Subject: [AVT] IETF 61 AVT minutes

From: Colin Perkins <[email-masked]>
Date: Fri, 10 Dec 2004 08:41:15 +0000

Audio/Video Transport Working Group Minutes

Reported by Colin Perkins from notes by Marshall Eubanks

Introduction and Status Update

     The Audio/Video Transport working group met twice at the 61st IETF
     meeting (Washington DC, 10th and 11th November 2004). Topics under
     discussion included the RTCP XR MIB, the new RTP Profile for TCP
     Friendly Rate Control, an RTP header extension for anti-shadow
     redundancy, and RTP payload formats for telephony tones and events,
     JPEG 2000, H.224, VMR-WB audio, and several variants of AMR audio.
     The group also discussed the issue of how the RTP timestamp should 
be
     interpreted for audio codecs where the input and output sampling 
rates
     are decoupled. The meeting was chaired by Colin Perkins and Magnus
     Westerlund.

     The meeting started with a working group status update presented by 
the
     chairs. The group has had no new RFCs published since the previous 
IETF
     meeting, but 4 drafts are with the RFC editor awaiting publication. 
In
     addition, we have requested publication for 11 more drafts which are
     currently awaiting review by our area director or the full IESG. The
     RTCP feedback draft is expected to go to the IESG for review 
shortly.
     The chairs briefly outlined the status of the drafts that were not 
to
     be discussed in detail, and solicited feedback via the mailing list.

     Colin Perkins noted that the grqup is slightly behind on some of its
     milestones, and that some of them will need to be updated. The group
     also needs to update its charter to include new work. The most 
pressing
     milestone to which no effort has been allocated is the updated RTP 
MIB;
     Alan Clark volunteered to work on this, but solicited help.

RTP Payload Format for MIDI
draft-ietf-avt-rtp-midi-format-06.txt
draft-ietf-avt-rtp-midi-guidelines-06.txt

     John Lazzaro couldn't be at the meeting to discuss the MIDI drafts, 
but
     sent a slide outlining the status of the work.  The draft has 
received
     an expert review from Dominique Fober from the MIDI community. This
     found no significant issues, and we expect the draft will be 
completed
     soon. Comments from the group were solicited.

RTCP XR MIB
draft-ietf-avt-rtcp-xr-mib-00.txt

     Alan Clark and Amy Pendleton discussed the RTCP XR MIB. This has 
been
     updated since the previous meeting, and is now a working group 
draft.
     Recent changes include a section on the relation between the RTCP XR
     MIB and RAQMON that will incorporate wording suggested by the RAQMON
     authors, plus a number of technical changes to the MIB to address
     comments by Magnus Westerlund and Dan Romascanu.

     After Alan introduced the work and gave the status update, Amy gave 
a
     detailed response to the comments received. Discussion centered 
around
     three areas: use of the SSRC as a session identifier, the complexity
     of the MIB, and the need for new parameters to be signalled.

     The issue with using the SSRC as a session identifier is that it is
     unique only within a session, not necessarily across sessions. There
     was much discussion, but no real conclusions were drawn: it is clear
     that using the SSRC isn't sufficient to correlate across sessions,
     but it's not clear what would be an appropriate session identifier.

     Regarding the complexity of the MIB, Dan Romascanu noted that the 
MIB
     is large and may not all be implemented by everyone. It might be a 
good
     idea to split the MIB, to allow implementations to claim conformance
     to subsets of the MIB with different tables? Jean Francois Mule also
     expressed interest in such a split.

     There was a lot of discussion regarding the calculation of the R 
factor
     and how the MOS-LQ is to be calculated. Jean Francois Mule was keen 
to
     see information added to the MIB to specify how particular 
parameters,
     that can be calculated in various ways, were defined. Alan Clark 
noted
     that RFC3611 does recommend a minimum algorithm (G.107), however, 
this
     is an evolving area: the state-of-the-art a year ago has been 
extended
     by now. It would be very useful to be able to specify enhanced 
methods
     in the future. It seems clear that something may need to be added to
     the MIB to address this: Colin Perkins suggested Jean Francois 
write-up
     a proposal for changes to send to the mailing list, and that 
discussion
     continue there.

