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Subject: [codec] Liaison from 3GPP
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IETF received the following liaison from 3GPP.

https://datatracker.ietf.org/liaison/562/



From swmike@swm.pp.se  Thu Sep  3 08:10:27 2009
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From: Mikael Abrahamsson <swmike@swm.pp.se>
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Subject: Re: [codec] Liaison from 3GPP
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On Thu, 3 Sep 2009, Cullen Jennings wrote:

> https://datatracker.ietf.org/liaison/562/

The point about how important "royalty free" was considered by a lot of 
the people at the mic at the WG BOF in Stockholm seems to have been missed 
by 3GPP (at least that is my conclusion after reading point 3 of the 
liaison).

How do we communicate that this is a fundamental requirement for an "Sound 
over Internet" standard?

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

From jmh@joelhalpern.com  Thu Sep  3 13:48:53 2009
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Subject: Re: [codec] Liaison from 3GPP
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One obvious path would be to have a process by which the IETF could 
determine what it considers to be the requirements, and get community 
confirmation of that agreement.  We can then make that avaialble to the 
ITU, 3GPP, et. al.   It would seem likely that the community would adopt 
wording about the Codec being royalty and license free.  (Whether it 
would adopt as strong wording as you want, or weaker, or even stronger, 
would be determined by the community.)

An IETF working group to define these requirements would seem a 
practical way to determine the community requirements.

Yours,
Joel M. Halpern

Mikael Abrahamsson wrote:
> On Thu, 3 Sep 2009, Cullen Jennings wrote:
> 
>> https://datatracker.ietf.org/liaison/562/
> 
> The point about how important "royalty free" was considered by a lot of 
> the people at the mic at the WG BOF in Stockholm seems to have been 
> missed by 3GPP (at least that is my conclusion after reading point 3 of 
> the liaison).
> 
> How do we communicate that this is a fundamental requirement for an 
> "Sound over Internet" standard?
> 

From stpeter@stpeter.im  Fri Sep  4 07:27:33 2009
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Subject: [codec] Minutes from IETF 75 BoF
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

========================================================================

Minutes for Internet Wideband Audio Codec BOF, IETF 75
Kongresshall C, City Conference Centre, Stockholm, Sweden
Thursday, July 30, 2009, 13:00-15:00 local time, 11:00-13:00 UTC

CHAIRS: Jason Fischl, Jean-Marc Valin

NOTE TAKERS: Olle E. Johansson and Bruce Lowekamp
JABBER SCRIBES: Alan Hawrylyshen and Peter Saint-Andre

MINUTES EDITOR: Peter Saint-Andre

AGENDA:

<http://www.ietf.org/proceedings/75/agenda/codec.txt>

1. Administrivia (5 minutes)
2. Introduction and Scoping of the BoF (10 minutes)
3. Goals of the Working Group (15 minutes)
4. Engineering Work (25 minutes)
5. Codec Expertise in the IETF (15 minutes)
6. Gauging Likelihood of Success (15 minutes)
7. Call for Consensus on Whether to Form a WG (15 minutes)
8. Charter Discussion (15 minutes)
9. Conclusions and Next Steps (5 minutes)

AUDIO: <http://www.ietf.org/audio/ietf75/ietf75-thur-kongressc-pm.mp3>

JABBER LOG: <http://jabber.ietf.org/logs/codec/2009-07-30.txt>

EDITOR'S NOTE: This was a very fast-pased BoF with a great many
speakers at the mic and much back-and-forth conversation. Although
the note-takers have attempted to capture all of the discussion, we
cannot guarantee that the minutes are truly complete. Please refer
to the audio stream and the Jabber logs for details. Approximate
minute markings have been provided for reference, with time in UTC
to synchronize with the Jabber logs.

========================================================================

1. ADMINISTRIVIA (11:06)

1a. Note well slide presented.

1b. Remote logistics, Jabber room, meeting materials, audio, etc.

1c. Volunteers noted for note takers and Jabber scribes.

1d. Agenda bashing.

Chairs note one modification: order of items 7 and 8 switched.

Roni Even @ mic: The agenda topic "Engineering Work" does not explain
what exactly will be discussed. Codecs or requirements?

Jason Fischl (Chair): The topic will be the *kind* of work that we
would be doing. We will look at some attributes that we would use as a
way of evaluating codecs, then show how the 5 codecs that have been
submitted would address those attributes. We will have short
presentations (5 minutes) about individual codecs that have been
submitted to indicate the kinds of work that might be in scope, but
these codecs are merely representative.

========================================================================

2. INTRODUCTION AND SCOPING OF THE BOF (11:11)

2a. Cullen Jennings (AD) presenting

* SLIDE 1: Things for the IESG/IAB to Learn
- - Is there a well-defined goal?
- - Do standardized solutions exist?
- - Is it technically feasible to do this work in a reasonable time?
- - Are there people that can do this work in the IETF?
- - If the work is completed, would it be used?

* SLIDE 2: Patent Related IPR at the IETF
- - If formed, working group would function within IETF IPR policy.
- - Not looking to change IETF IPR policy.

* SLIDE 3: BCP 79
- - [quotation from BCP 79]

* SLIDE 4: Goal is Royalty Free
- - from BCP 79: "In general, IETF working groups prefer technologies
with no known IPR claims..."

* SLIDE 5: No Mandate of Royalty Free
- - from BCP 79: "... working groups have the discretion to adopt
technology ... with no licensing commitment, if they feel that
this technology is superior..."

Xavier Marjou @ mic: Question about overlap of standardization bodies,
referring to RFC 2418 (BCP 25).

Cullen Jennings @ mic: Yes, it overlaps and we have a procedure for
handling this with other liaison SDOs. Generally the IAB focuses on
this. We need to make sure that our processes are consistent with that
BCP, and we're doing that. We have already received two liason requests
from two relevant organizations. New work is discussed on a mailing
list read by other SDOs and issues are addressed between SDOs and IAB
before deciding to form a working group.

Patrik FÃƒÂ¤ltstrÃƒÂ¶m @ mic (as liaison to the ITU-T): I am the liason to
the ITU-T and I am working together with the ITU-T on this.

========================================================================

3. GOALS OF THE WORKING GROUP (11:18)

3a. Chairs on the goals

Jean-Marc Valin presenting...

