesn't help you against the rtcweb data transfer that a =
family member generates in your home, as (s)he sends his file across the =
same shared bottleneck.
>>=20
>> MAR: This is not within scope of RMCAT or TSVWG as I see it. It is =
likely within scope of WebRTC if they choose to look at it - and they =
can punt to TSVWG or RMCAT to include in their charters if they desire.
>=20
> The "data transfer" bit isn't in scope, but coupled congestion control =
is. So my example wasn't quite ideal, all you need to do is to replace =
"latency-critical audio" with e.g. "high priority audio" and "data =
transfer" with e.g. "low priority video" in my "Pro's of multiple DSCPs" =
sentence above.
>=20
>=20
>> (*) Footnote: coupled congestion control should really only be =
applied when we can be pretty sure that the traffic under control =
traverses the same shared bottleneck. One simple way to achieve that for =
rtcweb is to say that all traffic going from the same source to the same =
destination shares the same bottleneck, but this only applies if =
"source" and "destination" includes the port number, i.e. if traffic is =
multiplexed over the same 5-tuple ***AND IF IT SHARES THE SAME DSCP*** =
(so here we have the DSCP constraint). Due to the amount of traffic on =
their list I find it hard to track rtcweb, but it seems to me that it's =
not guaranteed that all rtcweb traffic will indeed be multiplexed in =
this fashion. Moreover, such shared bottleneck detection is rather =
limiting - it's also quite possible that traffic from the same source to =
multiple destinations traverses the same shared bottleneck (typically =
the source's access link), but this method would then only apply coupled =
congestion control across flows towards the same destination.
>>=20
>> =46rom the above, it seems clear that we're better off using =
measurement-based shared bottleneck detection anyway, and we're working =
on that. Note that any measurement-based mechanism will have to look at =
correlations to determine if traffic shares the same queue, and =
therefore such a method should discover that flows which carry different =
DSCPs do *not* share the same bottleneck. Thus, it would automatically =
do the right thing.
>>=20
>> MAR: Thankfully, for the shared bottleneck problem you are =
considering here, the WebRTC (sub) stream which is causing the undue =
congestion relative to the total WebRTC stream (sometimes called =
"self-induced" congestion) can sense it quickly. So it is not clear to =
me that having a congestion measurement apparatus shared by all flows in =
a WebRTC session (which would appear to be the optimal solution if =
information flows between the flows were well specified and =
instantaneous) is actually better in practice.
>=20
> We'll hopefully be able to show some results in Berlin that will =
convince you that this is indeed better in practice.
> As for the information flow between flows being well specified, we're =
trying to address this here:
> http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-01
> and this document needs some discussion, so please take a look and =
comment!
>=20
> As for the information flow between flows being instantaneous, at =
least for now, we're working with the notion that a flow only needs to =
talk to our coupling entity (which we call "Flow State Exchange (FSE)") =
whenever it updates it sending rate as a result of its own congestion =
control, and the FSE does *not* need to immediately notify all other =
flows. So, the constraint regarding "instantaneous" information flow is =
quite relaxed - we did this because a design goal of ours was to =
minimize the changes required to applications already implementing one =
or another form of congestion control.
>=20
>=20
>> To summarize, it seems that using multiple DSCPs for rtcweb can help, =
but creates a conflict with something that we shouldn't really rely on =
for coupled congestion control. Therefore, it probably isn't really a =
big problem from a congestion control point of view. I think that =
simplicity is perhaps a more important argument for only one DSCP.
>>=20
>> MAR: Independent of your conclusion (and thank you for making it so =
clear), I believe it to be out of scope for RMCAT. I would suggest that =
these issues be brought to WebRTC.
>=20
> I do hope that I have now convinced you that shared bottleneck =
detection and congestion control coupling are in scope, and only my =
mentioning of "data" was out of scope (even though I do think that  =
including the data channel, at some point in the future, would lead to =
better overall behavior anyway).
>=20
> MAR2: You haven't convinced me that coupled congestion control across =
different hosts are in scope yet - because your own document at its =
present version number does not address the entire scope of the problem. =
Reasonable people can disagree. See you in Berlin.
>=20
> Cheers,
> Michael
>=20


From Karen.Nielsen@tieto.com  Mon Jul 22 03:03:00 2013
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Subject: Re: [tsvwg] draft-ietf-tsvwg-sctp-failover: comments and suggestions
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Hi Yoshifumi,

Please find some follow up below.

BR, Karen

Please accept the following addition comments:

The motivation for the 2MTU, I now understand, is to aim to mitigate unwant=
ed effects of
path bouncing resulting from a spurious T3-timeout followed by failover
and subsequent "immediate" switchback operation of RFC4960 once HB ACK
is acknowledged within the back-off'ed RTO.

While I think that one can very well say in this document that the CWND cou=
ld be set to 2MTU
in case the first HB sent on a PF path is successfully ack'ed, then I would=
 like to note that
this then is motivated by a desire, first and foremost, to be backward comp=
atible with RC4960, more than it
really necessarily addresses how to best mitigate throughput degradation ef=
fects of spurious T3-timeouts.
Indeed for implementations that operate with different switchback modes tha=
n the immediate switchback operation of RFC4960,
 (e.g. operate with certain permanent failover modes), then the set of the =
CWND to 2 MTU on the previous PF path
is not of a big significance as it can be most optimal, from a performance =
perspective,
to stay on the new data carrying path, which at this point in time may have=
 a CWND of IW, IW+1MTU or even higher.
I think that the motivation for setting the CWND to 2MTU possibly should be=
 made conditionally on whether the SCTP indeed follows
the immediate switchback operation of RFC4960.

One further observation is that in a situation where it is only the 2dn or =
the 3rd (or 4th..) of the HBs sent on the PF path, that is successful, then
setting the CWND to 2 MTU may actually constitute a decrease of the CWND. T=
his is the case if the implementation
follows RC4960 and raise the CWND to IW (4MTU) after an idle period of one =
RTO following the T3 timeout.
When in the text it is said that the CWND should be set to 1MTU at the stat=
e transition from PF to ACTIVE does this then very explicitly
addresses the fact that unsuccessful HBs, sent on a PF path, shall keep the=
 CWND down at 1MTU as otherwise data failures would ?
If so then I think that it could be good to add this clarification.

Ok. I see your point.
I think we should explicitly say that we should set CWND to 1MTU after unsu=
ccessful HBs.
I don't think using 4MTU after seeing HB timeout is good approach. Thanks f=
or pointing out.

[Karen] I think that there could an issue here. On a data carrying path the=
n the CWND is set to 1 MTU after a data T3 timeout.
But on other paths, then I think that it could be unfortunate if a random l=
oss of one HB (a periodic path monitoring HB)
made the CWND be reduced to 1 MTU.

I would opt for the set CWND to 1MTU after unsuccessful HBs to be performed=
 ONLY on a path in PF state, i.e., not for the HB that made the path enter =
PF state nor for HBs done of Paths that are INACTIVE state, thereby one wou=
ld make the CWND reductions be a little more robust towards random packet d=
rops (of HBs).

Alternatively one should do nothing special at all and keep the situation o=
f RFC4960 where the fate of the HBs do not impact the CWND at all.

Side note: A single drop of a data packet on a sparsely loaded path (or in =
data tail - not enough data for Fast Recovery to kick in) today also result=
s in CWND reduction and this of course compares with the loss of HB situati=
on. But I think that this is a problem, rather than a desired effect, in th=
e present algorithms that one continuously tries to find solutions for. We =
should not make this problem worse by also introducing if for single drop o=
f HBs. I think.


Final thoughts:

We have in real deployment experienced fault situations where HBs consisten=
tly were able to get through on the primary path,
but where the path broke due to congestion once it got loaded with data. Th=
ere are different ways in which one can improve the
protocol operation to best tackle such a situation. Our present solution, w=
hich is to deploy a permanent failover mode together with PF
and looking for a way to add reliability to the T3 timeouts, may not be the=
 ultimate way to solve this, but I would like to note that
in such a situation then consistently returning to the primary path and set=
ting CWND =3D 2MTU is not the right thing to do.


>- Section 5.3: Permanent Failover
>
>We have good experiences from live networks with running SCTP in a "no
>switchback" mode of operation with no automatic migration back to a prefer=
red
>primary path. Once the primary path is left due to it being potentially fa=
iled, then a
>selected alternative path automatically becomes the new primary path.
>
>This mode of operation corresponds to a simplified variant of the permanen=
t
>failover operation proposed in [CAR002] with alpha =3D beta =3D
>SCTP_PEER_ADDR:THLDS. The mode of operation is run with notifications of
>change of primary path being provided to upper layers via RFC6458
>SCTP_PEER_ADDR_CHANGE semantics (SCTP_ADDR_MADE_PRIM).
>
>The mode of operation does not suit all purposes, but in networks environm=
ents
>where a number of equally preferred paths are available, then by this mode=
 of
>operation one minimizes the effects of unnecessary path bouncing and the t=
hereby
>associated effects of throughput degradation due to path changes and slow =
start
>operation.
>
>Based on our experiences we would recommend and support that the Permanent
>Failover mode of operation, at least in the simple variant described above=
, is
>promoted as a valid alternative approach. For the purpose of activation of=
 this
>mode of operation a switchback (or failover) mode operation parameter coul=
d be
>introduced as configurable by the introduction of an appropriate new read/=
write
>socket option in the Socket API.
>
>
>Yes. I agree that there will be a situation where Permanent Failover work =
effectively.
>OTOH, supporting this prevents us from keeping RFC4960 behavior as RFC4960
>doesn't have this mode.
>But, I believe this is a very interesting point and we would like to get m=
ore feedback
>in order to think about this more.
>
I understand to continue to support RFC4960 behaviour, but I suppose that o=
ne can add to RFC4960
behaviour :-) We have the simple "no switchback mode" described above runni=
ng in deployment.

Yes, I think that's doable. I think the question here is the degree of reco=
mmendation for this.
The easiest one is just saying "An implementation MAY support no switchback=
 mode", but I'm gueesing some people might prefer more explicit expressions=
.

[Karen] Yes, I agree that better than MAY is desirable.
But as a starting point then I think that a MAY support other switchback mo=
des is much better than discouraging such,
which is like I read the draft at present.


>- Section 5.4: The impact of HB timeouts in association error counter
>
>The draft recommends for excluding HB timeouts from contributing to the
>(association) error counter. We fully support this approach for all timeou=
ts of HBs
>that are send to probe the state of destinations addresses that are in PF =
state and
>we would even propose for the draft to demand that generally timeouts stem=
ming
>from HBs, which are not sent on the Primary Path, will be excluded from th=
e
>association error counter.
>
>Thereby the association error counter and thus the association fate reflec=
ts the
>forward progress of the data transmission to peer endpoint while at the sa=
me time it
>will be appropriately isolated from timeouts of HBs sent on non-data carry=
ing paths.
>
>Allowing timeout of HBs sent on the Primary Path to impact the association=
 error
>counter will serve to validate the aliveness of the peer endpoint when the=
re is no
>data in transit.
>
>This suggestion follows the thought expressed in http://www.ietf.org/mail-
>archive/web/tsvwg/current/msg11294.html and it is thus believed to fulfil =
the original
>intend of RFC4960.
>
>This is a bit difficult point for me. I don't know much about the original=
 intention of
>RFC4960, but if we follow what is written there, i think we need to increm=
ent
>association error count. (Please correct me if I misunderstand this point)=
 So, my
>personal preference is to set bigger AMR to avoid early termination so tha=
t we can
>keep the RFC4960 behavior.
>However, we don't mind discussing to update the logic described in 4960 fo=
r more
>sophisticated solution.
>
Thanks !

Yes, I concur that the text of section 8.1 of RFC4960 stipulates that all H=
B Acks and HB timeouts
influence the association error counter. But from the referenced discussion=
 thread,
 http://www.ietf.org/mail-archive/web/tsvwg/current/msg11294.html  (simply =
copying it in here to avoid repeating the arguments)
 it also seems that by purpose no keywords were used when writing these par=
agraphs.
Further as the Potentially Feature was not defined at the time, I am not su=
re that
it would be valid to say that the text of RFC4960 apply (at least) to the H=
Bs sent as part of this function.

>From our perspective what we would like to support and see happening is
that all HBs, whether sent as part of the PF function or not, influence the=
 association error counter
if and only if they are transmitted on the Primary Path (and then with stan=
dard logic running making sure that all data transmissions
prevent (or cancel in race conditions) any influence from HBs).

Thereby the association error counter will be influenced by (and will thus =
reflect) the data transmission only.
Except in idle situation where HBs on Primary Path will serve to update the=
 association error counter in a similar manner as data would.
Thereby one get a very clean and reliable implementation of the association=
 error counter. By reliable, I mean that one can use
the same AMR value regardless of the level of HB probing that one have acti=
vated and regardless of whether the Potentially Failed feature is activated=
 or not.
Such an implementation also makes the AMR value be independent from the num=
ber of paths supported in the association context.
One can choose to use a value of AMR proportional with the number of paths =
in order to have all paths be tried a number of times before association cl=
osure, but
with the choice suggested, one is not forced to "artificially" raise the AM=
R value to make room for the HBs that will fire as part of the potentially =
failed feature as well
as for some fraction of the Path Monitoring HBs that can happen to fire dur=
ing the same timeframe.

An additional thought is that in the odd cases where HBs get through wherea=
s data do not, then I think that it is contra-productive to have the associ=
ation error counter be reset
as long as data consistently fail.

I, personally, does not so much like the suggestion to use a "maximal" or "=
bigger" AMR. I think that it will be complex to deduce exactly
what failure time (aka number of data retransmission)  that a certain AMR v=
alue will guarantee you,
whereas when only data transmission influences the AMR, things will be clea=
n, simple and predictable.

I agree that the approach described in the draft is not the best approach. =
But, since it doesn't violate RFC4960, we recommend it in the draft.
I think what you suggest sounds reasonable and prefer to update 4960 based =
on this, although I guess we'll need more discussions.
[Karen] I agree, I will try to initiate such more general discussions.
>
>- Section 5.1 Thoughts on the destination address state machine
>
>In RFC4960 address state machine, notification of cease of use of the curr=
ent
>primary path as transmission path for new data is provided to upper layer =
via
>(coupled with) the announcement of the destination as being UNREACHABLE. W=
ith
>the introduction of the potentially failed state as a semi-hidden state (n=
ot resulting in
>notified state transition to UL) the information on the fail-over to use a=
 new data
>carrying path is lost.
>While there may be merits in suppressing notifications on potentially fail=
ed
>addresses, then we believe that it is unfortunate that the notification of=
 failover from
>usage of the primary path for new data is lost.
>This is especially problematic in path bouncing scenarios where the primar=
y path is
>unstable but still works some. E.g., HBs may go through, but the path brea=
ks (data
>transmission fails) as soon as it is more loaded. In such a situation no n=
otifications
>may be generated to UL, even if new data transmission consistently cycles =
to and
>from the primary path. The solution to such path bouncing scenarios will h=
ave to
>come from more advanced path failover mechanism, like [CAR002] (and would =
not
>be solved just by notification), but it would be a great help if appropria=
te
>notifications were provided.
>
>I think it will be OK to send notification as long as it's a new notificat=
ion type. But
>we should not send notifications defined in RFC6458.
ok - I assume this is for backward compatible reasons.

Yes. As far as I can think that's the reason.
[Karen] Ok.

>I guess having a new notification type won't cause problems, but I would l=
ike to get
>more feedback on this.
Ok - we can come back with a suggestion.

Thank you so much for many valuable feedback. We really appreciate!