Open Issues in RFC 2833bis
draft-ietf-avt-rfc2833bis-05.txt

     Tom Taylor discussed the open issues in RFC 2833bis, starting with a
     review of the changes in the -04 and -05 drafts. There are a lot of
     changes (listed in the -05 draft) a number of which cause backwards
     compatibility issues: compliance no longer requires support of DTMF
     events, a number of trunk events have been discarded and alternative
     means of signalling them defined, and event codes 138 and 139 have 
been
     reassigned. Feedback was solicited on these changes, and Jean 
Francois
     Mule expressed concern over backwards compatibility of the 
signalling
     for DTMF, suggesting that the default be to assume DTMF is supported
     unless signalled otherwise. Jean Francois and Flemming Andreasen 
noted
     that DTMF is widely supported, but other events are not. Colin 
Perkins
     suggested splitting the draft, progressing the DTMF parts quickly. 
The
     idea of a split was well received.

     The E bit is usually set to indicate the end of an event, but there 
is
     an exception for events defined as states, when the E bit doesn't 
have
     to be set when moving between mutually exclusive states. This rarely
     happens, but complicates the code and has little real benefit. It 
was
     asked if this is worth the complication? Colin Perkins suggested to
     remove this feature, unless there were objections. No one spoke up 
for
     the continued presence of this feature.

     There are a number of tone events listed that are not well defined.
     Tom suggested to remove these? Stephan Wenger and others supported
     their removal.

     Tom noted that the draft is very long, and it's not clear that all 
the
     events are needed. The decision to split out DTMF into a separate 
draft
     will help with this problem, but we may also consider other events 
for
     removal.

     Tom asked if we should establish an IANA registry for the code 
points,
     or if it should be done when the draft was published as an RFC. 
Colin
     Perkins suggested this should go into the IANA considerations 
section
     now, rather than waiting for publication.

RTP Payload Format for JPEG 2000
draft-ietf-avt-jpeg2000-06.txt

     Andrew Leung discussed the RTP payload format for JPEG2000. Since 
the
     last meeting, an IPR statement has been published listing the patent
     applications that cover this draft, the non-intelligent mode has 
been
     removed, the MIME types and SDP parameters have been improved, and 
the
     payload header has been slightly reordered.

     After off-line discussion with the chairs at the meeting, it has 
been
     agreed to split this draft into two parts. The patent applications 
do
     not apply to the basic protocol, but to various enhanced features. 
It
     looks to be possible to split the draft into an unencumbered base 
and
     an encumbered extension draft. This will be done by the next 
meeting.
     Stephan Wenger applauded this split, and encouraged other companies
     with encumbered payload formats to consider similar moves.

     The non-intelligent packetization has been removed. There is only 
one
     branch in the code, making better use of the payload header and 
using
     the Main Header Flag Field.

     Regarding the MIME types, Colin Perkins noted that the uncompressed
     video draft has similar parameters to identify the colour space. It
     would be good if the two drafts could be made consistant.

     Revised drafts addressing these issues should be available by the 
end
     of the year.

Far End Camera Control/H.224
draft-even-avt-h224-00.txt

     Roni Even discussed the MIME type registration for the RTP payload
     format for H.224. ITU has defined, amongst other things, an 
approach to
     far end camera control in H.281 and H.224. H.323 annex Q defines 
how to
     transport this within RTP, however it does not register an 
appropriate
     MIME subtype. This draft registers the "application/h224" MIME 
subtype,
     referencing H.323 annex Q, allowing SIP applications to interwork 
with
     existing H.323 systems that use the format.

     Roni asked for feedback, Steve Casner noted that it looked fine in 
his
     view. Alan Johnston (the XCON co-chair) agreed, and noted that far 
end
     camera control is out of scope for XCON. Colin Perkins concluded 
with
     an acceptance of this as an AVT working group draft, unless there 
were
     objections via the mailing list.

RTP Payload for Anti-Shadow Redundancy
draft-xie-avt-rtp-anti-shadow-00.txt

     Qiaobing Xie discussed a new draft on anti-shadow redundancy for 
RTP.
     There may be IPR issues with Motorola: the IETF has been notified 
and
     an IPR statement will be forthcoming.

     This draft attempts to address the problem of shadowing in mobile 
radio
     communications, where blockage on signals can range from fractions 
of a
     second to a couple of minutes, and where traditional RTP-style FEC 
is
     not effective. The draft defines an RTP header extension, where the
     sender generates a base media stream and a forward-shifted 
anti-shadow
     stream sent at the same time. The amount of forward shift determines
     the maximum length of blockage that can be recovered without service
     interruption, and the amount of data that must be received before 
the
     first interruption. The anti-shadow stream may be lower quality than
     the original.