* SLIDE 1: Today's Codec Landscape

Jean-Marc (presenting): A representative list of some relevant codecs
received from the ITU-T liaison.

Henry Sinnreich @ mic: Why is the SILK codec not in the list?

Jason Fischl (Chair): The list is not comprehensive.

[unidentified] @ mic: There might be others that are royalty-free.

Jean-Marc Valin presenting...

* SLIDE 2: Applications

Ted Hardie @ mic: How fixed is this list? Is this application list
comprehensive or might there be others? AppArea might have a different
set of requirements.

Slava Borilin @ mic: Just an initial list to given an idea of the scope.
Defining a list of target applications would be part of the working
group's task, if it is approved.

Roni Even @ mic: The charter says only doing "interactive" application,
but not "streaming" applications (which might be similar).

Ted Hardie @ mic: Is this list constraining or not?

Jason Fischl (Chair): This list is intended to be a non-constraining
list of motivating examples.

Roni Even @ mic: I understood from the charter that there are some
limitations.

Jason Fischl (Chair): There is a slide later that defines what is out of
scope.

Joe Hildebrand @ mic: There are a number of AppArea people here as well
and folks should take requirements from applications into account.

Jason Fischl presenting...

* SLIDE 3: Goals
- - Widely and easily distributable and accessible standard codecs
- - Small number of codecs (1 or 2) that cover the operating space

Jason Fischl (presenting): Not proposing to start a "codec factory".

Stefan Bruhn @ mic: What is so unique with these goals for the IETF?
There are other SDOs who do work in this area. Why are we reformulating
these goals here in the IETF?

Harald Alvestrand @ mic: Is the operating space meant to be interactive
voice as opposed to stored voice?

Jean-Marc Valin (Chair): The scope is meant to be interactive audio,
not just voice.

Monty Montgomery @ mic: Within the IETF we would be solving problems
that IETFers have. Also gives us an opportunity for cross area review.
Feedback from application developers, transport protocol designers, etc.

Stefan Bruhn @ mic: I don't understand the motivation. We have the MPEG
(system agnostic), ITU-T (fixed line), 3GPP (wireless transports), so
we have the whole range, I don't see why we need to address this in the
IETF. What is so specific here to the Internet that it can't be
addressed in MPEG, ITU-T, or 3GPP?

Peter Saint-Andre @ mic: The IETF has an open participation model in
which the entire Internet community can participate, many other SDOs
and industry consortia do not have that model.

Monty Montgomery @ mic: The development model in the IETF culturally
matches the way we work in the developer community. BCP 79 avoids an
accidental, systemic bias against producing royalty-free codecs. Here
we have a fighting chance of doing something truly open.

Joe Hildebrand @ mic: It's a valid point that applications considered
at the IETF have requirements that applications considered by other
SDOs don't have.

Colin Perkins @ mic: I was chair of AVT WG. You need to pay attention to
the way the Internet works. I don't know if that is an argument about
whether to form a codec working group, but it is an argument that those
who design codecs need to interact with those who understand Internet
protocols. One could potentially make the argument that those who work
on codecs should interact heavily with the IETF. Maybe that doesn't
happen in other SDOs.

Christian Hoehne @ mic: Need to consider other requirements than have
traditionally been considered by other SDOs.

Colin Perkins @ mic: I'm not saying that other SDOs have not been
involved in that kind of interaction with the IETF, but that it doesn't
always happen.

Roni Even @ mic: IETF should produce requirements for submission to
other SDOs, which have the expertise to do the codec work. As to open
participation, anyone is welcome at ITU-T rapporteur meetings who is
approved by rapporteur (but they cannot vote).

Xavier Marjou @ mic: I confirm what Roni just said. I also add that
there is a liaison relationship between ITU-T and IETF, and the ITU-T
is open to taking on new work.

Anisse Taleb @ mic: The ITU-T or even MPEG are not less open than the
Internet community. Individuals are invited to submit requirements, and
MPEG calls for proposals are open to companies that are not members of
MPEG. I agree as well with Stefan Bruhn -- I don't see what is different
here in terms of requirements.

Joe Hildebrand @ mic: Definition of open used at IETF does not overlap
with needing to be invited to participate.

Roni Even @ mic: You need to pay to attend an IETF meeting.

Joe Hildebrand @ mic: These meetings are not canonical, it is the open
mailing lists that are canonical.

Roni Even @ mic: It is the same in the ITU-T.

Patrik FÃƒÂ¤ltstrÃƒÂ¶m @ mic (as ITU-T liaison): I'm a bit concerned about the
language being used at the mic. We have agreement between ITU-T and IETF
on how to communicate and how to cooperate regarding the possibility of
doing overlapping work. We are following that process in the IETF and
the ITU-T is following its process. It is not part of the process to
have the kind of discussion we are having in the room, so please stop
and let us move on to more productive topics.

Jean-Marc Valin presenting...

* SLIDE 4: Technical considerations
- - suitability for most fixed-point digital signal processors
- - medium- to high-quality speech at moderate bit-rates
- - high-quality music encoding at higher bit-rates
- - sampling rate profiles from narrow-band to full audio bandwidth
- - real-time adaptive bit-rate
- - source-controlled variable bit-rate (VBR)
- - low algorithmic delay

Christian Hoehne @ mic: What's the background behind the choice of
the chosen technical considerations? This looks like a description
of CELT and SILK.

Jean-Marc Valin (presenting): These considerations were originally
posted in the first proposed charter and adapted based on feedback.
They are by no means final and are open to modification.

Roni Even @ mic: This is a typical list for codec requirements, nothing
specific here to the Internet.

Jason Fischl (Chair): These are technical considerations, there are
other Internet-related considerations.

Ted Hardie @ mic: Is the goal to "pick a winner" (just one or two). Is
there a desire to avoid transcoding? We might want that for technical
reasons, such as avoiding delay or preventing fragility. But let's be
careful that the way to avoid transcoding is not to pick a winner
because then we have a collusion issue. Must be very open about what our
needs are to avoid the appearance of collusion. If this is not the
issue, they do we really need to avoid the codec factory approach?

Colin Perkins @ mic: I agree with Roni Even -- there is nothing here
that is particular for the Internet. Surely there must be technical
factors from the Internet transport that are relevant here.