Regards,
--
Yoshifumi


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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'f=
ont-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>Hi Yoshifum=
i,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0p=
t;font-family:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial"=
,"sans-serif";color:#1F497D'>Please find some follow up below.<o:p></o:p></=
span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"=
Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DM=
soNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";c=
olor:#1F497D'>BR, Karen<o:p></o:p></span></p><p class=3DMsoNormal><span sty=
le=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'><o:p=
>&nbsp;</o:p></span></p><div style=3D'border:none;border-left:solid blue 1.=
5pt;padding:0cm 0cm 0cm 4.0pt'><div><div><div><div><p class=3DMsoNormal>Ple=
ase accept the following addition comments:<br><br>The motivation for the 2=
MTU, I now understand, is to aim to mitigate unwanted effects of<br>path bo=
uncing resulting from a spurious T3-timeout followed by failover<br>and sub=
sequent &quot;immediate&quot; switchback operation of RFC4960 once HB ACK<b=
r>is acknowledged within the back-off'ed RTO.<br><br>While I think that one=
 can very well say in this document that the CWND could be set to 2MTU<br>i=
n case the first HB sent on a PF path is successfully ack'ed, then I would =
like to note that<br>this then is motivated by a desire, first and foremost=
, to be backward compatible with RC4960, more than it<br>really necessarily=
 addresses how to best mitigate throughput degradation effects of spurious =
T3-timeouts.<br>Indeed for implementations that operate with different swit=
chback modes than the immediate switchback operation of RFC4960,<br>&nbsp;(=
e.g. operate with certain permanent failover modes), then the set of the CW=
ND to 2 MTU on the previous PF path<br>is not of a big significance as it c=
an be most optimal, from a performance perspective,<br>to stay on the new d=
ata carrying path, which at this point in time may have a CWND of IW, IW+1M=
TU or even higher.<br>I think that the motivation for setting the CWND to 2=
MTU possibly should be made conditionally on whether the SCTP indeed follow=
s<br>the immediate switchback operation of RFC4960.<br><br>One further obse=
rvation is that in a situation where it is only the 2dn or the 3rd (or 4th.=
.) of the HBs sent on the PF path, that is successful, then<br>setting the =
CWND to 2 MTU may actually constitute a decrease of the CWND. This is the c=
ase if the implementation<br>follows RC4960 and raise the CWND to IW (4MTU)=
 after an idle period of one RTO following the T3 timeout.<br>When in the t=
ext it is said that the CWND should be set to 1MTU at the state transition =
from PF to ACTIVE does this then very explicitly<br>addresses the fact that=
 unsuccessful HBs, sent on a PF path, shall keep the CWND down at 1MTU as o=
therwise data failures would ?<br>If so then I think that it could be good =
to add this clarification.<o:p></o:p></p><div><p class=3DMsoNormal><o:p>&nb=
sp;</o:p></p></div><div><p class=3DMsoNormal>Ok. I see your point.<o:p></o:=
p></p></div><div><p class=3DMsoNormal>I think we should explicitly say that=
 we should set CWND to 1MTU after unsuccessful HBs. <o:p></o:p></p></div><d=
iv><p class=3DMsoNormal>I don't think using 4MTU after seeing HB timeout is=
 good approach. Thanks for pointing out.<o:p></o:p></p></div><div><p class=
=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p c=
lass=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:"Arial",=
"sans-serif";color:#1F497D'>[Karen] I think that there could an issue here.=
 On a data carrying path then the CWND is set to 1 MTU after a data T3 time=
out. <o:p></o:p></span></i></b></p><p class=3DMsoNormal><b><i><span style=
=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>But on=
 other paths, then I think that it could be unfortunate if a random loss of=
 one HB (a periodic path monitoring HB) <o:p></o:p></span></i></b></p><p cl=
ass=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:"Arial","=
sans-serif";color:#1F497D'>made the CWND be reduced to 1 MTU. <o:p></o:p></=
span></i></b></p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt=
;font-family:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></=
i></b></p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-f=
amily:"Arial","sans-serif";color:#1F497D'>I would opt for the </span></i></=
b>set CWND to 1MTU after unsuccessful HBs <b><i><span style=3D'font-size:10=
.0pt;font-family:"Arial","sans-serif";color:#1F497D'>to be performed ONLY o=
n a path in PF state, i.e., not for the HB that made the path enter PF stat=
e nor for HBs done of Paths that are INACTIVE state, thereby one would make=
 the CWND reductions be a little more robust towards random packet drops (o=
f HBs). <o:p></o:p></span></i></b></p><p class=3DMsoNormal><b><i><span styl=
e=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'><o:p>=
&nbsp;</o:p></span></i></b></p><p class=3DMsoNormal><b><i><span style=3D'fo=
nt-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>Alternativel=
y one should do nothing special at all and keep the situation of RFC4960 wh=
ere the fate of the HBs do not impact the CWND at all.<o:p></o:p></span></i=
></b></p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-fa=
mily:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></i></b></=
p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:"A=
rial","sans-serif";color:#1F497D'>Side note: A single drop of a data packet=
 on a sparsely loaded path (or in data tail - not enough data for Fast Reco=
very to kick in) today also results in CWND reduction and this of course co=
mpares with the loss of HB situation. But I think that this is a problem, r=
ather than a desired effect, in the present algorithms that one continuousl=
y tries to find solutions for. We should not make this problem worse by als=
o introducing if for single drop of HBs. I think. <o:p></o:p></span></i></b=
></p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family=
:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></i></b></p></=
div><blockquote style=3D'border:none;border-left:solid #CCCCCC 1.0pt;paddin=
g:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><p class=3DMsoNorma=
l><br>Final thoughts:<br><br>We have in real deployment experienced fault s=
ituations where HBs consistently were able to get through on the primary pa=
th,<br>but where the path broke due to congestion once it got loaded with d=
ata. There are different ways in which one can improve the<br>protocol oper=
ation to best tackle such a situation. Our present solution, which is to de=
ploy a permanent failover mode together with PF<br>and looking for a way to=
 add reliability to the T3 timeouts, may not be the ultimate way to solve t=
his, but I would like to note that<br>in such a situation then consistently=
 returning to the primary path and setting CWND =3D 2MTU is not the right t=
hing to do.<o:p></o:p></p><div><div><p class=3DMsoNormal style=3D'margin-bo=
ttom:12.0pt'><br><br>&gt;- Section 5.3: Permanent Failover<br>&gt;<br>&gt;W=
e have good experiences from live networks with running SCTP in a &quot;no<=
br>&gt;switchback&quot; mode of operation with no automatic migration back =
to a preferred<br>&gt;primary path. Once the primary path is left due to it=
 being potentially failed, then a<br>&gt;selected alternative path automati=
cally becomes the new primary path.<br>&gt;<br>&gt;This mode of operation c=
orresponds to a simplified variant of the permanent<br>&gt;failover operati=
on proposed in [CAR002] with alpha =3D beta =3D<br>&gt;SCTP_PEER_ADDR:THLDS=
. The mode of operation is run with notifications of<br>&gt;change of prima=
ry path being provided to upper layers via RFC6458<br>&gt;SCTP_PEER_ADDR_CH=
ANGE semantics (SCTP_ADDR_MADE_PRIM).<br>&gt;<br>&gt;The mode of operation =
does not suit all purposes, but in networks environments<br>&gt;where a num=
ber of equally preferred paths are available, then by this mode of<br>&gt;o=
peration one minimizes the effects of unnecessary path bouncing and the the=
reby<br>&gt;associated effects of throughput degradation due to path change=
s and slow start<br>&gt;operation.<br>&gt;<br>&gt;Based on our experiences =
we would recommend and support that the Permanent<br>&gt;Failover mode of o=
peration, at least in the simple variant described above, is<br>&gt;promote=
d as a valid alternative approach. For the purpose of activation of this<br=
>&gt;mode of operation a switchback (or failover) mode operation parameter =
could be<br>&gt;introduced as configurable by the introduction of an approp=
riate new read/write<br>&gt;socket option in the Socket API.<br>&gt;<br>&gt=
;<br>&gt;Yes. I agree that there will be a situation where Permanent Failov=
er work effectively.<br>&gt;OTOH, supporting this&nbsp;prevents us from kee=
ping RFC4960 behavior as RFC4960<br>&gt;doesn't have this mode.<br>&gt;But,=
 I believe this is a very interesting point and we would like to get more f=
eedback<br>&gt;in order to think about this more.<br>&gt;<o:p></o:p></p></d=
iv></div><p class=3DMsoNormal>I understand to continue to support RFC4960 b=
ehaviour, but I suppose that one can add to RFC4960<br>behaviour :-) We hav=
e the simple &quot;no switchback mode&quot; described above running in depl=
oyment.<o:p></o:p></p></blockquote><div><p class=3DMsoNormal><o:p>&nbsp;</o=
:p></p></div><div><p class=3DMsoNormal>Yes, I think that's doable. I think =
the question here is the degree of recommendation for this.<o:p></o:p></p><=
/div><div><p class=3DMsoNormal>The easiest one is just saying &quot;An impl=
ementation MAY support no switchback mode&quot;, but I'm gueesing some peop=
le might prefer more explicit expressions.&nbsp;<o:p></o:p></p></div><div><=
p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span><=
/p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:"=
Arial","sans-serif";color:#1F497D'>[Karen] Yes, I agree that better than MA=
Y is desirable.<o:p></o:p></span></i></b></p><p class=3DMsoNormal><b><i><sp=
an style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D=
'>But as a starting point then I think that a MAY support other switchback =
modes is much better than discouraging such,<o:p></o:p></span></i></b></p><=
p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:"Aria=
l","sans-serif";color:#1F497D'>which is like I read the draft at present. <=
o:p></o:p></span></i></b></p><p class=3DMsoNormal><b><i><span style=3D'font=
-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o=
:p></span></i></b></p></div><blockquote style=3D'border:none;border-left:so=
lid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:=
0cm'><div><p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>&gt;- Sec=
tion 5.4: The impact of HB timeouts in association error counter<br>&gt;<br=
>&gt;The draft recommends for excluding HB timeouts from contributing to th=
e<br>&gt;(association) error counter. We fully support this approach for al=
l timeouts of HBs<br>&gt;that are send to probe the state of destinations a=
ddresses that are in PF state and<br>&gt;we would even propose for the draf=
t to demand that generally timeouts stemming<br>&gt;from HBs, which are not=
 sent on the Primary Path, will be excluded from the<br>&gt;association err=
or counter.<br>&gt;<br>&gt;Thereby the association error counter and thus t=
he association fate reflects the<br>&gt;forward progress of the data transm=
ission to peer endpoint while at the same time it<br>&gt;will be appropriat=
ely isolated from timeouts of HBs sent on non-data carrying paths.<br>&gt;<=
br>&gt;Allowing timeout of HBs sent on the Primary Path to impact the assoc=
iation error<br>&gt;counter will serve to validate the aliveness of the pee=
r endpoint when there is no<br>&gt;data in transit.<br>&gt;<br>&gt;This sug=
gestion follows the thought expressed in <a href=3D"http://www.ietf.org/mai=
l-" target=3D"_blank">http://www.ietf.org/mail-</a><br>&gt;archive/web/tsvw=
g/current/msg11294.html and it is thus believed to fulfil the original<br>&=
gt;intend of RFC4960.<br>&gt;<br>&gt;This is a bit difficult point for me. =
I don't know much about the original intention of<br>&gt;RFC4960, but if we=
 follow what is written there, i think we need to increment<br>&gt;associat=
ion error count. (Please correct me if I misunderstand this point) So, my<b=
r>&gt;personal preference is to set bigger AMR to avoid early termination s=
o that we can<br>&gt;keep the RFC4960 behavior.<br>&gt;However, we don't mi=
nd discussing to update the logic described in 4960 for more<br>&gt;sophist=
icated solution.<br>&gt;<o:p></o:p></p></div><p class=3DMsoNormal>Thanks !<=
br><br>Yes, I concur that the text of section 8.1 of RFC4960 stipulates tha=
t all HB Acks and HB timeouts<br>influence the association error counter. B=
ut from the referenced discussion thread,<br>&nbsp;<a href=3D"http://www.ie=
tf.org/mail-archive/web/tsvwg/current/msg11294.html" target=3D"_blank">http=
://www.ietf.org/mail-archive/web/tsvwg/current/msg11294.html</a> &nbsp;(sim=
ply copying it in here to avoid repeating the arguments)<br>&nbsp;it also s=
eems that by purpose no keywords were used when writing these paragraphs.<b=
r>Further as the Potentially Feature was not defined at the time, I am not =
sure that<br>it would be valid to say that the text of RFC4960 apply (at le=
ast) to the HBs sent as part of this function.<br><br>From our perspective =
what we would like to support and see happening is<br>that all HBs, whether=
 sent as part of the PF function or not, influence the association error co=
unter<br>if and only if they are transmitted on the Primary Path (and then =
with standard logic running making sure that all data transmissions<br>prev=
ent (or cancel in race conditions) any influence from HBs).<br><br>Thereby =
the association error counter will be influenced by (and will thus reflect)=
 the data transmission only.<br>Except in idle situation where HBs on Prima=
ry Path will serve to update the association error counter in a similar man=
ner as data would.<br>Thereby one get a very clean and reliable implementat=
ion of the association error counter. By reliable, I mean that one can use<=
br>the same AMR value regardless of the level of HB probing that one have a=
ctivated and regardless of whether the Potentially Failed feature is activa=
ted or not.<br>Such an implementation also makes the AMR value be independe=
nt from the number of paths supported in the association context.<br>One ca=
n choose to use a value of AMR proportional with the number of paths in ord=
er to have all paths be tried a number of times before association closure,=
 but<br>with the choice suggested, one is not forced to &quot;artificially&=
quot; raise the AMR value to make room for the HBs that will fire as part o=
f the potentially failed feature as well<br>as for some fraction of the Pat=
h Monitoring HBs that can happen to fire during the same timeframe.<br><br>=
An additional thought is that in the odd cases where HBs get through wherea=
s data do not, then I think that it is contra-productive to have the associ=
ation error counter be reset<br>as long as data consistently fail.<br><br>I=
, personally, does not so much like the suggestion to use a &quot;maximal&q=
uot; or &quot;bigger&quot; AMR. I think that it will be complex to deduce e=
xactly<br>what failure time (aka number of data retransmission) &nbsp;that =
a certain AMR value will guarantee you,<br>whereas when only data transmiss=
ion influences the AMR, things will be clean, simple and predictable.<o:p><=
/o:p></p></blockquote><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div>=
<div><p class=3DMsoNormal>I agree that the approach described in the draft =
is not the best approach. But, since it doesn't violate RFC4960, we recomme=
nd it in the draft.<o:p></o:p></p></div><div><p class=3DMsoNormal style=3D'=
margin-bottom:12.0pt'>I think what you suggest sounds reasonable and prefer=
 to update 4960 based on this, although I guess we'll need more discussions=
.<o:p></o:p></p><p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><i><=
span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F49=
7D'>[Karen] I agree, I will try to initiate such more general discussions.<=
/span></i></b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-ser=
if";color:#1F497D'><o:p></o:p></span></p></div><blockquote style=3D'border:=
none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:=
4.8pt;margin-right:0cm'><div><p class=3DMsoNormal style=3D'margin-bottom:12=
.0pt'>&gt;<br>&gt;- Section 5.1 Thoughts on the destination address state m=
achine<br>&gt;<br>&gt;In RFC4960 address state machine, notification of cea=
se of use of the current<br>&gt;primary path as transmission path for new d=
ata is provided to upper layer via<br>&gt;(coupled with) the announcement o=
f the destination as being UNREACHABLE. With<br>&gt;the introduction of the=
 potentially failed state as a semi-hidden state (not resulting in<br>&gt;n=
otified state transition to UL) the information on the fail-over to use a n=
ew data<br>&gt;carrying path is lost.<br>&gt;While there may be merits in s=
uppressing notifications on potentially failed<br>&gt;addresses, then we be=
lieve that it is unfortunate that the notification of failover from<br>&gt;=
usage of the primary path for new data is lost.<br>&gt;This is especially p=
roblematic in path bouncing scenarios where the primary path is<br>&gt;unst=
able but still works some. E.g., HBs may go through, but the path breaks (d=
ata<br>&gt;transmission fails) as soon as it is more loaded. In such a situ=
ation no notifications<br>&gt;may be generated to UL, even if new data tran=
smission consistently cycles to and<br>&gt;from the primary path. The solut=
ion to such path bouncing scenarios will have to<br>&gt;come from more adva=
nced path failover mechanism, like [CAR002] (and would not<br>&gt;be solved=
 just by notification), but it would be a great help if appropriate<br>&gt;=
notifications were provided.<br>&gt;<br>&gt;I think it will be OK to send n=
otification as long as it's a new notification type. But<br>&gt;we should n=
ot send notifications defined in RFC6458.<o:p></o:p></p></div><p class=3DMs=
oNormal>ok - I assume this is for backward compatible reasons.<o:p></o:p></=
p></blockquote><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p=
 class=3DMsoNormal>Yes. As far as I can think that's the reason.<o:p></o:p>=
</p><p class=3DMsoNormal><b><i><span style=3D'font-size:10.0pt;font-family:=
"Arial","sans-serif";color:#1F497D'>[Karen] Ok.</span></i></b><span style=
=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'><o:p><=
/o:p></span></p></div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div>=
<blockquote style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><p class=3DMsoNorm=
al style=3D'margin-bottom:12.0pt'>&gt;I guess having a new notification typ=
e won't cause problems, but I would like to get<br>&gt;more feedback on thi=
s.<o:p></o:p></p></div><p class=3DMsoNormal>Ok - we can come back with a su=
ggestion.<o:p></o:p></p></blockquote><div><p class=3DMsoNormal><o:p>&nbsp;<=
/o:p></p></div><div><p class=3DMsoNormal>Thank you so much for many valuabl=
e feedback. We really appreciate!<o:p></o:p></p></div><div><p class=3DMsoNo=
rmal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>Regards,<o:p></o:=
p></p></div><div><p class=3DMsoNormal>--<o:p></o:p></p></div><div><p class=
=3DMsoNormal>Yoshifumi&nbsp;<o:p></o:p></p></div></div><p class=3DMsoNormal=
><o:p>&nbsp;</o:p></p></div></div></div></div></div></body></html>=