     There was a lot of discussion, with Lorenzo Vicisano, Joerg Ott, 
Steve
     Casner, Stephan Wenger, Colin Perkins, Magnus Westerlund, Anders 
Klemets
     and Marshall Eubanks noting that this is a simple but inefficient 
form
     of FEC, which has a long buffering delay and problems with startup, 
is
     not likely to work well with single packet losses, and is difficult 
to
     deploy when the lower-quality encoding has different timing 
properties
     to the primary. It was noted that the RMT working group, and others,
     have effective FEC schemes that can solve this problem. It was also
     noted that it may be more appropriate to solve this at the link 
layer,
     or via layer/parallel transmission.

     Colin Perkins concluded by stating that there is clear pushback 
against
     this draft as it stands. If there is a desire to proceed with the 
work,
     the authors need to define the problem and explain why it's 
something
     we need to solve within AVT, before working on solutions.

RTP Jitter Calculation

     At the end of the first session, Steve Casner noted that there has 
been
     recent discussion on the list about jitter calculation. This is 
defined
     very specifically and a particular algorithm is required so 
calculations
     are comparable both over time and between streams, if not 
necessarily
     between, say, a video and an audio codec. People should pay more 
attention
     to the receiver reports section of RFC 3550, which describes how to 
make
     use of this.

RTP Profile for TCP friendly rate control
draft-ietf-avt-tfrc-profile-03.txt

     The second session started with a presentation of the RTP Profile 
for
     TFRC (RTP/AVPCC) by Ladan Gharai. After summarising the operation of
     the profile, Ladan discussed the changes since the -02 draft. The 
RTP
     header previously included a 16 bit quad-RTT counter as a profile
     specific extension. This has been changed to a 32 bit send timestamp
     with a 32 bit RTT sent only when the RTT estimate changes 
significantly.
     The presence of the RTT is indicated by an "R" bit, taken from the 7
     bit payload type field (as extra marker bit, as a profile specific
     change to the RTP header). The RTP/AVPCC profile uses dynamic 
payload
     types only, and does not share the static payload type space with
     RTP/AVP. Joerg Ott noted that the presence of the R and M bits could
     cause the RTP payload type to alias the RTCP payload type values -
     could there be confusion? Colin Perkins noted that standard RTP has
     the same issue, and reserves some payload types. This should also be
     done for RTP/AVPCC.

     RTCP RR packets are extended with 4 32-bit fields, describing the
     timestamp of the last data packet received (t_i), the delay between
     receipt of the last data packet and the generation of this feedback
     packet (t_delay), the rate at which the receiver estimates that data
     was received since the last feedback report was sent (x_recv), and 
the
     loss event rate (p). Colin Perkins asked if the field p and t_delay
     could be reduced in size? Possibly, but it may be possible to use 
the
     data in sender reports to deduce the RTT in interactive sessions,
     saving more bandwidth.

     Another open issue is the RTCP minimum timing interval. TFRC says to
     send feedback once per RTT or once per packet, whichever is longer.
     However for flows with small round-trip time and data rate, this can
     require sending feedback packets more often than allowed by the RTCP
     timing rules, and consuming large amount of bandwidth. It was asked
     if the RTCP timing rules can be relaxed in these cases? Steve Casner
     agreed that they probably could be, since this is specifically for
     unicast and so will not have flash joins or leaves, however this 
does
     not change the excessive bandwidth usage for the feedback in these
     cases. An alternative might be to send less than one packet of 
feedback
     per round-trip time; this is a regime where TFRC hasn't been tested
     and, as noted by Tom Phelan, it is not clear that TFRC is 
TCP-friendly.
     More research will be needed to determine if TFRC can operate well 
at
     that rate.

     The draft adopts the SRTP specification (RFC 3711) for security.  
There
     can be a NULL security operation, if there is no desire for 
privacy. It
     was pointed out that SRTP requires RTCP packets to be authenticated,
     which may not be desirable. Anders Klemets expressed concern about 
this
     additional requirement, wanting secure RTP/AVPCC to be a separate
     profile to standard RTP/AVPCC. The sense of the room was to agree 
with
     Anders, rather than forcing authentication on all implementations.