Francois Audet @ mic: I have a similar point. How does this relate to
SIP or SDP? Issues like end-to-end negotiation of SDP parameters are
important. In-band or out-of-band? Not good to take the most general
approach. What are the IETF specifics? Identify what things IETF is
best at and include those in the scope.

Monty Montgomery @ mic: Transcoding points are excellent. It would make
the system better to avoid transcoding. Second point, those technical
considerations are items that are interesting to those of us who want to
do the work, and this work is not being done elsewhere.

Christian Hoehne @ mic: We haven't discussed transcoding because of the
end-to-end principle for IP packets. If you have e2e instead of some
other network principle (e.g., circuit-switched), then there's no need
for transcoding.

Mikael Abrahamsson @ mic: I'm a network engineer. I spend a lot of time
debugging VoIP traffic. Jitter, latency, packet reordering, packet
drops are Internet issues that are not commonly handled by codecs and
that cause codecs to lack robustness. I'm baffled that this is missing
from an IETF codec initiative.

Jean-Marc (presenting): This is not a complete list.

Julien Meuric @ mic: There are existing implementations that meet all
these requirements. These considerations are not IETF-specific. Why not
use or extend existing standards-based solutions?

Jason Fischl (Chair): Could you be more specific?

Julien Meuric @ mic: Transcoding not mentioned. There are standardized
tools that make this possible. What's the new issue that needs to be
solved?

Slava Borilin @ mic: Existing codecs don't meet these requirements,
either not easily distributable or not high quality. I agree that the
Internet-specific requirements need to be included. Those of us who
worked on the slides probably assumed that the Internet-specific issues
were so obvious that unfortunately we omitted from the slides. New work
will potentially be an extension of something existing.

Anisse Taleb @ mic: These are kind of default considerations for codec
work anywhere. What particular features would an Internet codec have?
Be more specific about what requirements existing codecs do not meet.
Need evidence. Robustness to frame losses is an important technical
consideration, for example.

Jean-Marc Valin (presenting): The requirement for robustness to frame
losses is in the charter but we cut it off the list to fit on the
slides.

Anisse Taleb @ mic: Best way to avoid transcoding is to stop
standardizing codecs.

Keith Drage @ mic: You don't have a slide about non-goals.

Jason Fischl (Chair): We do, it's next.

Slava Borilin @ mic: Two requirements, easily distributable and good
performance. Those are not met now.

Jason Fischl (Chair): Let's move on to the next couple of slides, then
have more comments.

Jean-Marc Valin presenting...

* SLIDE 5: What is out of scope?
- - Very low bitrate
- - Non-interactive music (e.g., MP3, AAC, Vorbis)
- - Unlimited number of codecs

Jason Fischl presenting...

* SLIDE 6: Why do this here?
- - IETF is where you find the people who understand the impact of the
  IP stack on codec design
- - Cross-area review
- - Whole Internet Community can participate
- - IETF will have change control and the resulting codecs will
  be different and better than the initial candidates

Stefan Bruhn @ mic: Are you saying that in other SDOs there are no
people who understanding the IP stack?

Jason Fischl (presenting): Not saying anything about other SDOs, the
only statement is that we have such people at the IETF.

Stefan Bruhn @ mic: I don't see what is so unique in the IETF. We have
all these things (cross-area review, whole Internet community can
participate) in other SDOs.

Keith Drage @ mic: I want to make sure there is a non-goal that we're
not trying to define the default codec for the Internet, and that the
resulting codec is not going to be mandated.

Roni Even @ mic: I don't see codec expertise on this slide. Also, IETF
doesn't mandate codec, this is done by other forums that create
profiles. Free codecs are available. Why do you need a standard codec?

Alan Duric @ mic: As an operator, I've been struggling for 6 years to
deploy VoIP and none of the vendors want to deploy Speex because it's
not a standard. If we don't have this work done as a standard, operators
will not be able to convince vendors to deploy it. G.722 is falling
apart on IP.

Xavier Marjou @ mic: We have already deployed a wideband audio codec
over IP. I think IP considerations are important, but other areas come
into account in codec design.

Alan Duric @ mic: Give me the evidence regarding G.722. Please post to
mailing list what these codecs are that meet these requirements.

Adam Roach @ mic (quoting from XMPP BoF minutes held at IETF 54):
"We should be chartering work based on the number of folks who want to
do things, not the number opposed." I think we've heard from a number of
people here who want to do the work and have competence in this area.

[unidentified] @ mic: [Comment about G.719?]

Patrik FÃƒÂ¤ltstrÃƒÂ¶m @ mic (as liaison to the ITU-T): Regarding the comment
that we should start work if there are enough people willing to do the
work: there are two different discussions here. One is the IETF/IESG
decision about whether to create a working group. There is a separate
discussion between the IETF and the ITU-T according to RFC 3356, the
agreed-upon procedure between IETF and ITU-T to determine if there is
overlapping work. That discussion is happening and will continue after
this BoF.

Ingemar Johansson @ mic: There is no problem with interaction between
SDOs. Let's define payload formats here and leave codec work to the
ITU-T.

Xavier Marjou @ mic: Why are people in the IETF saying that standards
coming from other SDOs a joke? Does putting "IP stack" in the slide make
the IETF qualified?

Monty Montgomery @ mic: Regarding why to do this within the IETF: there
are people who want to do this, are doing this, and who want to do it
through the IETF. That's not true elsewhere.

Christan Hoene @ mic: Things have to work together. This is a reason to
do the work here. Are there enough Internet, codec, and VoIP experts
sitting together in the same room? That is the question, and we need to
decide whether that can be done in the IETF, between the IETF and ITU-T,
or in some other way. I think it can be done in the IETF, but no matter
what we need to start this work.

Slava Borilin @ mic: Regarding why IETF and why standardized codec:
Because IETF management of change control is important for vendors.
Once change control moves from corporation / vendor to the IETF it
becomes safer for others to implement and deploy. The IETF has a group
of people that are willing to contribute, skilled and able to
contribute. We have expertise here. The codec has to be developed with
all aspects of the transport and IP stack considered for high quality on
the Internet. Existing codecs do not address that factor, and the few
codecs that are close to meeting those goals can't be freely distributed.
We have goals of freely distributable and high quality in the real
Internet. The IETF is the best organization that fits meeting both of
those goals.