--_000_CF340E42AED0874C81947E18863DE77B29DBD5C24EEXMB03eutieto_--

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Subject: Re: [tsvwg] comments draft-tuexen-tsvwg-sctp-prpolicies-01
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Hi Michael,
I think this addresses my earlier comments (and thanks for the responses on
the other questions earlier).

cheers,
jamal


On Fri, Jul 12, 2013 at 11:21 AM, Michael Tuexen <tuexen@fh-muenster.de>wrote:

> On Jul 10, 2013, at 11:48 AM, Jamal Hadi Salim <hadi@mojatatu.com> wrote:
>
> >
> > Hi Michael,
> >
> > On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen <tuexen@fh-muenster.de>
> wrote:
> >
> > Assume the sender sends a user message which is approximately of size
> 3*MTU.
> > So SCTP will fragment it into 3 DATA chunk and send them. If the user
> > has specified that the number of retransmissions is limited to 0 and
> > the last fragment gets lost in the network, the sender will increment
> > sprstat_abandoned_sent.
> >
> > Ok,  so the last fragment essentially doesnt get acknowledged before we
> > abandon ship. And the other stat indicates we abandoned while we still
> had some
> > sitting locally. Is the intersection of the two (some are sent but not
> acked and
> > some are still sitting locally) implying both stats will be incremented?
> >
> > Other than the perceived ambiguity -  why do i care to know about the
> > two types of abandoned messages in a user app? I  will have to sum
> > those two, so why not just give me one stat?
> >
> > BTW: the doc will benefit from such a small description as you gave me
> > above.
> The new revision contains some clarification:
> http://tools.ietf.org/html/draft-tuexen-tsvwg-sctp-prpolicies-02
>
> Best regards
> Michael
> >
> > What will happen at the receiver side if you assume that the first two
> > fragments have been received. If partial delivery was started, you will
> > get a SCTP_PARTIAL_DELIVERY_EVENT event indicating
> SCTP_PARTIAL_DELIVERY_ABORTED.
> > Without partial delivery being started, the receiver could also drop the
> > first part of the message.
> >
> >
> > You answered the essence of my question.
> > Silly follow up: I am assuming that event will come with an EOR?
> > To use your example (at receiver):
> > EPOLLIN, recvmsg chunk1, chunk2, EAGAIN
> > EPOLLIN, recvmsg SCTP_PARTIAL_DELIVERY_EVENT, EOR
> >
> > cheers,
> > jamal
>
>

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<div dir=3D"ltr">Hi Michael,<div>I think this addresses my earlier comments=
 (and thanks for the responses on</div><div>the other questions earlier).</=
div><div><br></div><div>cheers,</div><div>jamal<br><div class=3D"gmail_extr=
a">

<br><br><div class=3D"gmail_quote">On Fri, Jul 12, 2013 at 11:21 AM, Michae=
l Tuexen <span dir=3D"ltr">&lt;<a href=3D"mailto:tuexen@fh-muenster.de" tar=
get=3D"_blank">tuexen@fh-muenster.de</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

<div class=3D"im">On Jul 10, 2013, at 11:48 AM, Jamal Hadi Salim &lt;<a hre=
f=3D"mailto:hadi@mojatatu.com">hadi@mojatatu.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt; Hi Michael,<br>
&gt;<br>
&gt; On Wed, Jul 10, 2013 at 3:42 AM, Michael Tuexen &lt;<a href=3D"mailto:=
tuexen@fh-muenster.de">tuexen@fh-muenster.de</a>&gt; wrote:<br>
&gt;<br>
&gt; Assume the sender sends a user message which is approximately of size =
3*MTU.<br>
&gt; So SCTP will fragment it into 3 DATA chunk and send them. If the user<=
br>
&gt; has specified that the number of retransmissions is limited to 0 and<b=
r>
&gt; the last fragment gets lost in the network, the sender will increment<=
br>
&gt; sprstat_abandoned_sent.<br>
&gt;<br>
&gt; Ok, =A0so the last fragment essentially doesnt get acknowledged before=
 we<br>
&gt; abandon ship. And the other stat indicates we abandoned while we still=
 had some<br>
&gt; sitting locally. Is the intersection of the two (some are sent but not=
 acked and<br>
&gt; some are still sitting locally) implying both stats will be incremente=
d?<br>
&gt;<br>
&gt; Other than the perceived ambiguity - =A0why do i care to know about th=
e<br>
&gt; two types of abandoned messages in a user app? I =A0will have to sum<b=
r>
&gt; those two, so why not just give me one stat?<br>
&gt;<br>
&gt; BTW: the doc will benefit from such a small description as you gave me=
<br>
&gt; above.<br>
</div>The new revision contains some clarification:<br>
<a href=3D"http://tools.ietf.org/html/draft-tuexen-tsvwg-sctp-prpolicies-02=
" target=3D"_blank">http://tools.ietf.org/html/draft-tuexen-tsvwg-sctp-prpo=
licies-02</a><br>
<br>
Best regards<br>
<span class=3D"HOEnZb"><font color=3D"#888888">Michael<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5">&gt;<br>
&gt; What will happen at the receiver side if you assume that the first two=
<br>
&gt; fragments have been received. If partial delivery was started, you wil=
l<br>
&gt; get a SCTP_PARTIAL_DELIVERY_EVENT event indicating SCTP_PARTIAL_DELIVE=
RY_ABORTED.<br>
&gt; Without partial delivery being started, the receiver could also drop t=
he<br>
&gt; first part of the message.<br>
&gt;<br>
&gt;<br>
&gt; You answered the essence of my question.<br>
&gt; Silly follow up: I am assuming that event will come with an EOR?<br>
&gt; To use your example (at receiver):<br>
&gt; EPOLLIN, recvmsg chunk1, chunk2, EAGAIN<br>
&gt; EPOLLIN, recvmsg SCTP_PARTIAL_DELIVERY_EVENT, EOR<br>
&gt;<br>
&gt; cheers,<br>
&gt; jamal<br>
<br>
</div></div></blockquote></div><br></div></div></div>

--047d7b3a8d0c367fea04e1547ea1--

From gorry@erg.abdn.ac.uk  Wed Jul 24 01:21:39 2013
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] TSVWG Final Agenda -  Request for Speaker Slides!
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We have now uploaded the final Agenda for the meeting on Monday:
http://www.ietf.org/proceedings/87/agenda/agenda-87-tsvwg

There are two TSVWG sessions, the second will include a group of drafts 
supporting transport functions that may be applicable to RMCAT.

People intending to present should note that the allocated time is 
expected to include time for questions. You also must send the chairs 
slides by Sunday 17:00.

NOTE 1 - Because we have a 9:00 start on Monday, you *DO* need to send 
copies of your slides in advance!

NOTE 2 - We will need a minute-taker. In the past few meetings we used 
valuable meeting time to get volunteers, if some people volunteer in 
advance it would be appreciated by all!

Gorry, David & James.

From gorry@erg.abdn.ac.uk  Thu Jul 25 05:31:03 2013
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Subject: [tsvwg] TSVWG Status report 25th July 2013
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The list below shows the status of the working group documents as we see 
it. Please check below and comment on drafts using the list. Please do 
send any corrections/omissions to the chairs.

Best wishes,

Gorry, David & James
(TSVWG Chairs)
June 2013


---
Recently published:

draft-ietf-tsvwg-sctpsocket was published as RFC 6458.
draft-ietf-tsvwg-sctp-strrst was published as RFC 6525.
draft-ietf-tsvwg-source-quench was published as RFC 6633.
draft-ietf-tsvwg-sctp-udp-encap was published as RFC 6951.

-------------------------------------------------------------------
IDs in RFC Editor Queue:
http://www.rfc-editor.org/queue.html
None.

-------------------------------------------------------------------
IDs in IESG processing:
None.

-------------------------------------------------------------------
WG Drafts with Chairs:

draft-ietf-tsvwg-sctp-sack-immediately - Document Shepherd: Gorry
Replaces: draft-tuexen-tsvwg-sctp-sack-immediately
- WG adoption call after IETF-85, started 13/12/2013. Support {AB, IR, 
SL, MB, TD}
- Implemented/released in FreeBSD, partially implemented in Linux
WGLC concluded Friday 5th April 2013
- No major remaining comments at IETF-86, March 2013.
{Supported in WGLC: TD; MW; DB; GF; YN; AB} - summary sent 8th April 2013
{Comments include: DB; YN; AB; MW}
DUE: Authors to confirm draft is ready (Editor)
DUE Authors to confirm IPR status (All)
DUE: Write-up to be submitted (Gorry)
-------------------------------------------------------------------
WG Drafts:

---
draft-ietf-tsvwg-byte-pkt-congest - Document Shepherd: Gorry
IETF-81: discussed whether this should be BCP.
Status changed to BCP (new-ID WG -05)
WGLC concluded with comments on I-D, Friday 30th March 2012.
Draft version of write-up sent to list, comments received.
- Gorry working with authors to clear list of WGLC questions (5 sept 2012).
WG Chair received version that includes additional feedback during WGLC
WGLC Completed (post IESG Review) ended Friday, 28th June 2013.
STATUS: Awaiting revised-ID (subject to another WGLC)
---
draft-ietf-tsvwg-rsvp-pcn - Document Shepherd: James
Replaces: draft-karagiannis-pcn-tsvwg-rsvp-pcn
AD decision allowed this to be added to the milestones
Document adopted, status will be EXP (IETF-84 due to dependencies on PCN R
RSVP-DIR review received (BB).
MILESTONE EXPIRED.
DUE: Expected ready for WGLC
---
draft-ietf-tsvwg-natsupp-tsvwg - Document Shepherd: James
Replaces: draft-stewart-natsupp-tsvwg
Dependency from BEHAVE WG.
Adopted as a work item 21 Sept 2010 (Gorry).
WG -00, 29/11/2010 Uploaded as: draft-ietf-natsupp-tsvwg
Authors restructured draft (-03)
Added support for single and multi-homed support.
IETF-86: 3 people in meeting had read -04 or -05.
DUE: Authors please update following comments to TSVWG.
DUE: Please comment on list - MILESTONE EXPIRED.
---
draft-ietf-tsvwg-sctp-failover - Document Shepherd: James
Replaces: draft-nishida-tsvwg-sctp-failover
Individual-00 Presented in Prague, IETF-80.
- Understood not to conflict with draft-tuexen-tsvwg-sctp-multipath
- PF has been coded into FreeBSD
Adopted by WG 26/6/2012. [Milestone: Jul 2013, EXP]
DUE: Please comment on list.
---
draft-ietf-tsvwg-sctp-dtls-encaps - Document Shepherd: Gorry
Replaces: draft-tuexen-tsvwg-sctp-dtls-encaps-01
WG adopted 13/12/2013. Support {AB, IR, SL, MB, TD} [Milestone: Jan 
2014, PS]
Implemented in Firefox (FF21). Required by RTCweb.
DUE: Please comment on list.
---
draft-ietf-tsvwg-intserv-multiple-tspec - Document Shepherd: David
WG interest in this topic recorded at IETF-78.
Charter update needed to progress this work.
5 Reviews needed to determine energy/technical direction.
WG -05 (presented in Beijing, IETF-79)
WG -06 (presented in Prague, IETF-80)
2 reviews from RSVP-DIR received (Bruce Davie, ?)
2 additional reviews promised (Ken Carlberg, Francois LeF)
Chairs asked AD for a Charter update (IESG agreed)
Draft discussed at IETF-80, and request to update charter agreed
- AD advised 4 named reviewers will be required
Adopted for progression as PS, for May 2012
- New approach presented following WG comments, revised MILESTONE needed.
DUE: Discussion needed on list.
---

draft-ietf-tsvwg-port-use - Document Shepherd: Gorry
Replaces: draft-touch-tsvwg-port-use-00
- Intended to be advice to protocol designers needing a port.
5 people looked at this document, Prague IETF-80
Individual -01 July 2011
IETF-81 insufficient feedback from WG at this time.
IETF-82, discussed.
Adopted by WG
Presented IETF-86, March 2013
Milestone updated to June 2014
DUE: Please read individual sections and provide feedback!
---
draft-ietf-tsvwg-rsvp-app-id-vv-profiles
Replaces: draft-polk-tsvwg-rsvp-app-id-vv-profiles
-01 Presented in Beijing, IETF-79.
-02, 14-Mar-2011
Presented IETF-80.
Seek to coordinate with music with partner ID:
draft-polk-mmusic-traffic-class-for-sdp
Gorry liaised with MMUSIC WG Chairs on companion draft (norm ref from 
this tsvwg draft).
Adopted as a TSVWG work item, December 2013, [Milestone Jul 2013].
This will update a RFC 2872 (PS).
Presented at IETF-86, Mar 2013 (no comments)
DUE: Expected ready for WGLC;
DUE: Reviewers are needed to prepare for a WGLC.

-------------------------------------------------------------------
The following have received recent discussion at TSVWG meetings or on
the list. Inclusion in the list below does not indicate support for
these specific drafts and other contributions are also warmly welcomed.

draft-geib-tsvwg-diffserv-intercon
draft-yong-tsvwg-udp-encap-4-ip-tunneling
draft-shepherd-multicast-udp-guidelines
draft-turner-rsvp-auth-update
draft-wei-tsvwg-tunnel-congestion-feedback
draft-dhesikan-tsvwg-rtcweb-qos
draft-polk-tsvwg-delay-vs-loss-ds-service-classes
draft-stewart-tsvwg-sctp-ndata
draft-tuexen-tsvwg-sctp-prpolicies
draft-tuexen-tsvwg-sctp-scheduling

For other drafts linked to tsvwg, please see:
http://tools.ietf.org/wg/tsvwg/


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Date: Fri, 26 Jul 2013 00:11:41 -0700
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: "Karen E. E. Nielsen" <Karen.Nielsen@tieto.com>
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Subject: Re: [tsvwg] draft-ietf-tsvwg-sctp-failover: comments and suggestions
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Hi Karen,


On Mon, Jul 22, 2013 at 3:00 AM,  <Karen.Nielsen@tieto.com> wrote:

> I think we should explicitly say that we should set CWND to 1MTU after
> unsuccessful HBs.
>
> I don't think using 4MTU after seeing HB timeout is good approach. Thanks
> for pointing out.
>
>
> [Karen] I think that there could an issue here. On a data carrying path then
> the CWND is set to 1 MTU after a data T3 timeout.
>
> But on other paths, then I think that it could be unfortunate if a random
> loss of one HB (a periodic path monitoring HB)
>
> made the CWND be reduced to 1 MTU.
>
>
>
> I would opt for the set CWND to 1MTU after unsuccessful HBs to be performed
> ONLY on a path in PF state, i.e., not for the HB that made the path enter PF
> state nor for HBs done of Paths that are INACTIVE state, thereby one would
> make the CWND reductions be a little more robust towards random packet drops
> (of HBs).
> Alternatively one should do nothing special at all and keep the situation of
> RFC4960 where the fate of the HBs do not impact the CWND at all.

Agreed. I meant for only a path in PF status since PF only sends HBs a
path in PF status.
I think the behavior of HB timeout on a path in other states should follow 4960.

> Alternatively one should do nothing special at all and keep the situation of
> RFC4960 where the fate of the HBs do not impact the CWND at all.
>
>
>
> Side note: A single drop of a data packet on a sparsely loaded path (or in
> data tail - not enough data for Fast Recovery to kick in) today also results
> in CWND reduction and this of course compares with the loss of HB situation.
> But I think that this is a problem, rather than a desired effect, in the
> present algorithms that one continuously tries to find solutions for. We
> should not make this problem worse by also introducing if for single drop of
> HBs. I think.

I also think we need something here as single drop of HB might not be
an indication of severe congestion.
I'm thinking that something like draft-ietf-tcpm-newcwv might need to
be considered.

Thanks,
--
Yoshifumi

From gorry@erg.abdn.ac.uk  Fri Jul 26 01:05:49 2013
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] Fwd: Request for AD to progress draft-ietf-tsvwg-sctp-sack-immediately
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TSVWG for your info:

We have now submitted draft-ietf-tsvwg-sctp-sack-immediately for 
publication and I enclose a document write-up.

Gorry


-------- Original Message --------

TSV AD, IESG,

TSVWG would like the above draft to be considered for publication as a
PS. I enclose a document write-up.

Gorry
(TSVWG-Chair, Document Shepherd)



*** copy of write-up for draft-ietf-tsvwg-sctp-sack-immediately ***

As required by RFC 4858, this is the current template for the Document
Shepherd Write-Up.

Changes are expected over time. This version is dated 24 February 2012.

(1) What type of RFC is being requested (BCP, Proposed Standard,
Internet Standard, Informational, Experimental, or Historic)? Why is
this the proper type of RFC? Is this type of RFC indicated in the title
page header?

This document is intended as a PS.