     Finally, Ladan asked if the group thought it appropriate to include
     more introductory text on TFRC into the draft. Colin replied that 
this
     never hurts, but there are perhaps more pressing design issues to
     resolve with the draft.

RTP Timestamps for Variable Frequency Codecs.
draft-ietf-avt-variable-rate-audio-00.txt

     An issue which has generated a lot of traffic on the mailing list is
     how to handle the RTP timestamp for audio codecs that use different
     input and output sampling rates. Colin Perkins summarised the issues
     and presented options on how to proceed.

     The RTP timestamp reflects the sampling instance of the first octet 
in
     the RTP data packet, and MUST be derived from a clock that 
increments
     monotonically and linearly in time [RFC 3550, section 5.1]. This 
means
     that we cannot vary the RTP timestamp rate within a session.

     There are different conventions for use of the timestamp between 
audio
     and video formats. For audio, the RTP clock rate used for generating
     the RTP timestamp is independent of the number of channels and the
     encoding; it usually equals the number of sampling periods per 
second
     [RFC 3551, section 4.1]. There are two exceptions to this: G.722 
uses
     an 8kHz RTP timestamp clock for a 16kHz sampling rate codec, for
     compatibility with a mistake in RFC 1890; and MPEG audio uses a 
90kHz
     clock for compatibility with other MPEG systems [RFC 3551 sections
     4.5.2 and 4.5.13, RFC 3119]. The implication is that using RTP clock
     rate other than the sampling rate is allowed, but one must consider
     how it will affect the overall system design.

     For audio, the convention is that the sampling frequency SHOULD be 
drawn
     from the set 8000, 11025, 16000, 22050, 24000, 32000, 4400 and 
48000 Hz.
     However, most audio encodings are defined for a more restricted set 
of
     sampling frequencies [RFC 3551, section 4.1].

     Two new audio formats violate these conventions. VMR-WB accepts 
8kHz or
     16kHz input and produces 16kHz output by default, irrespective of 
the
     input sampling rate. It is desired to use a fixed 16kHz clock for 
the
     VMR-WB payload format. AMR-WB+ can accept a range of input sampling
     rates, re-samples within the codec to one of 12 internal sampling
     frequencies, and can produce output at one of 8, 16, 24, 32 or 
48kHz.
     The draft RTP payload format for AMR-WB+ uses a fixed 72kHz clock. 
The
     key points for both new codecs are that the input and output 
sampling
     rates are decoupled and that the decoder is agnostic of the input
     sampling rate (and that the rate can vary within a session).

     These codecs raise two issues: 1) the number of sampling periods per
     second varies in different parts of the system: the usual 
definition of
     the RTP clock rate for audio it not sufficient; and 2) is it 
necessary
     to separately signal the input and output rates, and, if so, how? 
Colin
     suggested that consistency across codecs is desirable, to simplify 
the
     protocol and implementations that support multiple codecs.

     There are four options going forward:

      1) Only support codecs with one sample rate. This makes for easy
         backwards compatibility, but won't support certain codecs.

      2) Mandate a common clock rate for all RTP codecs as described in
         draft-ietf-avt-variable-rate-audio-00.txt. Stephan Wenger gave
	an outline of this (the "uRTR" approach).

      3) Allow each codec to specify its own definition for the RTP
	timestamp and signaling

      4) Do something else?

     Finally, it was asked if we need a set of guidelines for developers
     of new audio codecs which might need transport in RTP?

     After presenting option 2, Stephan noted that his experience is that
     once Pandora's box is opened, the issue does not go away.  We'll see
     this with more and more codecs in the future - probably most future
     audio codecs. But this is good, as the ability to change sample 
rates
     means that bandwidth requirements will decline and efficiency will
     increase.

     Stephan noted several issues with option 2: Should this apply to all
     future audio codecs, or just the variable clock frequency ones? What
     about codecs that don't fit the subset sampling rates (e.g. Magnus
     Westerlund noted that DVD audio uses an 88.2kHz sampling rate). How
     do we cope with non-sample-exact mixing and synchronisation?

     Qiabing Xie would support the uRTR, but wondered what if we pick not
     one but two uRTRs? Is that so bad? Colin Perkins wondered what would
     we would gain from using two uRTRs? Stephan Casner replied that we'd
     gain exactness in calculations during synchronisation.