Jason Fischl @ mic (as individual): How to we get wideband codecs
broadly adopted? G.722 is brought out as an example and is starting to
become actively deployed. The trigger is expiration of the patents. We
haven't seen widely deployed codecs before. Widely distributed and
freely available codecs lead to broad deployment. The IETF can try to
achieve that, although there is no guarantee.

Wilhelm Wimmtreuter @ mic: We have people willing and able to contribute.
It matters who can fund it and support it. Why not let them do it?

Stefan Bruhn @ mic: I doubt that you have the expertise in this group.
Not traditionally at the IETF. Why are you not proposing this to the
ITU-T, even royalty-free, and face competition?

Slava Borilin @ mic: Because most of our stakeholders are here in the
IETF.

Jean-Marc Valin presenting some slides about codecs that have been
submitted as Internet-Drafts so far...

* SLIDE 7: CELT Codec Characteristics

* SLIDE 8: SILK Codec Characteristics

* SLIDE 9: IPMR Codec Characteristics

* SLIDE 10: BV32 Codec Characteristics

* SLIDE 11: A2DP SBC+PLC

Dave Oran @ mic: Loss robustness is not a boolean. And it is not a
scalar. Loss is a curve, not a scalar value.

Colin Perkins @ mic: How is it relevant to the Internet that we have a
list of codecs? How do these codecs solve Internet-specific problems?

Roni Even @ mic: What is the point of a list of codecs?

Jason Fischl (Chair): Trying to point out that there were a number of
submissions of real codecs.

Roni Even @ mic: That's interesting, but there are no requirements so
everyone is qualified.

Peter Saint-Andre @ mic (as individual, not Jabber scribe): People say
there is no competence in the IETF. The point of the list of codecs is
saying that we do have competence among the people who have developed
these codecs and contributed them to the Internet Standards Process.

Xavier Marjou @ mic: What is really new in terms of new requirements?

Jason Fischl (Chair): These have been submitted in the open and intended
to be open source.

Xavier Marjou @ mic: For a new codec there are likely patents coming.

Colin Perkins @ mic: I question the claim that they are royalty free.

Jason Fischl (Chair): Claim is not that they are royalty-free, but that
they are being provided without royalties on any of the patents. [?]

Slava Borilin @ mic: People are contributing codecs that are believed to
the best of their knowledge to be royalty free, and these people are
contributing / have expertise to contribute royalty free codecs that are
superior to existing alternatives. Based on what we know, we stand a
good chance of doing better than what is out there, and keeping it
freely available. We believe that this can be achieved such that quality
is as good or better than existing codecs, and to do so in a way that is
freely and easily distributable.

Alan Duric @ mic: As to competence, there are people in this room who
have developed codecs that run into the hundreds of millions of minutes
each day or each week. These codecs are superior to the existing codecs
out there and none of those are suitable for Internet or wireless LAN
if there is any kind of packet loss.

Peter Saint-Andre @ mic (as individual, not Jabber scribe): Could we
please try to get through the agenda?

========================================================================

4. ENGINEERING WORK (12:22)

Jason Fischl (Chair): Quick presentations on SILK and CELP...

4a. Koen Vos presenting about the SILK codec (12:22)

(draft-vos-silk-00)

4a. Monty Montgomery presenting about the CELT codec (12:24)

(draft-valin-celt-codec-00)

"If CELT comes out of a proposed working group intact, I will be sorely
disappointed."

Emphasizes the point that all of these codec developers want to merge
their work to produce a best-of-breed codec.

========================================================================

5. CODEC EXPERTISE IN THE IETF (12:27)

Jason Fischl (Chair): These brief presentations were trying to establish
that we do have codec expertise among IETF participants, but there are
also other kinds of contributions -- requirements, testing, etc.

Joel Halpern @ mic: I'm not going to question whether there are enough
people here to do the codec work, because that's clearly true. The
question is whether the *IETF* has the expertise. It's only sensible
for the IETF to do it if it works with the rest of the process. Can the
reviewers provide appropriate technical insights? Does the IESG have
expertise to provide oversight? If we don't have expertise in those
places, we can't do this as an organization.

Henny Sinnreich @ mic: I'm willing to contribute with usage scenarios
for Rich Internet Applications and the resulting requirements.

Slava Borilin @ mic: We definitely have enough people to work on the
codecs themselves, but do have enough qualified customers? I think so.

Colin Perkins @ mic: I do not believe the wider IETF can meaningfully
review the work (IESG and other areas).

Bernard Aboba: We have chartered IETF working groups in this kind of
situation before. For example, TRILL. We only looked at whether we had
people who could do the work. What we put into the charter is review by
appropriate outside experts and liaisons (in that case IEEE) to get
comments. We have been successful in that approach. The only thing
that was relevant is whether people are here with expertise and willing
to do work.

Roni Even @ mic: It's more complicated. You need to provide information
correctly to liaisons and outside experts. For example, need to work on
characterization.

Christan Hoene @ mic: There need to be experts on judging the quality
of the codecs. For example, there are standards for conducting listening
tests.

Jason Fischl (Chair): I want to make sure we are very focused on the
question of whether we have the expertise to do this work in the IETF.

Jean-Francois Mule @ mic: I do believe there is expertise. Anyone who
has done codec development and testing knows how to define test vectors
so that implementers can validate implementations. In this room we have
4-5 companies with codec test bench labs that could validate that the
codec has been implemented correctly according to the spec and also that
you have met your terms of reference and requirements. Also independent
labs. There are plenty of resources, and this is not rocket science.

Slava Borilin @ mic: At Spirit DSP have expertise in design and we are
engaged on the Internet application side. A working group can serve
most of all as an analyst to specify the right tasks for developing
the kind of codec that is needed for Internet applications. The major
value is identifying the technical and business requirements and
developing technical solutions to meet those requirements. The working
group might even end up with a decision that G.722 is the best choice.

Monty Montgomery @ mic: Two hours ago the argument was that we didn't
have the competence. That has shifted to "you are competent to design a
codec but you don't have the competence to test". In my experience,
testers outnumber implementers 10 to 1. This is irrelevant.

Adam Roach @ mic for Eric Rescorla in the Jabber chatroom: I'm not
concerned about Joel's point. We have similar problems in security,
where the ability to evaluate cryptographic techniques outside the
working group is extremely limited. But at least the performance of
codecs is measurable, whereas with security even that isn't true.