(2) The IESG approval announcement includes a Document Announcement
Write-Up. Please provide such a Document Announcement Write-Up. Recent
examples can be found in the "Action" announcements for approved
documents. The approval announcement contains the following sections:

Technical Summary:

    This document updates RFC 4960 by defining a method for the sender of
    a DATA chunk to indicate that the corresponding SACK chunk should be
    sent back immediately and not be delayed.  It is done by specifying a
    bit in the DATA chunk header, called the I-bit, which can get set
    either by the SCTP implementation or by the application using an SCTP
    stack.  Since unknown flags in chunk headers are ignored by SCTP
    implementations, this extension does not introduce any
    interoperability problems.


Working Group Summary:

There was consensus to adopt this as a WG document, review by the WG,
and agreement by the WG to finally publish this.


Document Quality:

This document is seen as ready to publish.

FreeBSD supports this extension since FreeBSD 7.2, released May 2009.
The Linux kernel supports it also (accessibility by a the user was added
recently to netinet/sctp.h).


Personnel:

Who is the Document Shepherd? Who is the Responsible Area Director?

I am the document shepherd, G Fairhurst, <gorry@erg.abdn.ac.uk>.
The responsible AD is: Spencer Dawkins <spencerdawkins.ietf@gmail.com>


(3) Briefly describe the review of this document that was performed by
the Document Shepherd. If this version of the document is not ready for
publication, please explain why the document is being forwarded to the IESG.

WGLC Announcement sent draft-ietf-tsvwg-sctp-sack-immediately-02 on 19th
March 2013 and concluded Friday 5th April 2013 with comments from 6
people provided. These have been addressed in the present version.

(4) Does the document Shepherd have any concerns about the depth or
breadth of the reviews that have been performed?

No.

(5) Do portions of the document need review from a particular or from
broader perspective, e.g., security, operational complexity, AAA, DNS,
DHCP, XML, or internationalization? If so, describe the review that took
place.

No

(6) Describe any specific concerns or issues that the Document Shepherd
has with this document that the Responsible Area Director and/or the
IESG should be aware of? For example, perhaps he or she is uncomfortable
with certain parts of the document, or has concerns whether there really
is a need for it. In any event, if the WG has discussed those issues and
has indicated that it still wishes to advance the document, detail those
concerns here.

None.

(7) Has each author confirmed that any and all appropriate IPR
disclosures required for full conformance with the provisions of BCP 78
and BCP 79 have already been filed. If not, explain why?

Yes.

(8) Has an IPR disclosure been filed that references this document? If
so, summarize any WG discussion and conclusion regarding the IPR
disclosures.

As far as I know, no IPR disclosures have been submitted.

(9) How solid is the WG consensus behind this document? Does it
represent the strong concurrence of a few individuals, with others being
silent, or does the WG as a whole understand and agree with it?

There was initial discussion of whether this mechanism was needed -
especially since no equivalent method exists in TCP. After use-cases
were provided there was strong support for this work and no objections.
There was review of the work within the WG, and consensus to publish. No
objections were noted.

(10) Has anyone threatened an appeal or otherwise indicated extreme
discontent? If so, please summarise the areas of conflict in separate
email messages to the Responsible Area Director. (It should be in a
separate email because this questionnaire is publicly available.)

No.

(11) Identify any ID nits the Document Shepherd has found in this
document. (See http://www.ietf.org/tools/idnits/ and the Internet-Drafts
Checklist). Boilerplate checks are not enough; this check needs to be
thorough.

None.

(12) Describe how the document meets any required formal review
criteria, such as the MIB Doctor, media type, and URI type reviews.

None.

(13) Have all references within this document been identified as either
normative or informative?

Yes, this OK.

(14) Are there normative references to documents that are not ready for
advancement or are otherwise in an unclear state? If such normative
references exist, what is the plan for their completion?

No external dependencies.

(15) Are there downward normative references references (see RFC 3967)?
If so, list these downward references to support the Area Director in
the Last Call procedure.

No.

(16) Will publication of this document change the status of any existing
RFCs? Are those RFCs listed on the title page header, listed in the
abstract, and discussed in the introduction? If the RFCs are not listed
in the Abstract and Introduction, explain why, and point to the part of
the document where the relationship of this document to the other RFCs
is discussed. If this information is not in the document, explain why
the WG considers it unnecessary.

Updates: 4960 - it is a backwards-compatible SCTP update.

(17) Describe the Document Shepherd's review of the IANA considerations
section, especially with regard to its consistency with the body of the
document. Confirm that all protocol extensions that the document makes
are associated with the appropriate reservations in IANA registries.
Confirm that any referenced IANA registries have been clearly
identified. Confirm that newly created IANA registries include a
detailed specification of the initial contents for the registry, that
allocations procedures for future registrations are defined, and a
reasonable name for the new registry has been suggested (see RFC 5226).

OK, normal SCTP allocation.

(18) List any new IANA registries that require Expert Review for future
allocations. Provide any public guidance that the IESG would find useful
in selecting the IANA Experts for these new registries.

None.

(19) Describe reviews and automated checks performed by the Document
Shepherd to validate sections of the document written in a formal
language, such as XML code, BNF rules, MIB definitions, etc.

None.




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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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On 2013-07-17 20:26, Wesley Eddy wrote:
> It is not wise, in my opinion, to have algorithm-specific DSCPs. The 
> DSCPs need to indicate the desired forwarding behavior, and not what 
> version of code the end-host runs. 
+1

BR

-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================


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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>, Wesley Eddy <wes@mti-systems.com>
Thread-Topic: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <3879D71E758A7E4AA99A35DD8D41D3D91D494900@xmb-rcd-x14.cisco.com> <51E6E1D5.1050006@mti-systems.com> <51F28499.7000604@ericsson.com>
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All,

While I agree with Wes, we all need to ask the following question:

What happens if the action of your congestion control CAUSES you not to hav=
e the PHB you desire under congestion?

I don't think we asked ourselves that question in this way before.

Let me state it a different way. If you have AF4X traffic and your network =
is traffic engineered such that ALL the endpoints putting stuff in AF4X is =
guaranteed (in some sense via CAC or some other bandwidth management arrang=
ement) to fit within the BW allowed for all the AF4X traffic ... then all i=
s OK. That is how a lot of enterprise use of AF4X is designed.

But what if the AF4X traffic comes from endpoints that do not have such a C=
AC/BWM arrangement ... and the endpoints desiring AF4X PHB keep increasing =
their rate until loss occurs. Well, if the AF4X queue is long enough, then =
NONE of the AF4X traffic will have the desired PHB!

That is the rub. The endpoints "want" a certain PHB ... but if there is a b=
ottleneck at a point and the CC of the endpoints all adapt based on loss ..=
. then the traffic won't get the PHB it desires (it will "self-congest" its=
elf).

So while I agree that we ought not to request a new DSCP per new CC protoco=
l/algorithm ... we need to answer this question.

[I know I will regret sending this email.]

Michael Ramalho

-----Original Message-----
From: Zaheduzzaman Sarker [mailto:zaheduzzaman.sarker@ericsson.com]=20
Sent: Friday, July 26, 2013 10:16 AM
To: Wesley Eddy
Cc: Mo Zanaty (mzanaty); Toerless Eckert (eckert); tsvwg@ietf.org; rmcat@ie=
tf.org; Michael Ramalho (mramalho); James Polk (jmpolk); Cheng-Jia Lai (che=
lai)
Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only r=
ate-adaptive for DiffServ


On 2013-07-17 20:26, Wesley Eddy wrote:
> It is not wise, in my opinion, to have algorithm-specific DSCPs. The=20
> DSCPs need to indicate the desired forwarding behavior, and not what=20
> version of code the end-host runs.
+1

BR

--=20

Zahed

=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
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegr=E4nd 11
97128 Lule=E5, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

=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


From swb@internet2.edu  Fri Jul 26 08:52:40 2013
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Date: Fri, 26 Jul 2013 11:52:18 -0400
From: Scott Brim <swb@internet2.edu>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for	DiffServ
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The endpoints want a particular defined service or PDB.  One part of
delivering that is the per-hop behaviors.  They can't deliver a service
alone.

Scott

On 07/26/13 11:07, Michael Ramalho (mramalho) allegedly wrote:
> All,
> 
> While I agree with Wes, we all need to ask the following question:
> 
> What happens if the action of your congestion control CAUSES you not to have the PHB you desire under congestion?
> 
> I don't think we asked ourselves that question in this way before.
> 
> Let me state it a different way. If you have AF4X traffic and your network is traffic engineered such that ALL the endpoints putting stuff in AF4X is guaranteed (in some sense via CAC or some other bandwidth management arrangement) to fit within the BW allowed for all the AF4X traffic ... then all is OK. That is how a lot of enterprise use of AF4X is designed.
> 
> But what if the AF4X traffic comes from endpoints that do not have such a CAC/BWM arrangement ... and the endpoints desiring AF4X PHB keep increasing their rate until loss occurs. Well, if the AF4X queue is long enough, then NONE of the AF4X traffic will have the desired PHB!
> 
> That is the rub. The endpoints "want" a certain PHB ... but if there is a bottleneck at a point and the CC of the endpoints all adapt based on loss ... then the traffic won't get the PHB it desires (it will "self-congest" itself).
> 
> So while I agree that we ought not to request a new DSCP per new CC protocol/algorithm ... we need to answer this question.
> 
> [I know I will regret sending this email.]
> 
> Michael Ramalho
> 
> -----Original Message-----
> From: Zaheduzzaman Sarker [mailto:zaheduzzaman.sarker@ericsson.com] 
> Sent: Friday, July 26, 2013 10:16 AM
> To: Wesley Eddy
> Cc: Mo Zanaty (mzanaty); Toerless Eckert (eckert); tsvwg@ietf.org; rmcat@ietf.org; Michael Ramalho (mramalho); James Polk (jmpolk); Cheng-Jia Lai (chelai)
> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
> 
> 
> On 2013-07-17 20:26, Wesley Eddy wrote:
>> It is not wise, in my opinion, to have algorithm-specific DSCPs. The 
>> DSCPs need to indicate the desired forwarding behavior, and not what 
>> version of code the end-host runs.
> +1
> 
> BR
> 


From brian.e.carpenter@gmail.com  Fri Jul 26 13:54:41 2013
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for DiffServ
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On 27/07/2013 03:52, Scott Brim wrote:
> The endpoints want a particular defined service or PDB.  One part of
> delivering that is the per-hop behaviors.  They can't deliver a service
> alone.

There are two aphorisms that apply here:

1. Queueing theory wins every time.

2. There is no such thing as free lunch.

I'm not trying to be funny. As Michael observes, if there isn't enough
capacity for the offered load, no queueing discipline can fix the problem.
Well-designed PHBs can make things better for a fraction of the traffic,
but only if there's a sufficient background of BE traffic to be dropped
or delayed when necessary.

And if you overload the AF4X queue at a given point, it will inevitably
end up behaving like a BE queue.

That's why the diffserv model includes the need for admission control.

   Brian

> 
> Scott
> 
> On 07/26/13 11:07, Michael Ramalho (mramalho) allegedly wrote:
>> All,
>>
>> While I agree with Wes, we all need to ask the following question:
>>
>> What happens if the action of your congestion control CAUSES you not to have the PHB you desire under congestion?
>>
>> I don't think we asked ourselves that question in this way before.
>>
>> Let me state it a different way. If you have AF4X traffic and your network is traffic engineered such that ALL the endpoints putting stuff in AF4X is guaranteed (in some sense via CAC or some other bandwidth management arrangement) to fit within the BW allowed for all the AF4X traffic ... then all is OK. That is how a lot of enterprise use of AF4X is designed.
>>
>> But what if the AF4X traffic comes from endpoints that do not have such a CAC/BWM arrangement ... and the endpoints desiring AF4X PHB keep increasing their rate until loss occurs. Well, if the AF4X queue is long enough, then NONE of the AF4X traffic will have the desired PHB!
>>
>> That is the rub. The endpoints "want" a certain PHB ... but if there is a bottleneck at a point and the CC of the endpoints all adapt based on loss ... then the traffic won't get the PHB it desires (it will "self-congest" itself).
>>
>> So while I agree that we ought not to request a new DSCP per new CC protocol/algorithm ... we need to answer this question.
>>
>> [I know I will regret sending this email.]
>>
>> Michael Ramalho
>>
>> -----Original Message-----
>> From: Zaheduzzaman Sarker [mailto:zaheduzzaman.sarker@ericsson.com] 
>> Sent: Friday, July 26, 2013 10:16 AM
>> To: Wesley Eddy
>> Cc: Mo Zanaty (mzanaty); Toerless Eckert (eckert); tsvwg@ietf.org; rmcat@ietf.org; Michael Ramalho (mramalho); James Polk (jmpolk); Cheng-Jia Lai (chelai)
>> Subject: Re: [rmcat] [tsvwg] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
>>
>>
>> On 2013-07-17 20:26, Wesley Eddy wrote:
>>> It is not wise, in my opinion, to have algorithm-specific DSCPs. The 
>>> DSCPs need to indicate the desired forwarding behavior, and not what 
>>> version of code the end-host runs.
>> +1
>>
>> BR
>>
> 
> 

From michawe@ifi.uio.no  Fri Jul 26 23:33:54 2013
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs.	loss-only rate-adaptive for DiffServ
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On Jul 26, 2013, at 10:54 PM, Brian E Carpenter =
<brian.e.carpenter@gmail.com> wrote:

[..]

> That's why the diffserv model includes the need for admission control.

+1, exactly my thinking when I read Michael Ramalho's message.

So, with rtcweb planning to use DSCPs just because the application =
decides to do so, rtcweb steps away from this usage of DiffServ, and yes =
you get prioritization between classes, but any form of guarantees for =
go out of the window.

I always assumed that ISPs would protect DiffServ traffic that needs to =
get a certain PHB by having their ingress routers scratch the DSCP for =
all other traffic (e.g. everything that didn't do admission control). If =
they don't, they better do now=85 and that would pretty much eliminate =
the problem, I think: rtcweb would then get priorities applied between =
its own flows wherever it can, and it won't harm anything that needs to =
be protected - in such situations, the network just defaults to putting =
all the rtcweb stuff in the same BE class, and not much harm is being =
done, I think rtcweb was meant for the BE Internet from day 1 anyway.

Problem solved?

Cheers,
Michael


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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 07/27/13 09:35, Kenneth Carlberg allegedly wrote:
> well, be it right or wrong, industry long ago expanded the model so that
> one can have a case of only the application deciding the code point

I don't mind the application deciding the codepoint, but the app still
needs to understand that the network will (should, anyway) do traffic
conditioning, and have a plan for adapting to it.

Scott (at the Pullman)


From dave.taht@gmail.com  Sat Jul 27 11:14:35 2013
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Date: Sat, 27 Jul 2013 11:14:22 -0700
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From: Dave Taht <dave.taht@gmail.com>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On Fri, Jul 26, 2013 at 11:33 PM, Michael Welzl <michawe@ifi.uio.no> wrote:
>
> On Jul 26, 2013, at 10:54 PM, Brian E Carpenter <brian.e.carpenter@gmail.=
com> wrote:
>
> [..]
>
>> That's why the diffserv model includes the need for admission control.
>
> +1, exactly my thinking when I read Michael Ramalho's message.
>
> So, with rtcweb planning to use DSCPs just because the application decide=
s to do so, rtcweb steps away from this usage of DiffServ, and yes you get =
prioritization between classes, but any form of guarantees for go out of th=
e window.
>
> I always assumed that ISPs would protect DiffServ traffic that needs to g=
et a certain PHB by having their ingress routers scratch the DSCP for all o=
ther traffic (e.g. everything that didn't do admission control). If they do=
n't, they better do now=85

What I observe is the only diffserv marking with e2e survivability on
ipv4 is CS1. Everything else gets re-marked. IPv6 has thus far

That said, there is a great deal of internal-to-site traffic that
might benefit, and if the bottleneck gateway from that site is
respecting some markings you have some hope in case of congestion to
the outside.

Where it might matter some more is on wifi, although the current (vs
expected) results of 802.11e, at least on Linux, is dismal, without
admission control. (aggregation works much better)

Given that the only marking I see consistently respected is BE and BK,
I can suggest a couple things for rtcweb - when an browser is running
rtcweb it could try to open less flows than normal for other tasks,
and it could also signal e2e intent by dropping the classification to
CS1 on things like tabs loading in the background.

>and that would pretty much eliminate the problem, I think: rtcweb would th=
en get priorities applied between its own flows wherever it can, and it won=
't harm anything that needs to be protected - in such situations, the netwo=
rk just defaults to putting all the rtcweb stuff in the same BE class, and =
not much harm is being done, I think rtcweb was meant for the BE Internet f=
rom day 1 anyway.

One of the things I like very much about rtcweb is that traffic inside
your own domain - a call from upstairs to downstairs - STAYS inside
your own network, so IF CS4/CS5 classification is used, over wifi, you
would see this traffic drop into the underused hardware wifi VI queue.
Whether or not this would be a good thing would require some
testing...