     Magnus Westerlund noted that the problem he sees with uRTR is with
     codecs that don't fit the rate.  You will wind up with frames with
     fractional lengths.  That is a real complication. This will cause
     error. There was a lot of discussion of the details of this, between
     Magnus and Stephan.

     Steve Casner asked what is the strong gain that comes from mandating
     one number? We know that there are some downsides, what is the gain?
     Colin replied that the strong gain comes from if the uRTR time rate 
is
     an integer multiple of all possible time rates. Doubt was expressed
     that this is possible for a realistic common rate.

     The group took a "hum" on the preferred outcome. Option 3 got the
     majority. Options 1 and 4 got nothing, option 2 very little. Colin
     Perkins wrapped up the session by stating that it was clear option
     3 was the consensus, and we just have to accept the complexity it
     causes for implementations.

RTP Payload Format and File Format for VMR-WB
draft-ietf-avt-vmr-wb-file-format-00.txt
draft-ietf-avt-rtp-vmr-wb-05.txt

     Sassan Ahmadi presented the open issues with the VMR-WB payload 
format,
     now that the previous discussion has given the go-ahead to its use 
of a
     fixed 16kHz timestamp.

     Since the last meeting, the draft has been split into two: one for 
the
     payload format, one for the file format. Currently there is one MIME
     subtype with four magic numbers for the file format, but it has been
     suggested that four MIME subtypes should be defined with appropriate
     headers and parameters (to match typical MIME usage). Colin Perkins
     also noted that the AMR file format was origianlly defined in a very
     similar manner to this VMR-WB file format, but that 3GPP is moving 
away
     from this towards using an ISO-based file format. It may be 
appropriate
     to consider using an ISO-based file format for VMR-WB+, defined 
within
     3GPP2, rather than repeating a design choice that didn't work well 
for
     3GPP. There are concerns that this is not an appropriate file 
format,
     based on experience with AMR, and the group would like for 3GPP2 to
     look at this again and think about what their requirements are. Once
     they do that, we can move forward.

RTP Payload Format for AMR and AMR-WB
draft-ietf-avt-rtp-amr-bis-00.txt

     Magnus Westerlund presented an update to RFC 3267, intended to 
address
     its shortcomings and to move the AMR and AMR-WB payload formats to
     draft standard. In addition to a number of clarifications, the big
     change in this draft is a new offer/answer section.

     After private discussion during the week, the authors have decided 
that
     this new offer/answer section needs revision: there are issues with 
the
     gateway-related paramters (mode-set, mode-change-period, and 
mode-change
     neighbour) which may result in a new parameter 
(mode-change-capabilities)
     being defined. In addition, the mode-change-neighbour parameter may 
be
     changed to be only a hint, rather than a strict requirement on 
possible
     mode changes (as it will usually work changing between arbitrary 
modes).
     A new draft will be produced to include these changes before the 
end of
     the year.

     Magnus noted that Ericsson and Nokia have performed tests of about 
50%
     of the various parameter combinations. However, help with conducting
     the reminder of the interoperability tests was solicited.

RTP Payload Format for AMR-WB+
draft-ietf-avt-rtp-amrwbplus-02.txt

     Magnus Westerlund also presented the AMR-WB+ payload format. He 
noted
     that Nokia believes there is an unpublished Nokia patent application
     that may be relevant to this draft (an IPR statement is available on
     the IETF website). AMR-WB+ is one of the recommended codecs for 3GPP
     PSS and MMS, and the technical specifications of the codec are ready
     and published now. It is aslo likely be become a mandatory codec in
     3GPP MBMS.

     The payload format as written has a problem with overhead, with its
     flexibility coming with an unacceptable price. It turns out that 
this
     flexibility isn't always needed, since the internal sampling 
frequency
     and frame type are rarely switched. By making assumptions that these
     will be constant for certain frame grouping, it's possible to reduce
     the overhead significantly. An outline new proposal was presented
     (see slides) that makes this change. A revised version of the draft,
     including this proposal, will be published shortly. Steve Casner
     expressed approval for the new proposal, noting that this is a 
change
     in the direction that he generally favours.