Adam Roach @ mic: At least codecs can be evaluated directly by everyone
in the IETF, since we all have working ears.

Stefan Bruhn @ mic: Do we have expertise? Depends on the companies
sending their experts to this group. Experts are sent to ITU-T and MPEG
and 3GPP, are you sure they will be sent here? For experts here, why
aren't you at MPEP and ITU-T and 3GPP? Are you afraid of facing
competition?

Derek McDonald @ mic: It seems that we do have the expertise. But the
IETF retains the right to fail. It isn't like we've lost a war if the
working group fails. Don't let FUD block us.

Mikael Abrahamsson @ mic: The Internet should demand things from the
codec rather than the codec demanding things from Internet. We know IP,
we know how the Internet works.

Alan Duric @ mic: I fully agree, and if we brought this to ITU-T or
ETSI, yes these proposals were brought forward 8 years ago and shot
down without consideration.

Julien Meuric @ mic: People are interested, but do you have any proof
that the technical skill is present here?

Roni Even @ mic: We are trying to define the whole process, need
expertise in defining the whole process. But really need a beauty
contest. Either approve all the specs that are submitted or you need
to select a few and define how to do that.

Xavier Marjou @ mic: IETF documents are good because they are so
readable, but a codec RFC would be unreadable. Can't write formulas in
an Internet-Draft.

Jean-Francois Mule @ mic: Responding to request for evidence, Broadcom
is willing to bring expertise and competence. Dr. Chen and his group
have worked for many years in ITU-T.

Slava Borilin @ mic: Spirit DSP has developed more than 10 proprietary
codecs standardized under various bodies like radio standards. We are
not going to ITU-T because there are no stakeholders interested in the
result (freely distributable, high quality). In IETF there are.

Monty Montgomery @ mic: First, we will mix inputs from different
contributors. Second, Roni is right that a lot of characterization,
testing, and documentation needs to be done. We're aware of that and
we have a lot of work to do.

Ye-Kui Wang @ mic: Who in IESG is capable of reviewing audio codecs?

Ted Hardie @ mic: Please stop talking about stakeholders -- not a useful
construct in the IETF. The decision is whether to charter work that
would benefit the Internet. Serious question is whether this is the best
place and whether this work benefits the Internet. Stakeholders are
users, operators, vendors, etc. Please try to take a whole Internet view
rather than a particular stakeholder view or we will ask ourselves the
wrong question.

Cullen Jennings @ mic, as AD who will be asked to evaluate codecs: Do
we have competence on this topic? The IESG relies a lot on expert
advisers across areas. We try to assess whether things cause harm. We
have to listen to other people and try to evaluate whether claims of
brokenness are legitimate or not. Often our expertise lies in finding
the right people to provide input and in listening to them in order to
judge which input is reliable. For example, will have to review IDNAbis
(internationalized domain names) but there are no linguistics experts
on the IESG, so we will have to rely on other people and work with them
and try to integrate their feedback. Another factor is that IESG members
are selected based on skills needed to do the work and review it. This
codec work is really not different from other work and the important
thing is that there's expertise to get the right comments so the IESG
can make good decisions.

[12:50, T-10 minutes]

QUESTION from Chairs: Who here would call themselves codec experts and
would be willing to develop codecs?

Outcome: 10-12 real hands and 5+ virtual hands from the Jabber room.

QUESTION from Chairs: Who is willing to participate in any way listed on
slide 1 (guidelines, process, requirements, codec proposals, commenting,
evaluation)?

Outcome: 40-60 real hands and virtual hands.

Lars Eggert @ mic: Of the codec experts, would you still participate
if a proposal you contributed is ruled out by the working group?

Outcome: Same as for the first question.

HUM REQUEST from Chairs: Is or is not the technical scope of the work
well defined?

[one hum for yes, one hum for no]

Outcome: Inconclusive.

HUM REQUEST from Chairs: Do we or do we not have enough expertise to
form a working group that can meet the goals?

[one hum for yes, one hum for no]

Outcome: Inconclusive.

Ted Hardie @ mic: Large grey area here, what if you have no idea if
there's enough expertise to complete the work because you don't know if
it's well-scoped in the first place? Perhaps a separate hum is in order.

Cullen Jennings @ mic: Could you type the question in a slide? My
experience from other BoFs is that the question can get very confusing.

[12:58 conclave at front of room among Chairs, AD, BoF Sponsor, etc.]

HUM REQUEST from Chairs: Do we or do we not have enough expertise to
properly scope the work?

[one hum for yes, one hum for no]

Outcome: Rough consensus in favor.

[conclave continues]

Cullen Jennings @ mic as Area Director:

- - I think we've identified that it's prematuve to ask whether to form a
  working group
- - We have the information we need from the BoF and will go forward with
  discussions on the mailing list

[meeting adjourned at 13:01]

========================================================================
6. Gauging Likelihood of Success

- - not discussed -

========================================================================
7. Call for Consensus on Whether to Form a WG

- - not discussed -

========================================================================
8. Charter Discussion

- - not discussed -

========================================================================
9. Conclusions and Next Steps

- - not discussed -

END



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From jean-marc.valin@octasic.com  Fri Sep  4 12:07:48 2009
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Subject: [codec] Codec guidelines and requirements drafts
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Hi,

As a follow-up to the codec BoF in Stockholm, a group of interested IETF 
participants has been working on how to move forward regarding this work at 
the IETF.  As a result, today we have submitted two Internet-Drafts:

(1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
(2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt

We welcome feedback on these drafts on the codec@ietf.org list.

Cheers,

	Jean-Marc

From stpeter@stpeter.im  Fri Sep  4 12:17:36 2009
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Subject: [codec] [Fwd: I-D Action:draft-valin-codec-guidelines-00.txt]
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FYI.

/psa

- -------- Original Message --------
Subject: I-D Action:draft-valin-codec-guidelines-00.txt
Date: Fri,  4 Sep 2009 11:00:01 -0700 (PDT)
From: Internet-Drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : Guidelines for proposed CODEC working group
	Author(s)       : J. Valin, et al.
	Filename        : draft-valin-codec-guidelines-00.txt
	Pages           : 17
	Date            : 2009-09-04

This document provides general guidelines for specifying audio codecs
within the IETF.  These guidelines cover the development process, as
well as the evaluation and intellectual property issues.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-valin-codec-guidelines-00.txt


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From swmike@swm.pp.se  Fri Sep  4 12:49:45 2009
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From: Mikael Abrahamsson <swmike@swm.pp.se>
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On Fri, 4 Sep 2009, Jean-Marc Valin wrote:

> Hi,
>
> As a follow-up to the codec BoF in Stockholm, a group of interested IETF 
> participants has been working on how to move forward regarding this work at 
> the IETF.  As a result, today we have submitted two Internet-Drafts:
>
> (1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
> (2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt
>
> We welcome feedback on these drafts on the codec@ietf.org list.