> Cheers,
> Michael
>



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

From dave.taht@gmail.com  Sat Jul 27 12:18:50 2013
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References: <201307160608.r6G680RY026646@rcdn-core-4.cisco.com> <025A278C-55F2-4EB4-A12F-677F8F01A0D0@netapp.com> <D21571530BF9644D9A443D6BD95B910315595CC7@xmb-rcd-x12.cisco.com> <D21571530BF9644D9A443D6BD95B910315595D97@xmb-rcd-x12.cisco.com> <A860EC86B79FA646BF3F89165A88626415339C25@xmb-aln-x11.cisco.com> <201307162111.r6GLBSwF029514@rcdn-core-5.cisco.com> <3879D71E758A7E4AA99A35DD8D41D3D91D494900@xmb-rcd-x14.cisco.com> <51E6E1D5.1050006@mti-systems.com> <51F28499.7000604@ericsson.com> <D21571530BF9644D9A443D6BD95B9103155BF3CB@xmb-rcd-x12.cisco.com> <51F29B32.3010602@internet2.edu> <51F2E215.4000203@gmail.com> <312469BF-347D-4427-8636-51536549DAC4@ifi.uio.no> <CAA93jw7_vzNaYwuP-dGQ641p-PCq7rpseOm-baW9-3vHkp3V6g@mail.gmail.com>
Date: Sat, 27 Jul 2013 12:18:47 -0700
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From: Dave Taht <dave.taht@gmail.com>
To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On Sat, Jul 27, 2013 at 11:14 AM, Dave Taht <dave.taht@gmail.com> wrote:
> On Fri, Jul 26, 2013 at 11:33 PM, Michael Welzl <michawe@ifi.uio.no> wrot=
e:
>>
>> On Jul 26, 2013, at 10:54 PM, Brian E Carpenter <brian.e.carpenter@gmail=
.com> wrote:
>>
>> [..]
>>
>>> That's why the diffserv model includes the need for admission control.
>>
>> +1, exactly my thinking when I read Michael Ramalho's message.
>>
>> So, with rtcweb planning to use DSCPs just because the application decid=
es to do so, rtcweb steps away from this usage of DiffServ, and yes you get=
 prioritization between classes, but any form of guarantees for go out of t=
he window.
>>
>> I always assumed that ISPs would protect DiffServ traffic that needs to =
get a certain PHB by having their ingress routers scratch the DSCP for all =
other traffic (e.g. everything that didn't do admission control). If they d=
on't, they better do now=85
>
> What I observe is the only diffserv marking with e2e survivability on
> ipv4 is CS1. Everything else gets re-marked. IPv6 has thus far

"escaped much of the same fate." Sorry, thinko. Some experiments with
mosh on ipv4 set to AF42 by default showed a couple middleboxes that
would drop such-marked packets after a few seconds, btw. So I'd argue
that even attempting to use classification should be dynamic and fall
back to BE in case of a failure of connectivity.

JG recently posted a summary of our thinking about latency management
in general which I hope could inform this debate somewhat:

http://gettys.wordpress.com/2013/07/10/low-latency-requires-smart-queuing-t=
raditional-aqm-is-not-enough/



>
> That said, there is a great deal of internal-to-site traffic that
> might benefit, and if the bottleneck gateway from that site is
> respecting some markings you have some hope in case of congestion to
> the outside.
>
> Where it might matter some more is on wifi, although the current (vs
> expected) results of 802.11e, at least on Linux, is dismal, without
> admission control. (aggregation works much better)
>
> Given that the only marking I see consistently respected is BE and BK,
> I can suggest a couple things for rtcweb - when an browser is running
> rtcweb it could try to open less flows than normal for other tasks,
> and it could also signal e2e intent by dropping the classification to
> CS1 on things like tabs loading in the background.
>
>>and that would pretty much eliminate the problem, I think: rtcweb would t=
hen get priorities applied between its own flows wherever it can, and it wo=
n't harm anything that needs to be protected - in such situations, the netw=
ork just defaults to putting all the rtcweb stuff in the same BE class, and=
 not much harm is being done, I think rtcweb was meant for the BE Internet =
from day 1 anyway.
>
> One of the things I like very much about rtcweb is that traffic inside
> your own domain - a call from upstairs to downstairs - STAYS inside
> your own network, so IF CS4/CS5 classification is used, over wifi, you
> would see this traffic drop into the underused hardware wifi VI queue.
> Whether or not this would be a good thing would require some
> testing...
>
>
>> Cheers,
>> Michael
>>
>
>
>
> --
> Dave T=E4ht
>
> Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscrib=
e.html



--=20
Dave T=E4ht

Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscribe.=
html

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Date: Sun, 28 Jul 2013 08:44:19 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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To: Scott Brim <swb@internet2.edu>
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Cc: "tsvwg@ietf.org" <tsvwg@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>, "rmcat@ietf.org" <rmcat@ietf.org>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>, "James Polk \(jmpolk\)" <jmpolk@cisco.com>
Subject: Re: [tsvwg] [rmcat] FW: Fwd: Submitted ID on delay vs. loss-only rate-adaptive for DiffServ
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On 28/07/2013 01:30, Scott Brim wrote:
> On 07/27/13 09:35, Kenneth Carlberg allegedly wrote:
>> well, be it right or wrong, industry long ago expanded the model so that
>> one can have a case of only the application deciding the code point
> 
> I don't mind the application deciding the codepoint, but the app still
> needs to understand that the network will (should, anyway) do traffic
> conditioning, and have a plan for adapting to it.

Indeed, and of course a classifier somewhere downstream can always
override the code point set by the application, if the traffic exits
the initial domain.

    Brian

From david.black@emc.com  Sun Jul 28 05:14:17 2013
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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Sun, 28 Jul 2013 08:13:52 -0400
Thread-Topic: Two problems with draft-geib-tsvwg-diffserv-intercon-03
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Subject: [tsvwg] Two problems with draft-geib-tsvwg-diffserv-intercon-03
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<WG chair hat on>

There are two items that I believe need to be corrected in
draft-geib-tsvwg-diffserv-intercon-03 before it's reasonable to consider th=
at
draft for adoption as an official tsvwg WG draft.

(1) The draft confuses the Class Selector Codepoints (see RFC 2474) with se=
rvice
classes (see RFC 4594).  This paragraph in the introduction illustrates the
problems which occur in numerous places in the draft:

   IP Precedence has been deprecated when DiffServ was standardised.  It
   is common practice today however to copy the DSCPs Bits 0-2 (called
   Class Selector Codepoints in the following) into MPLS TC or Ethernet
   P-Bits.  This is also reflected by the DiffServ codepoint definitions
   of AF and EF.  The Class Selector Codepoints shouldn't be used for
   backward compatibility only.  Class based PHBs may be applied in core
   network sections rather than then DSCP based PHBs.=20

a) The class selector codepoints are *not* DSCP bits 0-2, see RFC 2474.  Th=
e
	attempt to redefine that term in Section 2 is not acceptable.  I realize
	that this draft is defining the class select*ion* codepoints in contrast
	to the class select*or* codepoints in RFC 2474, but I suspect that the
	latter were intended, and if not, those two terms are too close and
	easily confusable.
b) The class selector codepoints are used for more than backwards compatibi=
lity,
	see RFC 4594.
c) "Class based PHBs" means Service Classes, see RFC 4594.  PHBs are not "b=
ased
	on codepoints" - see the specifications of EF and AF for examples or
	RFC 4594.

(2) The discussion of CS6 and routing traffic at the end of Section 4 is
problematic.  This draft sets out to define 4 interconnect classes, and the=
n
effectively adds a fifth based on CS6, but is rather unclear about it, weak=
ening
the entire draft.

I believe we've received enough input from the Routing Area on the importan=
ce
of CS6 for BGP at carrier interconnects that it needs to be dealt with dire=
ctly
in contrast to this draft's approach of admitting that it might be necessar=
y
to use CS6, but discouraging its use.

</WG chair hat on>

<WG chair hat off>

I would also suggest revising the discussion about abstract classes at the =
end
of Section 2 and connecting it to RFC 4594's discussion of service classes =
as
much of this has already been covered there.

</WG chair hat off>

Thanks,
--David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 FAX: +1 (508) 293-778=
6
david.black@emc.com=A0=A0=A0=A0=A0=A0=A0 Mobile: +1 (978) 394-7754
----------------------------------------------------


From david.black@emc.com  Sun Jul 28 05:58:12 2013
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From: "Black, David" <david.black@emc.com>
To: "Shitanshu Shah (svshah)" <svshah@cisco.com>, "idr@ietf.org" <idr@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Sun, 28 Jul 2013 08:57:29 -0400
Thread-Topic: I-D Action: draft-ietf-idr-sla-exchange-01.txt
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Shitanshu,

The use of the IPFIX IANA registry and the TSpec are significant improvemen=
ts.

I noticed one more thing:

The 0x07  =3D DROP_THRESHOLD Classifier Element value in a Traffic Class ap=
pears
to allow mixing of different types of drop thresholds.  That seems like a b=
ad
idea, as I would expect most networks to be managed with a single primary t=
ype
of drop thresholds.  I'd suggest one of two approaches:

	- Put the drop threshold type in the classifier element, and remove it
		from each drop-priority record.
	- Leave the type where it is, and state that mixing types SHOULD NOT be
		done, with a warning that inconsistent and unexpected behavior
		may result if types are mixed because the mappings between types
		may vary by network.

Thanks,
--David


> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Shitanshu Shah (svshah)
> Sent: Thursday, June 27, 2013 10:37 AM
> To: idr@ietf.org; tsvwg@ietf.org
> Subject: [tsvwg] FW: I-D Action: draft-ietf-idr-sla-exchange-01.txt
>=20
>=20
> Dear WG,
>=20
> Following is a revised draft incorporating feed-back/comments from the
> last IETF.
>=20
> Notable changes from the last revision,
> - use of IPFIX IANA registry for Classifier Elements
> - use of Tspec for SLA rates
> - L2_OVERHEAD attribute to allow SLA advertise L3, L2 or customized L2
> rates
>=20
> Please provide comments,
>=20
> Regards,
> Shitanshu
>=20
>=20
> On 6/27/13 7:28 AM, "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> wrote:
>=20
> >
> >A New Internet-Draft is available from the on-line Internet-Drafts
> >directories.
> > This draft is a work item of the Inter-Domain Routing Working Group of
> >the IETF.
> >
> >	Title           : Inter-domain SLA Exchange
> >	Author(s)       : Shitanshu Shah
> >                          Keyur Patel
> >                          Sandeep Bajaj
> >                          Luis Tomotaki
> >                          Mohamed Boucadair
> >	Filename        : draft-ietf-idr-sla-exchange-01.txt
> >	Pages           : 24
> >	Date            : 2013-06-27
> >
> >Abstract:
> >   Network administrators typically provision QoS (Quality of Service)
> >   policies for their application traffic (such as voice, video) based
> >   on SLAs (Service Level Agreements) negotiated with their providers,
> >   and translate those SLAs to vendor specific configuration language.
> >   Both learning of SLA, either thru SLA documents or via some other
> >   out-of-band method, and translating them to vendor specific
> >   configuration language is a complex, many times manual, process and
> >   prone to errors.  This document proposes an in-band method of SLA
> >   signaling which can help to simplify some of the complexities.
> >
> >   This document defines an operational transitive attribute to signal
> >   SLA details in-band, across administrative boundaries (considered as
> >   Autonomous Systems (AS)), thus simplify and speed-up some of the
> >   complex provisioning tasks.
> >
> >   Though the use case with the proposed attribute is explicitly defined
> >   in this document, purpose of this attribute is not limited to this
> >   use case only.
> >
> >
> >The IETF datatracker status page for this draft is:
> >https://datatracker.ietf.org/doc/draft-ietf-idr-sla-exchange
> >
> >There's also a htmlized version available at:
> >http://tools.ietf.org/html/draft-ietf-idr-sla-exchange-01
> >
> >A diff from the previous version is available at:
> >http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-idr-sla-exchange-01
> >
> >
> >Internet-Drafts are also available by anonymous FTP at:
> >ftp://ftp.ietf.org/internet-drafts/
> >
> >_______________________________________________
> >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
>=20


From gorry@erg.abdn.ac.uk  Sun Jul 28 13:00:25 2013
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Date: Sun, 28 Jul 2013 22:00:17 +0200
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] TSVWG Final Agenda -  *FINAL* Request for Speaker Slides!
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There are still some presenters listed in the Agenda who have not yet 
sent us slides.

Could you please send me any slides you wish to present asap? - If I 
don't receive slides by 7:00am tomorrow, I'll assume you do not need to 
present  (some others will appreciate the extra time to discuss other 
drafts!)

Gorry


-------- Original Message --------

We have now uploaded the final Agenda for the meeting on Monday:
http://www.ietf.org/proceedings/87/agenda/agenda-87-tsvwg

There are two TSVWG sessions, the second will include a group of drafts
supporting transport functions that may be applicable to RMCAT.

People intending to present should note that the allocated time is
expected to include time for questions. You also must send the chairs
slides by Sunday 17:00.

NOTE 1 - Because we have a 9:00 start on Monday, you *DO* need to send
copies of your slides in advance!

NOTE 2 - We will need a minute-taker. In the past few meetings we used
valuable meeting time to get volunteers, if some people volunteer in
advance it would be appreciated by all!

Gorry, David & James.




From brian.e.carpenter@gmail.com  Sun Jul 28 13:07:27 2013
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Date: Mon, 29 Jul 2013 08:07:13 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] Two problems with draft-geib-tsvwg-diffserv-intercon-03
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On 29/07/2013 00:13, Black, David wrote:
> <WG chair hat on>
> 
> There are two items that I believe need to be corrected in
> draft-geib-tsvwg-diffserv-intercon-03 before it's reasonable to consider that
> draft for adoption as an official tsvwg WG draft.
> 
> (1) The draft confuses the Class Selector Codepoints (see RFC 2474) with service
> classes (see RFC 4594).  This paragraph in the introduction illustrates the
> problems which occur in numerous places in the draft:
> 
>    IP Precedence has been deprecated when DiffServ was standardised.  It
>    is common practice today however to copy the DSCPs Bits 0-2 (called
>    Class Selector Codepoints in the following) into MPLS TC or Ethernet
>    P-Bits.  This is also reflected by the DiffServ codepoint definitions
>    of AF and EF.

It's a fact that the CS PHBs (with mandatory 6-bit DSCPs) and the EF
and AF PHBs (with  recommended 6-bit DSCPs) were artfully chosen such that
if they were interpreted by a legacy router supporting 3-bit IP Precedence,
something reasonable would happen. An unintended side-effect of that
is that if bits 0-2 are mapped into a 3-bit priority field in another
technology, something reasonable will happen. But *that's all*. It's
a logical error to assert anything else. I fully agree with David's
comments below.

>    The Class Selector Codepoints shouldn't be used for
>    backward compatibility only.  Class based PHBs may be applied in core
>    network sections rather than then DSCP based PHBs. 
> 
> a) The class selector codepoints are *not* DSCP bits 0-2, see RFC 2474.  The
> 	attempt to redefine that term in Section 2 is not acceptable.  I realize
> 	that this draft is defining the class select*ion* codepoints in contrast
> 	to the class select*or* codepoints in RFC 2474, but I suspect that the
> 	latter were intended, and if not, those two terms are too close and
> 	easily confusable.
> b) The class selector codepoints are used for more than backwards compatibility,
> 	see RFC 4594.
> c) "Class based PHBs" means Service Classes, see RFC 4594.  PHBs are not "based
> 	on codepoints" - see the specifications of EF and AF for examples or
> 	RFC 4594.
> 
> (2) The discussion of CS6 and routing traffic at the end of Section 4 is
> problematic.  This draft sets out to define 4 interconnect classes, and then
> effectively adds a fifth based on CS6, but is rather unclear about it, weakening
> the entire draft.
> 
> I believe we've received enough input from the Routing Area on the importance
> of CS6 for BGP at carrier interconnects that it needs to be dealt with directly
> in contrast to this draft's approach of admitting that it might be necessary
> to use CS6, but discouraging its use.

Absolutely. That's been true for 20+ years, too.