				  -- + --

_______________________________________________
Audio/Video Transport Working Group
[email-masked]
https://www1.ietf.org/mailman/listinfo/avt

---

From: Qiaobing Xie <[email-masked]>
Date: Fri, 10 Dec 2004 16:14:05 -0600

Some comments on the comments made to draft-xie-avt-rtp-anti-shadow-00.txt at IETF 61

....
> draft-xie-avt-rtp-anti-shadow-00.txt
....
>     There was a lot of discussion, with Lorenzo Vicisano, Joerg Ott, Steve
>     Casner, Stephan Wenger, Colin Perkins, Magnus Westerlund, Anders Klemets
>     and Marshall Eubanks noting that this is a simple but inefficient form
>     of FEC, which has a long buffering delay 

This is not true (and may be due to a misread of the draft). One of the motivations of this 
proposal is to remove the buffer delay at the start of a session - a problem other schemes 
(including FEC and normal jitter buffer) suffer from. As clearly stated in the draft, it 
achieves this objective, i.e., no buffer delay at the start under any situation.

>     and problems with startup, is

It is made clear in the draft that there will be no anti-shadow (AS) protection at the 
beginning of a session for the duration equal to the amount of the forward-shift. But in a 
sense this is a tradeoff between "being forced to wait for x sec before the play can start" 
(e.g., FEC or jitter buffer) and "playing start instantly but no AS for the first x sec" 
(this approach). For one, I would prefer the latter without a doubt.

>     not likely to work well with single packet losses, 

How come? Please give more specifics of this comment since I think it will work well under 
this situation.

 >     and is difficult to
>     deploy when the lower-quality encoding has different timing properties
>     to the primary. 

Sure, if one chooses to use different timing properties between the two streams, it simply 
creates a harder problem for him/herself. But why one would want to do that? Normally, this 
is not an issue since both streams are originated from the same source and it is not hard to 
keep the timing properties the same.

In addition, this AS scheme does not require tight timing sync between the two streams at 
all. This is important IMO because it will make things easier to deploy.

>     It was noted that the RMT working group, and others,
>     have effective FEC schemes that can solve this problem. 

That would be good and I'd like to see proposals based on those methods. The FEC solutions 
that I am aware of so far do not provide *effective* anti-shadow protection.

>     It was also
>     noted that it may be more appropriate to solve this at the link layer,

I am not sure of solving this at the link layer, but probably parallel trans is an alternate 
way to do it.

> 
>     Colin Perkins concluded by stating that there is clear pushback against
>     this draft as it stands. If there is a desire to proceed with the work,
>     the authors need to define the problem and explain why it's something

This (whether to solve it within AVT) is a valid question. The proposed AS scheme itself can 
be deployed without avt doing any additional spec or payload work (e.g., one could use two 
separate RTP streams to accomplish the AS algorithm since there is no tight coupling needed 
for the timing). However, I think a payload format from avt could improve the delivery 
efficiency (less traffic overhead).

regards,
-Qiaobing

---

From: Marshall Eubanks <[email-masked]>
Date: Fri, 10 Dec 2004 22:56:57 -0500

Hello;

> achieves this objective, i.e., no buffer delay at the start under any situation.

This is to me very similar to the SLAP protection of multicast audio
that OnTheI has
broadcast for years. I do regard this as logically a form of FEC, but
it is different enough
to warrent separate treatment.

The reason why we never pursued this as a RFC is that we could not think of an
elegant way to make different streams at different bit rates match in
general. (Different bit rate
streams in general will have different frame durations for most
codecs, so they will not
line up evenly in time, which is nasty at playback time.) I do not see
that this has been solved in this
proposal; it might be sufficient just to point out the problem and
require its avoidance.

This in practice works very well if you have control over what the
codec does, which is one reason why XM uses it.

> (this approach). For one, I would prefer the latter without a doubt.

Which is what we did too - making the first few seconds unprotected is
a reasonable tradeoff
to making the startup buffering very short.

> How come? Please give more specifics of this comment since I think it will work well under

I don't see this either. Generally single packet losses are
undetectable even if the backup
stream has a much lower quality.

> keep the timing properties the same.

Not so trival for many codecs. For MP3's, for example, you have to
keep the same sampling rate
to keep the same audio frame duration (it's a fixed number of
samples). The default for many MP3
encoding packages is to lower the sample rate as the quality / bit
rate declines, which is not what
you want. This can be fixed, but it requires control of the codec.

Ogg Vorbis is even messier and I don't think this would work without
control of the codec code.