Hi,

from my network engineering side these look good apart from the fact that 
they do not mention adaptive jitter buffer for the transport. Is this an 
omission done because these drafts are codec and not transport centric, or 
has this requirement been overlooked?

I firmly belive (as I also stated at the mic at IETF75) that any Internet 
codec+transport needs to gracefully handle initial jitter of 10 ms or less 
with small (perhaps 20-40 ms) jitter buffer during parts of the call, and 
then it needs to adapt to jitter going up to multiple seconds (thus 
inferring multiple second jitter buffer and end-to-end delay), and then 
when conditions on the IP packet layer improve, it needs to gracefully 
adapt back to much smaller jitter buffer again.

This is because on the packet layer we have packet transport networks that 
do re-transmits itself and thus can deliver packets quite late 
(802.11abgn, 2G/3G GSM/UMTS networks, just to name a few), but they still 
get delivered but with huge delay/jitter spikes (can be for prolonged 
periods of time as well).

I believe these conditions should be in the requrements/guidelines since 
Packet Loss Concealment is there? (unless people disagree with my opinion 
that any Internet codec+transport should handle and behave the way I 
described above, of course)

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

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It has been brought to my attention that my previous message was
unreadable in certain email user agents. The minutes will eventually be
posted on the IETF website, but until then you can find them here:

http://www.saint-andre.com/tmp/codec-minutes.txt

/psa
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From jean-marc.valin@octasic.com  Fri Sep  4 13:03:28 2009
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Hi,

Mikael Abrahamsson wrote:
> from my network engineering side these look good apart from the fact 
> that they do not mention adaptive jitter buffer for the transport. Is 
> this an omission done because these drafts are codec and not transport 
> centric, or has this requirement been overlooked?

I agree that adaptive jitter buffering is a very important aspect of a 
complete voice transport system. The only reason that this is not mentioned 
in this particular draft is that we are only addressing codec requirements. 
In general, an adaptive jitter buffer should be able to work for any codec 
(you don't want a separate jitter buffer for each codec). The only part 
that *is* relevant for codec requirements is any aspect that the jitter 
buffer would want from the codec. For example, when a packet is late or 
lost, the jitter needs to conceal the loss and make sure to minimise its 
propagation in time. This is a job for the codec and is part of the 
requirements.

I really do think that there is important work to be done regarding jitter 
buffers and transport, but I think these should be done in a separate place 
(AVT WG?). Of course, the people who are working on that can (and should) 
talk to the codec people and tell them what "services" the codec can 
provide that would improve the overall audio quality. For example, some 
people have suggested that codecs should be able to facilitate time 
stretching/compressing. While this is not something that is currently in 
the requirements, I could very well imaging adding that (once there is 
agreement on the details).

	Jean-Marc

From stpeter@stpeter.im  Fri Sep  4 13:06:07 2009
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Subject: [codec] [Fwd: I-D Action:draft-valin-codec-requirements-00.txt]
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FYI.

/psa

- -------- Original Message --------
Subject: I-D Action:draft-valin-codec-requirements-00.txt
Date: Fri,  4 Sep 2009 11:00:01 -0700 (PDT)
From: Internet-Drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts
directories.

	Title           : Codec Requirements
	Author(s)       : J. Valin, et al.
	Filename        : draft-valin-codec-requirements-00.txt
	Pages           : 20
	Date            : 2009-09-04

This document provides specific requirements for Internet audio
codecs.  These requirements address quality, sampling rate, bit-rate,
and packet loss robustness, as well as other desirable properties.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-valin-codec-requirements-00.txt


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From swmike@swm.pp.se  Fri Sep  4 13:14:01 2009
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From: Mikael Abrahamsson <swmike@swm.pp.se>
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On Fri, 4 Sep 2009, Jean-Marc Valin wrote:

> that would improve the overall audio quality. For example, some people have 
> suggested that codecs should be able to facilitate time 
> stretching/compressing. While this is not something that is currently in the 
> requirements, I could very well imaging adding that (once there is agreement 
> on the details).

That sounds good, I think this is a key requirement. I believe 
stretching/compressing needs to be fairly fast/agressive, to be able to go 
from 1000 to 40 ms jitter buffer in just a few seconds (and as fast as 
needed when increasing it). The spatial compression could perhaps also be 
done via agressive silence suppression (at least for voice communication), 
but this can't be the only mechanism because it won't work if there is 
never silence.

Does anyone more into AVT WG know if there are already IETF 
standards/framework to handle this at the transport layer, or is there a 
lot of development needed there as well?

How do we assure that the end-to-end solution works when all is said and 
done? I'm a bit afraid that unless there is very tight cooperation, the 
end solution won't be as good as it could be if all of this was done in a 
single WG doing all the requirements and making sure that everything works 
together.

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

From jean-marc.valin@octasic.com  Fri Sep  4 13:27:31 2009
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Mikael Abrahamsson wrote:
> That sounds good, I think this is a key requirement. I believe 
> stretching/compressing needs to be fairly fast/agressive, to be able to 
> go from 1000 to 40 ms jitter buffer in just a few seconds (and as fast 
> as needed when increasing it). 

Again, it would be the jitter buffer's job to decide how aggressive to be. 
The only thing the codec (actually just the decoder) would do at this point 
is "obey" what the jitter buffer tell it to do.

> Does anyone more into AVT WG know if there are already IETF 
> standards/framework to handle this at the transport layer, or is there a 
> lot of development needed there as well?

Actually, an even more fundamental question here is "should there be a 
standard for jitter buffers?". My *personal* opinion on this is that we 
probably want to define "best practices" for jitter buffering (possibly 
with reference code), but we do not need an actual standard because at this 
point I do not see any inter-operability issue.