   Brian

> 
> </WG chair hat on>
> 
> <WG chair hat off>
> 
> I would also suggest revising the discussion about abstract classes at the end
> of Section 2 and connecting it to RFC 4594's discussion of service classes as
> much of this has already been covered there.
> 
> </WG chair hat off>
> 
> Thanks,
> --David
> ----------------------------------------------------
> David L. Black, Distinguished Engineer
> EMC Corporation, 176 South St., Hopkinton, MA  01748
> +1 (508) 293-7953             FAX: +1 (508) 293-7786
> david.black@emc.com        Mobile: +1 (978) 394-7754
> ----------------------------------------------------
> 
> 

From Karen.Nielsen@tieto.com  Sun Jul 28 23:08:03 2013
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>> Side note: A single drop of a data packet on a sparsely loaded path
>> (or in data tail - not enough data for Fast Recovery to kick in) today
>> also results in CWND reduction and this of course compares with the loss=
 of HB
>situation.
>> But I think that this is a problem, rather than a desired effect, in
>> the present algorithms that one continuously tries to find solutions
>> for. We should not make this problem worse by also introducing if for
>> single drop of HBs. I think.
>
>I also think we need something here as single drop of HB might not be an i=
ndication
>of severe congestion.
>I'm thinking that something like draft-ietf-tcpm-newcwv might need to be
>considered.
>

Yes, I think that it is relevant to consider to complement or replace the e=
xponential decay of the=20
CWND of SCTP with the mechanisms of draft-ietf-tcmp-newcwv. However, I see =
this as a more general issue.
The problem space here, I believe, is more in the area of F_RTO, TPL and al=
ikes.
Namely to avoid "unnecessary" T3-timeouts and resulting undesirable effects=
 in terms of drastic CWND reduction
and, for SCTP, path failover.

For now, in this draft, still the best may simply be to stick with RFC4960 =
operation and do nothing special with
the CWND in the PF state. Then more advanced handling in PF states could co=
me in a follow up ?

Thanks,

Karen

>Thanks,
>--
>Yoshifumi

From internet-drafts@ietf.org  Mon Jul 29 05:27:39 2013
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-06.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

	Title           : Generic Aggregation of Resource ReSerVation Protocol (RS=
VP) for IPv4 And IPv6 Reservations over PCN domains
	Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
	Filename        : draft-ietf-tsvwg-rsvp-pcn-06.txt
	Pages           : 27
	Date            : 2013-07-29

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-06


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-06.txt
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Hi all,



After working out the comments of Bob we have just submitted v06, see below=
!



In order to see the differences between v05 and v06, please see:

http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-06



Best regards,

Georgios

________________________________
Van: tsvwg-bounces@ietf.org [tsvwg-bounces@ietf.org] namens internet-drafts=
@ietf.org [internet-drafts@ietf.org]
Verzonden: maandag 29 juli 2013 14:27
To: i-d-announce@ietf.org
Cc: tsvwg@ietf.org
Onderwerp: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-06.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Transport Area Working Group Working Grou=
p of the IETF.

        Title           : Generic Aggregation of Resource ReSerVation Proto=
col (RSVP) for IPv4 And IPv6 Reservations over PCN domains
        Author(s)       : Georgios Karagiannis
                          Anurag Bhargava
        Filename        : draft-ietf-tsvwg-rsvp-pcn-06.txt
        Pages           : 27
        Date            : 2013-07-29

Abstract:
   This document specifies extensions to Generic Aggregated RSVP
   [RFC4860] for support of the PCN Controlled Load (CL) and Single
   Marking (SM) edge behaviors over a Diffserv cloud using Pre-
   Congestion Notification.





The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-06


Please note that it may take a couple of minutes from the time of submissio=
n
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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E: 10pt">
<p>Hi all,</p>
<p>&nbsp;</p>
<p>After working out the comments of Bob we have just submitted v06, see be=
low!</p>
<p>&nbsp;</p>
<p>In order to see the differences between v05 and v06, please see:</p>
<p><a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-=
06" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-r=
svp-pcn-06</a><br>
</p>
<p>&nbsp;</p>
<p>Best regards,</p>
<p>Georgios&nbsp;</p>
<div>
<hr tabindex=3D"-1">
<div id=3D"x_divRplyFwdMsg"><font color=3D"#000000" size=3D"2" face=3D"Taho=
ma"><b>Van:</b> tsvwg-bounces@ietf.org [tsvwg-bounces@ietf.org] namens inte=
rnet-drafts@ietf.org [internet-drafts@ietf.org]<br>
<b>Verzonden:</b> maandag 29 juli 2013 14:27<br>
<b>To:</b> i-d-announce@ietf.org<br>
<b>Cc:</b> tsvwg@ietf.org<br>
<b>Onderwerp:</b> [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-pcn-06.txt<br>
</font><br>
</div>
<div></div>
</div>
<font size=3D"2"><span style=3D"FONT-SIZE: 10pt">
<div class=3D"PlainText"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
&nbsp;This draft is a work item of the Transport Area Working Group Working=
 Group of the IETF.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Generic Aggregation of Resource ReSerVa=
tion Protocol (RSVP) for IPv4 And IPv6 Reservations over PCN domains<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp; : Georgios Karagiannis<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 Anurag Bhargava<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; : draft-ietf-tsvwg-rsvp-pcn-06.txt<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 27<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2013-07-29<br>
<br>
Abstract:<br>
&nbsp;&nbsp; This document specifies extensions to Generic Aggregated RSVP<=
br>
&nbsp;&nbsp; [RFC4860] for support of the PCN Controlled Load (CL) and Sing=
le<br>
&nbsp;&nbsp; Marking (SM) edge behaviors over a Diffserv cloud using Pre-<b=
r>
&nbsp;&nbsp; Congestion Notification.<br>
<br>
<br>
<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-pcn</a=
><br>
<br>
There's also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-06" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-pcn-06</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp-pcn-06"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rsvp=
-pcn-06</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at tools.ietf.org.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
</div>
</span></font></div>
</body>
</html>

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From svshah@cisco.com  Mon Jul 29 07:23:34 2013
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From: "Shitanshu Shah (svshah)" <svshah@cisco.com>
To: "Black, David" <david.black@emc.com>, "idr@ietf.org" <idr@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: I-D Action: draft-ietf-idr-sla-exchange-01.txt
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Subject: Re: [tsvwg] I-D Action: draft-ietf-idr-sla-exchange-01.txt
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Thank you David for your review and comments..

Regarding drop thresholds, indeed intention is to provide a facility to
take multiple (type-value, burst) pairs but for a single primary type.
Agreed, current definition in the draft is loose to interpret eligible use
of multiple types for drop thresholds.

I'll modify the structure for drop thresholds to take type definition once
with multiple (type-value, burst) pairs.

Regards,
Shitanshu

On 7/28/13 5:57 AM, "Black, David" <david.black@emc.com> wrote:

>
>Shitanshu,
>
>The use of the IPFIX IANA registry and the TSpec are significant
>improvements.
>
>I noticed one more thing:
>
>The 0x07  =3D DROP_THRESHOLD Classifier Element value in a Traffic Class
>appears
>to allow mixing of different types of drop thresholds.  That seems like a
>bad
>idea, as I would expect most networks to be managed with a single primary
>type
>of drop thresholds.  I'd suggest one of two approaches:
>
>	- Put the drop threshold type in the classifier element, and remove it
>		from each drop-priority record.
>	- Leave the type where it is, and state that mixing types SHOULD NOT be
>		done, with a warning that inconsistent and unexpected behavior
>		may result if types are mixed because the mappings between types
>		may vary by network.
>
>Thanks,
>--David
>
>
>> -----Original Message-----
>> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf
>>Of
>> Shitanshu Shah (svshah)
>> Sent: Thursday, June 27, 2013 10:37 AM
>> To: idr@ietf.org; tsvwg@ietf.org
>> Subject: [tsvwg] FW: I-D Action: draft-ietf-idr-sla-exchange-01.txt
>>=20
>>=20
>> Dear WG,
>>=20
>> Following is a revised draft incorporating feed-back/comments from the
>> last IETF.
>>=20
>> Notable changes from the last revision,
>> - use of IPFIX IANA registry for Classifier Elements
>> - use of Tspec for SLA rates
>> - L2_OVERHEAD attribute to allow SLA advertise L3, L2 or customized L2
>> rates
>>=20
>> Please provide comments,
>>=20
>> Regards,
>> Shitanshu
>>=20
>>=20
>> On 6/27/13 7:28 AM, "internet-drafts@ietf.org"
>><internet-drafts@ietf.org>
>> wrote:
>>=20
>> >
>> >A New Internet-Draft is available from the on-line Internet-Drafts
>> >directories.
>> > This draft is a work item of the Inter-Domain Routing Working Group of
>> >the IETF.
>> >
>> >	Title           : Inter-domain SLA Exchange
>> >	Author(s)       : Shitanshu Shah
>> >                          Keyur Patel
>> >                          Sandeep Bajaj
>> >                          Luis Tomotaki
>> >                          Mohamed Boucadair
>> >	Filename        : draft-ietf-idr-sla-exchange-01.txt
>> >	Pages           : 24
>> >	Date            : 2013-06-27
>> >
>> >Abstract:
>> >   Network administrators typically provision QoS (Quality of Service)
>> >   policies for their application traffic (such as voice, video) based
>> >   on SLAs (Service Level Agreements) negotiated with their providers,
>> >   and translate those SLAs to vendor specific configuration language.
>> >   Both learning of SLA, either thru SLA documents or via some other
>> >   out-of-band method, and translating them to vendor specific
>> >   configuration language is a complex, many times manual, process and
>> >   prone to errors.  This document proposes an in-band method of SLA
>> >   signaling which can help to simplify some of the complexities.
>> >
>> >   This document defines an operational transitive attribute to signal
>> >   SLA details in-band, across administrative boundaries (considered as
>> >   Autonomous Systems (AS)), thus simplify and speed-up some of the
>> >   complex provisioning tasks.
>> >
>> >   Though the use case with the proposed attribute is explicitly
>>defined
>> >   in this document, purpose of this attribute is not limited to this
>> >   use case only.
>> >
>> >
>> >The IETF datatracker status page for this draft is:
>> >https://datatracker.ietf.org/doc/draft-ietf-idr-sla-exchange
>> >
>> >There's also a htmlized version available at:
>> >http://tools.ietf.org/html/draft-ietf-idr-sla-exchange-01
>> >
>> >A diff from the previous version is available at:
>> >http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-idr-sla-exchange-01
>> >
>> >
>> >Internet-Drafts are also available by anonymous FTP at:
>> >ftp://ftp.ietf.org/internet-drafts/
>> >
>> >_______________________________________________
>> >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
>>=20
>


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To: "Black, David" <david.black@emc.com>, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
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Subject: Re: [tsvwg] resolution with draft-ietf-tsvwg-rsvp-pcn
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David, Gorry,

Just to confirm that
- we resolved the outstanding issue over=20
aggregation over lunch - I was concerned it would=20
delay response to flash crowds, but it was a=20
misunderstanding on my part - so no change needed to the draft on this=
 count.
- the deltas posted in -06 this afternoon do what=20
I asked (not sending PCN-specific error messages=20
to end-systems). My original review requested=20
this, and Georgios had tried to comply, but there=20
were some places in the doc that still said the=20
old stuff (incl in IANA Considerations). Fixed now.

So I'm OK with WGLC - and I'll try to give it one last read over for that.


Bob

At 12:35 29/07/2013, Black, David wrote:
>Not a problem =96 we ran ahead of time on other=20
>agenda items and decided to dispense w/the=20
>rsvp-pcn draft in the morning in order to free=20
>up some time this afternoon =96 Anurag was there,=20
>and everything=92s in fine shape wrt this draft=20
>now.  I look forward to seeing the revised version.
>
>Thanks,
>--David
>
>From: karagian@cs.utwente.nl [mailto:karagian@cs.utwente.nl]
>Sent: Monday, July 29, 2013 7:31 AM
>To: Black, David; bhargava.anurag@gmail.com;=20
>gorry@erg.abdn.ac.uk; bob.briscoe@bt.com; anuragb@cisco.com
>Cc: jmpolk@cisco.com
>Subject: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
>
>
>Hi all,
>
>
>
>Thanks very much! As already mentined by Anurag=20
>we will make the requested changes by Bob and=20
>upload the new version before tomorrow morning!
>
>Sorry for not attending the tsvwg meeting, but I=20
>had to attend another meeting  at that time! I=20
>thought that the rsvp-pcn draft will be presented during the afternoon=
 slot!
>
>
>
>Best regards,
>
>Georgios
>
>----------
>Van: Black, David [david.black@emc.com]
>Verzonden: maandag 29 juli 2013 13:04
>To: Anurag Bhargava;=20
><mailto:gorry@erg.abdn.ac.uk>gorry@erg.abdn.ac.uk;=20
><mailto:bob.briscoe@bt.com>bob.briscoe@bt.com;=20
>Karagiannis, G. (EWI); <mailto:anuragb@cisco.com>anuragb@cisco.com
>Cc: James Polk (jmpolk)=20
>(<mailto:jmpolk@cisco.com>jmpolk@cisco.com); Black, David
>Onderwerp: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
>Excellent =96 that sounds like a plan.
>
>Thanks,
>--David
>
>From: Anurag Bhargava=20
>[<mailto:bhargava.anurag@gmail.com>mailto:bhargava.anurag@gmail.com]
>Sent: Monday, July 29, 2013 6:57 AM
>To:=20
><mailto:gorry@erg.abdn.ac.uk>gorry@erg.abdn.ac.uk;=20
>Black, David;=20
><mailto:bob.briscoe@bt.com>bob.briscoe@bt.com;=20
>Georgios Karagiannis; <mailto:anuragb@cisco.com>anuragb@cisco.com
>Subject: Re: resolution with draft-ietf-tsvwg-rsvp-pcn
>
>Hi Gorry/David,
>
>   Bob, Georgios and I just had a discussion on=20
> the draft. Bob is quite comfortable now and=20
> have asked us to make a couple of changes.
>We will be doing the change today and will post=20
>a new version later today or tomorrow morning.
>After we post the new version, please see if=20
>WGLC can be started (that will put pressure on the folks to review :)).
>
>--
>BR,
>-Anurag

________________________________________________________________
Bob Briscoe,                                                  BT=20
--=====================_1149764013==.ALT
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<body>
David, Gorry,<br><br>
Just to confirm that <br>
- we resolved the outstanding issue over aggregation over lunch - I was
concerned it would delay response to flash crowds, but it was a
misunderstanding on my part - so no change needed to the draft on this
count.<br>
- the deltas posted in -06 this afternoon do what I asked (not sending
PCN-specific error messages to end-systems). My original review requested
this, and Georgios had tried to comply, but there were some places in the
doc that still said the old stuff (incl in IANA Considerations). Fixed
now.<br><br>
So I'm OK with WGLC - and I'll try to give it one last read over for
that.<br><br>
<br>
Bob<br><br>
At 12:35 29/07/2013, Black, David wrote:<br>
<blockquote type=3Dcite class=3Dcite cite=3D"">Not a problem =96 we ran=
 ahead of
time on other agenda items and decided to dispense w/the rsvp-pcn draft
in the morning in order to free up some time this afternoon =96 Anurag was
there, and everything=92s in fine shape wrt this draft now.&nbsp; I look
forward to seeing the revised version.<br>
&nbsp;<br>
Thanks,<br>
--David<br>
&nbsp;<br>
<b>From:</b> karagian@cs.utwente.nl
[<a href=3D"mailto:karagian@cs.utwente.nl" eudora=3D"autourl">
mailto:karagian@cs.utwente.nl</a>] <br>
<b>Sent:</b> Monday, July 29, 2013 7:31 AM<br>
<b>To:</b> Black, David; bhargava.anurag@gmail.com; gorry@erg.abdn.ac.uk;
bob.briscoe@bt.com; anuragb@cisco.com<br>
<b>Cc:</b> jmpolk@cisco.com<br>
<b>Subject:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<br>
&nbsp;<br><br>
Hi all,<br><br>
&nbsp;<br><br>
Thanks very much! As already mentined by Anurag we will make the
requested changes by Bob and upload the new version before tomorrow
morning!<br><br>
Sorry for not attending the tsvwg meeting, but I had to attend another
meeting&nbsp; at that time! I thought that the rsvp-pcn draft will be
presented during the afternoon slot!<br><br>
&nbsp;<br><br>
Best regards,<br><br>
Georgios<br>
<hr>
<div align=3D"center"></div>
<b>Van:</b> Black, David [david.black@emc.com]<br>
<b>Verzonden:</b> maandag 29 juli 2013 13:04<br>
<b>To:</b> Anurag Bhargava;
<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>;
<a href=3D"mailto:bob.briscoe@bt.com">bob.briscoe@bt.com</a>; Karagiannis,
G. (EWI); <a href=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a><br>
<b>Cc:</b> James Polk (jmpolk)
(<a href=3D"mailto:jmpolk@cisco.com">jmpolk@cisco.com</a>); Black,
David<br>
<b>Onderwerp:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<br>
Excellent =96 that sounds like a plan.<br>
&nbsp;<br>
Thanks,<br>
--David<br>
&nbsp;<br>
<b>From:</b> Anurag Bhargava
[<a href=3D"mailto:bhargava.anurag@gmail.com">
mailto:bhargava.anurag@gmail.com</a>] <br>
<b>Sent:</b> Monday, July 29, 2013 6:57 AM<br>
<b>To:</b>
<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>; Black,
David; <a href=3D"mailto:bob.briscoe@bt.com">bob.briscoe@bt.com</a>;
Georgios Karagiannis;
<a href=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a><br>
<b>Subject:</b> Re: resolution with draft-ietf-tsvwg-rsvp-pcn<br>
&nbsp;<br>
Hi Gorry/David,<br>
&nbsp;<br>
&nbsp; Bob, Georgios and I just had a discussion on the draft. Bob is
quite comfortable now and have asked us to make a couple of changes.<br>
We will be doing the change today and will post a new version later today
or tomorrow morning.<br>
After we post the new version, please see if WGLC can be started (that
will put pressure on the folks to review :)).<br><br>
-- <br>
BR,<br>
-Anurag </blockquote>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_1149764013==.ALT--

From david.black@emc.com  Mon Jul 29 11:20:00 2013
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From: "Black, David" <david.black@emc.com>
To: Bob Briscoe <bob.briscoe@bt.com>, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Date: Mon, 29 Jul 2013 14:18:51 -0400
Thread-Topic: resolution with draft-ietf-tsvwg-rsvp-pcn
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Subject: Re: [tsvwg] resolution with draft-ietf-tsvwg-rsvp-pcn
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Bob,

Thank you for following up on this.