Video can be even more messy as many codecs drop video frames, etc.,
depending on quality.

> >     have effective FEC schemes that can solve this problem.

They do not in my opinion for streaming. What RFCs are you referring to ?

> efficiency (less traffic overhead).

Regards
Marshall

---

From: Colin Perkins <[email-masked]>
Date: Sat, 11 Dec 2004 12:11:38 +0000

[cc line trimmed]

On 11 Dec 2004, at 03:56, Marshall Eubanks wrote:
>> Some comments on the comments made to 
>> draft-xie-avt-rtp-anti-shadow-00.txt at IETF 61
>> ....
>> It is made clear in the draft that there will be no anti-shadow (AS) 
>> protection at the beginning of a session for the duration equal to 
>> the amount of the forward-shift. But in a sense this is a tradeoff 
>> between "being forced to wait for x sec before the play can start" 
> a reasonable tradeoff to making the startup buffering very short.

I agree that it's likely a reasonable trade-off; however it was 
highlighted as a potential issue during the discussion.

>> How come? Please give more specifics of this comment since I think it 
>> will work well under this situation.
>
> undetectable even if the backup stream has a much lower quality.

The notes I took during the meeting say:

- [Lorenzo Vicisano]: this is a form of simple, inefficient, fec. good
   at covering big gaps, but not at single packet losses (if there's a
   correlation between time shift and gaps). might be useful when sending
   lower quality redundant stream

I think the point was that other FEC schemes may be more robust than a 
single delayed transmission, since there are several ways to recover 
each loss.

...
> They do not in my opinion for streaming. What RFCs are you referring 

Lorenzo and Stephan Wenger both pointed to the FEC building block work 
going on in RMT and in 3GPP; Joerg Ott mentioned FLUTE. I don't know if 
they can solve your problem, but I think the conclusion of the 
discussion was that the anti-shadow draft needs to explain why the 
existing FEC work in IETF doesn't work.

>> That would be good and I'd like to see proposals based on those 
>> methods. The FEC solutions that I am aware of so far do not provide 
>> *effective* anti-shadow protection.
>>
>>>     It was also noted that it may be more appropriate to solve this 
>>> at the link layer, or via layer/parallel transmission.
>>
>> I am not sure of solving this at the link layer, but probably 
>> parallel trans is an alternate way to do it.

Discussion of the effectiveness of the alternatives would be valuable.

...
>> This (whether to solve it within AVT) is a valid question. The 
>> proposed AS scheme itself can be deployed without avt doing any 
>> additional spec or payload work (e.g., one could use two separate RTP 
>> streams to accomplish the AS algorithm since there is no tight 
>> coupling needed for the timing). However, I think a payload format 
>> from avt could improve the delivery

A standard technique is clearly desirable. My overarching impression 
from the discussion, though, is that the group was not convinced that 
the approach you have chosen is the best solution to the problem: it 
might be, but you need to justify your design choices and explain why 
the alternatives aren't suitable.

Cheers,
Colin

---

From: Marshall Eubanks <[email-masked]>
Date: Sat, 11 Dec 2004 11:24:40 -0500

Hello;

>    lower quality redundant stream

Lorenzo is correct - this is not particularly efficient at correcting
for single packet losses. However,
the psycho acoustics and psycho visual aspects are interesting, as
replacing a single dropped frame
(either a small set of audio samples or even an entire video frame in
a reasonably high motion video) with a lower quality backup is
typically very hard for a human to detect, as obvious as it may be on
a spectrum analyzer. Since actual packet losses tend to be bursty,
this technique works very well in conjunction with interleaving (to
convert contiguous loss bursts in real time to isolated single packet
bursts at  playback time).

> single delayed transmission, since there are several ways to recover

Like always in engineering, there are different tradeoffs depending on
what path you take and
what you are actually trying to do. Clearly, this is not as simple and
may not be as efficient as XOR FEC for repairing Poisson packet
losses. On the  other hand, I don't see much that's Poisson about
packet losses on the Internet.

We used 3 separate transmissions with interleaving, and therefore had
200% redundancy, could support layering and could repair quite long
packet dropouts (up to 5+ seconds) without loss of signal, all for a
bit rate increase of about 60%. Its hard for me to see how FEC could
meet those goals without basically being driven to a very similar
solution.

Regards
Marshall