> How do we assure that the end-to-end solution works when all is said and 
> done? I'm a bit afraid that unless there is very tight cooperation, the 
> end solution won't be as good as it could be if all of this was done in 
> a single WG doing all the requirements and making sure that everything 
> works together.

Well, you can cooperate across groups and in fact all IETF groups *have* to 
cooperate with other groups. But at the same time, you need to divide the 
problem otherwise the entire IETF would be a single working group ;-)

Cheers,

	Jean-Marc

From Borilin@spiritdsp.com  Fri Sep  4 13:30:10 2009
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Mikael Abrahamsson" <swmike@swm.pp.se>, <codec@ietf.org>
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Mikael,

As it has been already discussed on the list, the goal is not to
design/specify "end-to-end" voice system, but only specify what is
MANDATORY for interop (=3Dcodec), and leave the upper layers to
proprietary solutions.

Of course, codec should be OPTIMIZED for such proprietary solutions as
much as possible, and that's why we are working on the requirements from
"overall system" perspective, which, however, should not be read that
"codec has to do everything", it should be read as "codec has to be good
for the systems that do the best practice on upper layers".

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Mikael Abrahamsson
Sent: Saturday, September 05, 2009 12:14 AM
To: codec@ietf.org
Subject: Re: [codec] Codec guidelines and requirements drafts

On Fri, 4 Sep 2009, Jean-Marc Valin wrote:

> that would improve the overall audio quality. For example, some people
have=20
> suggested that codecs should be able to facilitate time=20
> stretching/compressing. While this is not something that is currently
in the=20
> requirements, I could very well imaging adding that (once there is
agreement=20
> on the details).

That sounds good, I think this is a key requirement. I believe=20
stretching/compressing needs to be fairly fast/agressive, to be able to
go=20
from 1000 to 40 ms jitter buffer in just a few seconds (and as fast as=20
needed when increasing it). The spatial compression could perhaps also
be=20
done via agressive silence suppression (at least for voice
communication),=20
but this can't be the only mechanism because it won't work if there is=20
never silence.

Does anyone more into AVT WG know if there are already IETF=20
standards/framework to handle this at the transport layer, or is there a

lot of development needed there as well?

How do we assure that the end-to-end solution works when all is said and

done? I'm a bit afraid that unless there is very tight cooperation, the=20
end solution won't be as good as it could be if all of this was done in
a=20
single WG doing all the requirements and making sure that everything
works=20
together.

--=20
Mikael Abrahamsson    email: swmike@swm.pp.se
_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

From simon.perreault@viagenie.ca  Fri Sep  4 13:37:20 2009
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On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:
> For example, some
> people have suggested that codecs should be able to facilitate time
> stretching/compressing.

Does someone have a precise idea of what that means?

Thanks,
Simon
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From bmschwar@fas.harvard.edu  Fri Sep  4 14:11:27 2009
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Simon Perreault wrote:
> On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:
>> For example, some
>> people have suggested that codecs should be able to facilitate time
>> stretching/compressing.
>=20
> Does someone have a precise idea of what that means?

How precise?

Audio data arrives at some apparent rate, which we can measure in samples=

per second on the local system sample clock.  If the link latency and
buffering are constant, and the two endpoints' sample clocks tick at the
same rate, then the audio data will arrive at the "right" speed, and can
simply be decoded and thrown into the sound card's output buffer.

If the buffer has to expand or contract over some period of time, or if
one of the endpoints ticks faster than the other, then fewer or more
samples will be received than the sound card requires to be played.  The
codec must provide some mechanism to resolve this discrepancy.  Commonly
suggested mechanisms include resampling (which bends pitch), smart
time-stretching to preserve pitch, and "adjust during silence".

To me, the important question here is: what sort of interface does the
codec need to expose for time dilation? For example, it might be a
function that says "turn the next N coded samples into M decoded samples,=

please".

--Ben


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From ron@debian.org  Fri Sep  4 16:10:21 2009
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On Fri, Sep 04, 2009 at 04:03:47PM -0400, Jean-Marc Valin wrote:
> For example, some people have suggested that codecs
> should be able to facilitate time stretching/compressing.

This would be a very attractive feature for the decoder to support
directly, as we increasingly move toward transports where it is
impossible or impractical to achieve precise clock synchronisation
between endpoints.

While it's not impossible to do this external to the codec, I'd be
quite surprised if gains could not be made to both computational
complexity and quality if this were to be factored into the decoder
stage and encoded data from the outset.

  Ron



From stephen.botzko@gmail.com  Fri Sep  4 18:32:56 2009
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--00151747afa4975d3b0472ca99a7
Content-Type: text/plain; charset=ISO-8859-1

I agree that there is a need for receivers to resample (or otherwise
drop/add samples) in order to match the encoder's A/D clock rate with the
receiver's D/A clock rate. If normal commercial crystals are used for clock
sources, the A/D and D/A clocks may be mismatched by up to about 10 samples
per second (for 48 khz audio).

I am puzzled as to why you would want to build this resamplng into the codec
however - I see no technical advantage in doing this

The resampling is not related to the decoder operation itself, and needs to
be done no motter what codec is in use..Some form of fixed resampling is
often required anyway, in order to match (for instance) a 32 kHz sample rate
used by a codec to the 48 kHz sample rate used by the sampling hardware.

Stephen Botzko
Polycom

On Fri, Sep 4, 2009 at 5:11 PM, Benjamin M. Schwartz <
bmschwar@fas.harvard.edu> wrote:

> Simon Perreault wrote:
> > On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:
> >> For example, some
> >> people have suggested that codecs should be able to facilitate time
> >> stretching/compressing.
> >
> > Does someone have a precise idea of what that means?
>
> How precise?
>
> Audio data arrives at some apparent rate, which we can measure in samples
> per second on the local system sample clock.  If the link latency and
> buffering are constant, and the two endpoints' sample clocks tick at the
> same rate, then the audio data will arrive at the "right" speed, and can
> simply be decoded and thrown into the sound card's output buffer.
>
> If the buffer has to expand or contract over some period of time, or if
> one of the endpoints ticks faster than the other, then fewer or more
> samples will be received than the sound card requires to be played.  The
> codec must provide some mechanism to resolve this discrepancy.  Commonly
> suggested mechanisms include resampling (which bends pitch), smart
> time-stretching to preserve pitch, and "adjust during silence".
>
> To me, the important question here is: what sort of interface does the
> codec need to expose for time dilation? For example, it might be a
> function that says "turn the next N coded samples into M decoded samples,
> please".
>
> --Ben
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>
>