I've also taken a closer look at the two text changes reported on the rsvp-=
pcn draft slides used today, and they both look fine to me (one was wrt a c=
omment of mine).  Gorry and I will consult on getting the WGLC started for =
this draft.

Thanks,
--David

From: Bob Briscoe [mailto:bob.briscoe@bt.com]
Sent: Monday, July 29, 2013 1:37 PM
To: Black, David; gorry@erg.abdn.ac.uk
Cc: karagian@cs.utwente.nl; bhargava.anurag@gmail.com; anuragb@cisco.com; j=
mpolk@cisco.com; Black, David; tsvwg IETF list
Subject: RE: resolution with draft-ietf-tsvwg-rsvp-pcn

David, Gorry,

Just to confirm that
- we resolved the outstanding issue over aggregation over lunch - I was con=
cerned it would delay response to flash crowds, but it was a misunderstandi=
ng on my part - so no change needed to the draft on this count.
- the deltas posted in -06 this afternoon do what I asked (not sending PCN-=
specific error messages to end-systems). My original review requested this,=
 and Georgios had tried to comply, but there were some places in the doc th=
at still said the old stuff (incl in IANA Considerations). Fixed now.

So I'm OK with WGLC - and I'll try to give it one last read over for that.


Bob

At 12:35 29/07/2013, Black, David wrote:

Not a problem - we ran ahead of time on other agenda items and decided to d=
ispense w/the rsvp-pcn draft in the morning in order to free up some time t=
his afternoon - Anurag was there, and everything's in fine shape wrt this d=
raft now.  I look forward to seeing the revised version.

Thanks,
--David

From: karagian@cs.utwente.nl<mailto:karagian@cs.utwente.nl> [ mailto:karagi=
an@cs.utwente.nl]
Sent: Monday, July 29, 2013 7:31 AM
To: Black, David; bhargava.anurag@gmail.com<mailto:bhargava.anurag@gmail.co=
m>; gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>; bob.briscoe@bt.com<m=
ailto:bob.briscoe@bt.com>; anuragb@cisco.com<mailto:anuragb@cisco.com>
Cc: jmpolk@cisco.com<mailto:jmpolk@cisco.com>
Subject: RE: resolution with draft-ietf-tsvwg-rsvp-pcn


Hi all,



Thanks very much! As already mentined by Anurag we will make the requested =
changes by Bob and upload the new version before tomorrow morning!

Sorry for not attending the tsvwg meeting, but I had to attend another meet=
ing  at that time! I thought that the rsvp-pcn draft will be presented duri=
ng the afternoon slot!



Best regards,

Georgios
________________________________
Van: Black, David [david.black@emc.com]
Verzonden: maandag 29 juli 2013 13:04
To: Anurag Bhargava; gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>; bob=
.briscoe@bt.com<mailto:bob.briscoe@bt.com>; Karagiannis, G. (EWI); anuragb@=
cisco.com<mailto:anuragb@cisco.com>
Cc: James Polk (jmpolk) (jmpolk@cisco.com<mailto:jmpolk@cisco.com>); Black,=
 David
Onderwerp: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
Excellent - that sounds like a plan.

Thanks,
--David

From: Anurag Bhargava [ mailto:bhargava.anurag@gmail.com]
Sent: Monday, July 29, 2013 6:57 AM
To: gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>; Black, David; bob.br=
iscoe@bt.com<mailto:bob.briscoe@bt.com>; Georgios Karagiannis; anuragb@cisc=
o.com<mailto:anuragb@cisco.com>
Subject: Re: resolution with draft-ietf-tsvwg-rsvp-pcn

Hi Gorry/David,

  Bob, Georgios and I just had a discussion on the draft. Bob is quite comf=
ortable now and have asked us to make a couple of changes.
We will be doing the change today and will post a new version later today o=
r tomorrow morning.
After we post the new version, please see if WGLC can be started (that will=
 put pressure on the folks to review :)).

--
BR,
-Anurag

________________________________________________________________
Bob Briscoe,                                                  BT

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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'f=
ont-size:10.0pt;font-family:"Courier New";color:black'>Bob,<o:p></o:p></spa=
n></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Cou=
rier New";color:black'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>Thank y=
ou for following up on this.<o:p></o:p></span></p><p class=3DMsoNormal><spa=
n style=3D'font-size:10.0pt;font-family:"Courier New";color:black'><o:p>&nb=
sp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Courier New";color:black'>I&#8217;ve also taken a closer look at=
 the two text changes reported on the rsvp-pcn draft slides used today, and=
 they both look fine to me (one was wrt a comment of mine).&nbsp; Gorry and=
 I will consult on getting the WGLC started for this draft.<o:p></o:p></spa=
n></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Cou=
rier New";color:black'><o:p>&nbsp;</o:p></span></p><div><div><p class=3DMso=
Normal><span style=3D'font-size:10.0pt;font-family:"Courier New";color:blac=
k'>Thanks,<br>--David</span><span style=3D'font-size:11.0pt;font-family:"Co=
urier New";color:black'><o:p></o:p></span></p></div></div><p class=3DMsoNor=
mal><span style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>=
<o:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-left:solid blu=
e 1.5pt;padding:0in 0in 0in 4.0pt'><div><div style=3D'border:none;border-to=
p:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><s=
pan style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</spa=
n></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> B=
ob Briscoe [mailto:bob.briscoe@bt.com] <br><b>Sent:</b> Monday, July 29, 20=
13 1:37 PM<br><b>To:</b> Black, David; gorry@erg.abdn.ac.uk<br><b>Cc:</b> k=
aragian@cs.utwente.nl; bhargava.anurag@gmail.com; anuragb@cisco.com; jmpolk=
@cisco.com; Black, David; tsvwg IETF list<br><b>Subject:</b> RE: resolution=
 with draft-ietf-tsvwg-rsvp-pcn<o:p></o:p></span></p></div></div><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>David, Gorry,<br><br=
>Just to confirm that <br>- we resolved the outstanding issue over aggregat=
ion over lunch - I was concerned it would delay response to flash crowds, b=
ut it was a misunderstanding on my part - so no change needed to the draft =
on this count.<br>- the deltas posted in -06 this afternoon do what I asked=
 (not sending PCN-specific error messages to end-systems). My original revi=
ew requested this, and Georgios had tried to comply, but there were some pl=
aces in the doc that still said the old stuff (incl in IANA Considerations)=
. Fixed now.<br><br>So I'm OK with WGLC - and I'll try to give it one last =
read over for that.<br><br><br>Bob<br><br>At 12:35 29/07/2013, Black, David=
 wrote:<br><br><o:p></o:p></p><p class=3DMsoNormal>Not a problem &#8211; we=
 ran ahead of time on other agenda items and decided to dispense w/the rsvp=
-pcn draft in the morning in order to free up some time this afternoon &#82=
11; Anurag was there, and everything&#8217;s in fine shape wrt this draft n=
ow.&nbsp; I look forward to seeing the revised version.<br>&nbsp;<br>Thanks=
,<br>--David<br>&nbsp;<br><b>From:</b> <a href=3D"mailto:karagian@cs.utwent=
e.nl">karagian@cs.utwente.nl</a> [<a href=3D"mailto:karagian@cs.utwente.nl"=
> mailto:karagian@cs.utwente.nl</a>] <br><b>Sent:</b> Monday, July 29, 2013=
 7:31 AM<br><b>To:</b> Black, David; <a href=3D"mailto:bhargava.anurag@gmai=
l.com">bhargava.anurag@gmail.com</a>; <a href=3D"mailto:gorry@erg.abdn.ac.u=
k">gorry@erg.abdn.ac.uk</a>; <a href=3D"mailto:bob.briscoe@bt.com">bob.bris=
coe@bt.com</a>; <a href=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a><=
br><b>Cc:</b> <a href=3D"mailto:jmpolk@cisco.com">jmpolk@cisco.com</a><br><=
b>Subject:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<br>&nbsp;<br><=
br>Hi all,<br><br>&nbsp;<br><br>Thanks very much! As already mentined by An=
urag we will make the requested changes by Bob and upload the new version b=
efore tomorrow morning!<br><br>Sorry for not attending the tsvwg meeting, b=
ut I had to attend another meeting&nbsp; at that time! I thought that the r=
svp-pcn draft will be presented during the afternoon slot!<br><br>&nbsp;<br=
><br>Best regards,<br><br>Georgios<o:p></o:p></p><div class=3DMsoNormal ali=
gn=3Dcenter style=3D'text-align:center'><hr size=3D2 width=3D"100%" align=
=3Dcenter></div><p class=3DMsoNormal><b>Van:</b> Black, David [david.black@=
emc.com]<br><b>Verzonden:</b> maandag 29 juli 2013 13:04<br><b>To:</b> Anur=
ag Bhargava; <a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</=
a>; <a href=3D"mailto:bob.briscoe@bt.com">bob.briscoe@bt.com</a>; Karagiann=
is, G. (EWI); <a href=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a><br=
><b>Cc:</b> James Polk (jmpolk) (<a href=3D"mailto:jmpolk@cisco.com">jmpolk=
@cisco.com</a>); Black, David<br><b>Onderwerp:</b> RE: resolution with draf=
t-ietf-tsvwg-rsvp-pcn<br>Excellent &#8211; that sounds like a plan.<br>&nbs=
p;<br>Thanks,<br>--David<br>&nbsp;<br><b>From:</b> Anurag Bhargava [<a href=
=3D"mailto:bhargava.anurag@gmail.com"> mailto:bhargava.anurag@gmail.com</a>=
] <br><b>Sent:</b> Monday, July 29, 2013 6:57 AM<br><b>To:</b> <a href=3D"m=
ailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>; Black, David; <a href=
=3D"mailto:bob.briscoe@bt.com">bob.briscoe@bt.com</a>; Georgios Karagiannis=
; <a href=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a><br><b>Subject:=
</b> Re: resolution with draft-ietf-tsvwg-rsvp-pcn<br>&nbsp;<br>Hi Gorry/Da=
vid,<br>&nbsp;<br>&nbsp; Bob, Georgios and I just had a discussion on the d=
raft. Bob is quite comfortable now and have asked us to make a couple of ch=
anges.<br>We will be doing the change today and will post a new version lat=
er today or tomorrow morning.<br>After we post the new version, please see =
if WGLC can be started (that will put pressure on the folks to review :)).<=
br><br>-- <br>BR,<br>-Anurag <o:p></o:p></p><p>____________________________=
____________________________________<br>Bob Briscoe,&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;&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; BT<o:p></o:p></p></div></div=
></body></html>=

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From: bhargava.anurag@gmail.com
Date: Tue, 30 Jul 2013 01:42:06 +0200
To: "Black, David" <david.black@emc.com>
Cc: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, tsvwg IETF list <tsvwg@ietf.org>, "anuragb@cisco.com" <anuragb@cisco.com>, "jmpolk@cisco.com" <jmpolk@cisco.com>
Subject: Re: [tsvwg] resolution with draft-ietf-tsvwg-rsvp-pcn
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Thanks all.

Br,
Anurag

Sent from my iPhone

On Jul 29, 2013, at 8:18 PM, "Black, David" <david.black@emc.com> wrote:

> Bob,
> =20
> Thank you for following up on this.
> =20
> I=E2=80=99ve also taken a closer look at the two text changes reported on t=
he rsvp-pcn draft slides used today, and they both look fine to me (one was w=
rt a comment of mine).  Gorry and I will consult on getting the WGLC started=
 for this draft.
> =20
> Thanks,
> --David
> =20
> From: Bob Briscoe [mailto:bob.briscoe@bt.com]=20
> Sent: Monday, July 29, 2013 1:37 PM
> To: Black, David; gorry@erg.abdn.ac.uk
> Cc: karagian@cs.utwente.nl; bhargava.anurag@gmail.com; anuragb@cisco.com; j=
mpolk@cisco.com; Black, David; tsvwg IETF list
> Subject: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
> =20
> David, Gorry,
>=20
> Just to confirm that=20
> - we resolved the outstanding issue over aggregation over lunch - I was co=
ncerned it would delay response to flash crowds, but it was a misunderstandi=
ng on my part - so no change needed to the draft on this count.
> - the deltas posted in -06 this afternoon do what I asked (not sending PCN=
-specific error messages to end-systems). My original review requested this,=
 and Georgios had tried to comply, but there were some places in the doc tha=
t still said the old stuff (incl in IANA Considerations). Fixed now.
>=20
> So I'm OK with WGLC - and I'll try to give it one last read over for that.=

>=20
>=20
> Bob
>=20
> At 12:35 29/07/2013, Black, David wrote:
>=20
> Not a problem =E2=80=93 we ran ahead of time on other agenda items and dec=
ided to dispense w/the rsvp-pcn draft in the morning in order to free up som=
e time this afternoon =E2=80=93 Anurag was there, and everything=E2=80=99s i=
n fine shape wrt this draft now.  I look forward to seeing the revised versi=
on.
> =20
> Thanks,
> --David
> =20
> From: karagian@cs.utwente.nl [ mailto:karagian@cs.utwente.nl]=20
> Sent: Monday, July 29, 2013 7:31 AM
> To: Black, David; bhargava.anurag@gmail.com; gorry@erg.abdn.ac.uk; bob.bri=
scoe@bt.com; anuragb@cisco.com
> Cc: jmpolk@cisco.com
> Subject: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
> =20
>=20
> Hi all,
>=20
> =20
>=20
> Thanks very much! As already mentined by Anurag we will make the requested=
 changes by Bob and upload the new version before tomorrow morning!
>=20
> Sorry for not attending the tsvwg meeting, but I had to attend another mee=
ting  at that time! I thought that the rsvp-pcn draft will be presented duri=
ng the afternoon slot!
>=20
> =20
>=20
> Best regards,
>=20
> Georgios
> Van: Black, David [david.black@emc.com]
> Verzonden: maandag 29 juli 2013 13:04
> To: Anurag Bhargava; gorry@erg.abdn.ac.uk; bob.briscoe@bt.com; Karagiannis=
, G. (EWI); anuragb@cisco.com
> Cc: James Polk (jmpolk) (jmpolk@cisco.com); Black, David
> Onderwerp: RE: resolution with draft-ietf-tsvwg-rsvp-pcn
> Excellent =E2=80=93 that sounds like a plan.
> =20
> Thanks,
> --David
> =20
> From: Anurag Bhargava [ mailto:bhargava.anurag@gmail.com]=20
> Sent: Monday, July 29, 2013 6:57 AM
> To: gorry@erg.abdn.ac.uk; Black, David; bob.briscoe@bt.com; Georgios Karag=
iannis; anuragb@cisco.com
> Subject: Re: resolution with draft-ietf-tsvwg-rsvp-pcn
> =20
> Hi Gorry/David,
> =20
>   Bob, Georgios and I just had a discussion on the draft. Bob is quite com=
fortable now and have asked us to make a couple of changes.
> We will be doing the change today and will post a new version later today o=
r tomorrow morning.
> After we post the new version, please see if WGLC can be started (that wil=
l put pressure on the folks to review :)).
>=20
> --=20
> BR,
> -Anurag
> ________________________________________________________________
> Bob Briscoe,                                                  BT

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>Thanks all.</div><div><br></div><div>B=
r,</div><div>Anurag</div><div><br>Sent from my iPhone</div><div><br>On Jul 2=
9, 2013, at 8:18 PM, "Black, David" &lt;<a href=3D"mailto:david.black@emc.co=
m">david.black@emc.com</a>&gt; wrote:<br><br></div><blockquote type=3D"cite"=
><div><meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-a=
scii"><meta name=3D"Generator" content=3D"Microsoft Word 14 (filtered medium=
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--><div class=3D"WordSection1"><p class=3D"Ms=
oNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;=
;color:black">Bob,<o:p></o:p></span></p><p class=3D"MsoNormal"><span style=3D=
"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:black"><o:p>&nbs=
p;</o:p></span></p><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;fo=
nt-family:&quot;Courier New&quot;;color:black">Thank you for following up on=
 this.<o:p></o:p></span></p><p class=3D"MsoNormal"><span style=3D"font-size:=
10.0pt;font-family:&quot;Courier New&quot;;color:black"><o:p>&nbsp;</o:p></s=
pan></p><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&=
quot;Courier New&quot;;color:black">I=E2=80=99ve also taken a closer look at=
 the two text changes reported on the rsvp-pcn draft slides used today, and t=
hey both look fine to me (one was wrt a comment of mine).&nbsp; Gorry and I w=
ill consult on getting the WGLC started for this draft.<o:p></o:p></span></p=
><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p><div><div><p class=3D=
"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&qu=
ot;;color:black">Thanks,<br>--David</span><span style=3D"font-size:11.0pt;fo=
nt-family:&quot;Courier New&quot;;color:black"><o:p></o:p></span></p></div><=
/div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quo=
t;Courier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p><div style=3D"b=
order:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt"><div><div=
 style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0=
in"><p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&q=
uot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font=
-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Bob Bri=
scoe [<a href=3D"mailto:bob.briscoe@bt.com">mailto:bob.briscoe@bt.com</a>] <=
br><b>Sent:</b> Monday, July 29, 2013 1:37 PM<br><b>To:</b> Black, David; <a=
 href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a><br><b>Cc:</b>=
 <a href=3D"mailto:karagian@cs.utwente.nl">karagian@cs.utwente.nl</a>; <a hr=
ef=3D"mailto:bhargava.anurag@gmail.com">bhargava.anurag@gmail.com</a>; <a hr=
ef=3D"mailto:anuragb@cisco.com">anuragb@cisco.com</a>; <a href=3D"mailto:jmp=
olk@cisco.com">jmpolk@cisco.com</a>; Black, David; tsvwg IETF list<br><b>Sub=
ject:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<o:p></o:p></span></p=
></div></div><p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p><p class=3D"MsoNorm=
al">David, Gorry,<br><br>Just to confirm that <br>- we resolved the outstand=
ing issue over aggregation over lunch - I was concerned it would delay respo=
nse to flash crowds, but it was a misunderstanding on my part - so no change=
 needed to the draft on this count.<br>- the deltas posted in -06 this after=
noon do what I asked (not sending PCN-specific error messages to end-systems=
). My original review requested this, and Georgios had tried to comply, but t=
here were some places in the doc that still said the old stuff (incl in IANA=
 Considerations). Fixed now.<br><br>So I'm OK with WGLC - and I'll try to gi=
ve it one last read over for that.<br><br><br>Bob<br><br>At 12:35 29/07/2013=
, Black, David wrote:<br><br><o:p></o:p></p><p class=3D"MsoNormal">Not a pro=
blem =E2=80=93 we ran ahead of time on other agenda items and decided to dis=
pense w/the rsvp-pcn draft in the morning in order to free up some time this=
 afternoon =E2=80=93 Anurag was there, and everything=E2=80=99s in fine shap=
e wrt this draft now.&nbsp; I look forward to seeing the revised version.<br=
>&nbsp;<br>Thanks,<br>--David<br>&nbsp;<br><b>From:</b> <a href=3D"mailto:ka=
ragian@cs.utwente.nl">karagian@cs.utwente.nl</a> [<a href=3D"mailto:karagian=
@cs.utwente.nl"> mailto:karagian@cs.utwente.nl</a>] <br><b>Sent:</b> Monday,=
 July 29, 2013 7:31 AM<br><b>To:</b> Black, David; <a href=3D"mailto:bhargav=
a.anurag@gmail.com">bhargava.anurag@gmail.com</a>; <a href=3D"mailto:gorry@e=
rg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>; <a href=3D"mailto:bob.briscoe@bt.co=
m">bob.briscoe@bt.com</a>; <a href=3D"mailto:anuragb@cisco.com">anuragb@cisc=
o.com</a><br><b>Cc:</b> <a href=3D"mailto:jmpolk@cisco.com">jmpolk@cisco.com=
</a><br><b>Subject:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<br>&nb=
sp;<br><br>Hi all,<br><br>&nbsp;<br><br>Thanks very much! As already mentine=
d by Anurag we will make the requested changes by Bob and upload the new ver=
sion before tomorrow morning!<br><br>Sorry for not attending the tsvwg meeti=
ng, but I had to attend another meeting&nbsp; at that time! I thought that t=
he rsvp-pcn draft will be presented during the afternoon slot!<br><br>&nbsp;=
<br><br>Best regards,<br><br>Georgios<o:p></o:p></p><div class=3D"MsoNormal"=
 align=3D"center" style=3D"text-align:center"><hr size=3D"2" width=3D"100%" a=
lign=3D"center"></div><p class=3D"MsoNormal"><b>Van:</b> Black, David [<a hr=
ef=3D"mailto:david.black@emc.com">david.black@emc.com</a>]<br><b>Verzonden:<=
/b> maandag 29 juli 2013 13:04<br><b>To:</b> Anurag Bhargava; <a href=3D"mai=
lto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>; <a href=3D"mailto:bob.br=
iscoe@bt.com">bob.briscoe@bt.com</a>; Karagiannis, G. (EWI); <a href=3D"mail=
to:anuragb@cisco.com">anuragb@cisco.com</a><br><b>Cc:</b> James Polk (jmpolk=
) (<a href=3D"mailto:jmpolk@cisco.com">jmpolk@cisco.com</a>); Black, David<b=
r><b>Onderwerp:</b> RE: resolution with draft-ietf-tsvwg-rsvp-pcn<br>Excelle=
nt =E2=80=93 that sounds like a plan.<br>&nbsp;<br>Thanks,<br>--David<br>&nb=
sp;<br><b>From:</b> Anurag Bhargava [<a href=3D"mailto:bhargava.anurag@gmail=
.com"> mailto:bhargava.anurag@gmail.com</a>] <br><b>Sent:</b> Monday, July 2=
9, 2013 6:57 AM<br><b>To:</b> <a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@=
erg.abdn.ac.uk</a>; Black, David; <a href=3D"mailto:bob.briscoe@bt.com">bob.=
briscoe@bt.com</a>; Georgios Karagiannis; <a href=3D"mailto:anuragb@cisco.co=
m">anuragb@cisco.com</a><br><b>Subject:</b> Re: resolution with draft-ietf-t=
svwg-rsvp-pcn<br>&nbsp;<br>Hi Gorry/David,<br>&nbsp;<br>&nbsp; Bob, Georgios=
 and I just had a discussion on the draft. Bob is quite comfortable now and h=
ave asked us to make a couple of changes.<br>We will be doing the change tod=
ay and will post a new version later today or tomorrow morning.<br>After we p=
ost the new version, please see if WGLC can be started (that will put pressu=
re on the folks to review :)).<br><br>-- <br>BR,<br>-Anurag <o:p></o:p></p><=
p>________________________________________________________________<br>Bob Br=
iscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; BT<=
o:p></o:p></p></div></div></div></blockquote></body></html>=

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From Ruediger.Geib@telekom.de  Wed Jul 31 01:12:05 2013
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From: <Ruediger.Geib@telekom.de>
To: <eckert@cisco.com>
Date: Wed, 31 Jul 2013 10:11:05 +0200
Thread-Topic: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Subject: Re: [tsvwg] Using intelligent dropping to improve video quality under congestion (draft-lai-tsvwg-normalizer-00.txt)
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Toerless,

discussion seems to conclude with:

- fairness must be enforced if senders don't set DSCPs according to the glo=
bal
  fairness scheme.
- video applications have a hard time setting DSCPs anyway due to a missing=
 API.
- ECN is not meant to indicate drop priorities and offers an insufficient
  number of states.

It might be easier to use a single PHB/DSCP and add PCN like ECN. In an e2e=
 mode with receivers giving PCN congestion feedback to senders. My definiti=
on of PCN would be "rate control on a congested link" (rather than admissio=
n control and traffic termination as applied by the WG - I've been involved=
 in the WGs work).

This should result in:
- avoidance of sudden overload in that class (no burst drops)
- rate adaption of senders if congestion is pending
- a simpler design as compared to what you and Ken discussed

I'm going to drop another message soon.

Regards,

Ruediger

-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of T=
oerless Eckert (eckert)
Sent: Friday, July 12, 2013 3:00 AM
To: Dave Taht
Cc: rmcat@ietf.org; rtcweb@ietf.org; tsvwg@ietf.org
Subject: Re: [tsvwg] Using intelligent dropping to improve video quality un=
der congestion (draft-lai-tsvwg-normalizer-00.txt)

On Wed, Jul 10, 2013 at 12:02:26PM -0700, Dave Taht wrote:
> I've begun to look at webrtc of late in the context of fq_codel.
>
> One thing that might work would be ECN marking important frames and
> not ECN marking others.

True. But i think there are at least 3 layers of priority that would
really be great to have: really important packet (long term reference
frames), "normal" packets, and low-proirity-packets (eg: discardable P).
And ECN could only do two.

Most of the time when you would want intelligent dropping, your drop
rate should be fairly low, eg< 20% instantaneous within the queue. If
you can do discardable P-frames it's of course best to drop those first.
Only when you instantaneously have really big bursts would you want to
drop "normal" packets and refrain to protection really important packets.

This does work really well with different queue-length (or WRED profiles).
If you only had two priorities (via ECN) it's hard to say how you
would best assign these two priorities to at least those three priority
levels.

> Problem overall is that ecn can be gamed and is largely not
> enabled, and so on...

Well. The gaming could be inhibited exactly by this draft because it
would try to guarantee the same amount of drops on a per-flow or "class"
basis, so if you're trying to game the system, you would be put into
your own class and would still see the same amount of drops as other flows.

Well, the main issue is really to redesignate ECN for a different
semantic than what it has today. And indicating packet priority is just
not quite the same as congestion feedback.

> In the case of the fq_codel algorithm I can see a "delayed drop"
> mechanism happening,
> where upon deciding to drop, it could defer a drop up to a few packets
> based on the classification
> of the packet, say 3 packets at most. Since flows are already
> identified, this is kind of easy
> by adding another state variable...

Yes, this is all quite interesting. We also played around with droppping
during enqueuing vs. during dequeuing, so there are a bunch of things
that could optimize intelligent dropping if you can build a more
intelligent queue.

[...]

>
> So I can see trying to utilize the AF markings as you describe in your
> draft in codel's
> drop decision. While this is also subject to being gamed, the
> disincentives are slight.

Lets separate the marking from the mechanism to inhibit gaming. This
draft really is about the latter, and the marking described with AF is
really just an example. Yes, it would be lovely to agree on markings
too.

> And what codel depends on is a definition of "TCP friendly". That if a
> stream is not going to behave in a tcp friendly manner, then an entirely
> different  strategy for such flows seems necessary, and glomming on some
> additional state isn't going to be the right thing.

Who stood up at RMCAT some time back and claimed to have worked for
> 10 years on TCP and declared TCP not self friendly to itself, so RMCAT
shouldn't bother ? ;-))

Kidding aside: Intelligent dropping IMHO makes perfect sense when being
applied to RT traffic sharing a queue friendly with TCP. And intelligent
dropping can then still improve the performance.

I am pretty positive though that RMCAT traffic in general will behave
better and get lower delay and more video-appropriate loss if it can have
its own queue. Primarily because it will just take decates to root out
all the bad TCP traffic.

I don't think that the strategies if you do or if you don't share the queue
with TCP would be fundamentally be different.

> a) how well can it work in routers to shift packet drops to low-prio pack=
ets
>
> I guess where I fall apart here is how to shift the encoders to send thin=
ner
> streams at  any level of packet drop.

Video codec coders are known to be able to do crazy cool things.
There are a bunch of known mechanisms and probably more proprietary ones.

> I'd be interested in specific codecs, example video streams, and test
> scripts to try and fiddle with this, as well as viable metrics.

Sure. Best done in person. If we're going to be at the WG for this, we can
talk on the side how we've been doing this.

Cheers
    Toerless

> > Thanks!
> >     Toerless
> >
> > In-Reply-To: <A860EC86B79FA646BF3F89165A886264151D0AB9@xmb-aln-x11.cisc=
o.com>
> >
> > On Sat, Feb 09, 2013 at 03:00:48AM +0000, Cheng-Jia Lai (chelai) wrote:
> >> Hi all,
> >>
> >> We have submitted a new Internet draft to describe a Normalization Mar=
ker (NM) for DiffServ AF PHB groups, based on our work at Cisco. The goal o=
f NM deployment is to create a new incentive for video encoders to generate=
 more discardable packets when using AF PHB in DIffServ. We are looking for=
ward to your review comments.
> >>
> >> Best regards,
> >> CJ
> >>
> >> -----Original Message-----
> >> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> >> Sent: Friday, February 08, 2013 6:25 PM
> >> To: Cheng-Jia Lai (chelai)
> >> Cc: Wenyi Wang (wenywang); Stan Yang (stanyang); Toerless Eckert (ecke=
rt); Fred Yip (fyip)
> >> Subject: New Version Notification for draft-lai-tsvwg-normalizer-00.tx=
t
> >>
> >>
> >> A new version of I-D, draft-lai-tsvwg-normalizer-00.txt
> >> has been successfully submitted by Cheng-Jia Lai and posted to the
> >> IETF repository.
> >>
> >> Filename:      draft-lai-tsvwg-normalizer
> >> Revision:      00
> >> Title:                 Normalization Marker for AF PHB Group in DiffSe=
rv
> >> Creation date:         2013-02-08
> >> WG ID:                 Individual Submission
> >> Number of pages: 15
> >> URL:             http://www.ietf.org/internet-drafts/draft-lai-tsvwg-n=
ormalizer-00.txt
> >> Status:          http://datatracker.ietf.org/doc/draft-lai-tsvwg-norma=
lizer
> >> Htmlized:        http://tools.ietf.org/html/draft-lai-tsvwg-normalizer=
-00
> >>
> >>
> >> Abstract:
> >>    This memo describes a Normalization Marker (NM) at DiffServ (DS)
> >>    nodes to normalize the distribution of DS codepoint (DSCP) markings
> >>    for an Assured Forwarding (AF) Per-Hop Behavior (PHB) group.  NM is
> >>    useful for traffic conditioning with Active Queue Management (AQM)
> >>    when the AF PHB group is deployed for multimedia service classes th=
at
> >>    carry video packets with advanced codec technologies.  Using NM can
> >>    offer an incentive that is however missing in today's ecosystem of
> >>    practical deployment, i.e., when congestion occurs in the AF PHB
> >>    group, the network should allow a video codec to benefit from its
> >>    effort of generating finer layers of intra-flow precedence (IFP) wi=
th
> >>    discardable packets in a more balanced distribution.
> >>
> >>
> >>
> >>
> >> The IETF Secretariat
> >>
> >
> > --
> > ---
> > Toerless Eckert, eckert@cisco.com
> > Cisco NSSTG Systems & Technology Architecture
> > SDN: Let me play with the network, mommy!
> >
>
>
>
> --
> Dave T=E4ht
>
> Fixing bufferbloat with cerowrt: http://www.teklibre.com/cerowrt/subscrib=
e.html

--
---
Toerless Eckert, eckert@cisco.com
Cisco NSSTG Systems & Technology Architecture
SDN: Let me play with the network, mommy!


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From: <Ruediger.Geib@telekom.de>
To: <eckert@cisco.com>
Date: Wed, 31 Jul 2013 10:26:48 +0200
Thread-Topic: [tsvwg] Using intelligent dropping to improve video.....adding admission control?
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Subject: Re: [tsvwg] Using intelligent dropping to improve video.....adding admission control?
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Toerless,

prior to setting up a new call, some kind of available bandwidth measuremen=
t or indication might be added. This isn't part of your draft. But an addit=
ional video stream may cause congestion on a bottleneck.

Measurement based admission control has been controversial so far. If used,=
 the main point is to use a measurement class sharing the same queue as the=
 traffic to be admitted, but having a higher drop probability.

- PCN is an option, but requires a new marking behaviour
 (threshold marker). End to end PCN as mentioned in another email.
- Using AF41 for admitted traffic and AF42 to perform measurements
  is an option to me, both should support ECN marking (AF42 seeing
  the higher marking and drop probability).

As marked packets indicating an issue can be interpreted by routers along a=
 path, while packets dropped earlier elsewhere can't be, I prefer marking t=
o dropping. I also think, it is easier to detect packet marks than delay va=
riation measurements in routers, if it is about detecting congestion.

Take the above as an example. I used a separate thread as this isn't part o=
f your draft an I accept if this shouldn't be discussed in the context of t=
his draft. It fit's well to the RTCWeb QoS discussion though, I think.

Regards,

Ruediger