--00151747afa4975d3b0472ca99a7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I agree that there is a need for receivers to resample (or otherwise drop/a=
dd samples) in order to match the encoder&#39;s A/D clock rate with the rec=
eiver&#39;s D/A clock rate. If normal commercial crystals are used for cloc=
k sources, the A/D and D/A clocks may be mismatched by up to about 10 sampl=
es per second (for 48 khz audio). <br>
<br>I am puzzled as to why you would want to build this resamplng into the =
codec however - I see no technical advantage in doing this=A0 <br><br>The r=
esampling is not related to the decoder operation itself, and needs to be d=
one no motter what codec is in use..Some form of fixed resampling is often =
required anyway, in order to match (for instance) a 32 kHz sample rate used=
 by a codec to the 48 kHz sample rate used by the sampling hardware. <br>
<br>Stephen Botzko<br>Polycom<br><br><div class=3D"gmail_quote">On Fri, Sep=
 4, 2009 at 5:11 PM, Benjamin M. Schwartz <span dir=3D"ltr">&lt;<a href=3D"=
mailto:bmschwar@fas.harvard.edu">bmschwar@fas.harvard.edu</a>&gt;</span> wr=
ote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class=3D"im"=
>Simon Perreault wrote:<br>
&gt; On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:<br>
&gt;&gt; For example, some<br>
&gt;&gt; people have suggested that codecs should be able to facilitate tim=
e<br>
&gt;&gt; stretching/compressing.<br>
&gt;<br>
&gt; Does someone have a precise idea of what that means?<br>
<br>
</div>How precise?<br>
<br>
Audio data arrives at some apparent rate, which we can measure in samples<b=
r>
per second on the local system sample clock. =A0If the link latency and<br>
buffering are constant, and the two endpoints&#39; sample clocks tick at th=
e<br>
same rate, then the audio data will arrive at the &quot;right&quot; speed, =
and can<br>
simply be decoded and thrown into the sound card&#39;s output buffer.<br>
<br>
If the buffer has to expand or contract over some period of time, or if<br>
one of the endpoints ticks faster than the other, then fewer or more<br>
samples will be received than the sound card requires to be played. =A0The<=
br>
codec must provide some mechanism to resolve this discrepancy. =A0Commonly<=
br>
suggested mechanisms include resampling (which bends pitch), smart<br>
time-stretching to preserve pitch, and &quot;adjust during silence&quot;.<b=
r>
<br>
To me, the important question here is: what sort of interface does the<br>
codec need to expose for time dilation? For example, it might be a<br>
function that says &quot;turn the next N coded samples into M decoded sampl=
es,<br>
please&quot;.<br>
<br>
--Ben<br>
<br>
<br>_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
<br></blockquote></div><br>

--00151747afa4975d3b0472ca99a7--

From steveu@coppice.org  Fri Sep  4 19:34:27 2009
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Subject: Re: [codec] Codec guidelines and requirements drafts
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Hi Stephen,

The subject was not resampling, but time stretching. A time stretcher 
can become trivial if it is implemented within the codec, and generally 
produces better sounding results.

Steve


On 09/05/2009 09:33 AM, stephen botzko wrote:
> I agree that there is a need for receivers to resample (or otherwise 
> drop/add samples) in order to match the encoder's A/D clock rate with 
> the receiver's D/A clock rate. If normal commercial crystals are used 
> for clock sources, the A/D and D/A clocks may be mismatched by up to 
> about 10 samples per second (for 48 khz audio).
>
> I am puzzled as to why you would want to build this resamplng into the 
> codec however - I see no technical advantage in doing this
>
> The resampling is not related to the decoder operation itself, and 
> needs to be done no motter what codec is in use..Some form of fixed 
> resampling is often required anyway, in order to match (for instance) 
> a 32 kHz sample rate used by a codec to the 48 kHz sample rate used by 
> the sampling hardware.
>
> Stephen Botzko
> Polycom
>
> On Fri, Sep 4, 2009 at 5:11 PM, Benjamin M. Schwartz 
> <bmschwar@fas.harvard.edu <mailto:bmschwar@fas.harvard.edu>> wrote:
>
>     Simon Perreault wrote:
>     > On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:
>     >> For example, some
>     >> people have suggested that codecs should be able to facilitate time
>     >> stretching/compressing.
>     >
>     > Does someone have a precise idea of what that means?
>
>     How precise?
>
>     Audio data arrives at some apparent rate, which we can measure in
>     samples
>     per second on the local system sample clock.  If the link latency and
>     buffering are constant, and the two endpoints' sample clocks tick
>     at the
>     same rate, then the audio data will arrive at the "right" speed,
>     and can
>     simply be decoded and thrown into the sound card's output buffer.
>
>     If the buffer has to expand or contract over some period of time,
>     or if
>     one of the endpoints ticks faster than the other, then fewer or more
>     samples will be received than the sound card requires to be
>     played.  The
>     codec must provide some mechanism to resolve this discrepancy.
>      Commonly
>     suggested mechanisms include resampling (which bends pitch), smart
>     time-stretching to preserve pitch, and "adjust during silence".
>
>     To me, the important question here is: what sort of interface does the
>     codec need to expose for time dilation? For example, it might be a
>     function that says "turn the next N coded samples into M decoded
>     samples,
>     please".
>


From hoene@uni-tuebingen.de  Fri Sep  4 21:24:04 2009
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Hello,

We discussed the need for time stretching already three months ago
http://www.ietf.org/mail-archive/web/codec/current/msg00090.html
May I cite Koen Vos again: " For instance, we've had detailed =
discussions
about what it takes to make a codec work on the Internet. We seemed to =
agree
that the codec should support not only packet loss concealment, but also
time compression/stretching to handle network jitter. (Incidentally, =
such
time modification is not supported by any ITU, 3GPP or ETSI codec, which
complicates their use for Internet applications.)"
http://www.ietf.org/mail-archive/web/codec/current/msg00456.html
I never read anything against this requirement on this list. Thus, it =
can be
seen as a topic of clear consensus.=20

Thus, Jean Marc, I am wondering why the input from this mailing list is
ignored in your IETF requirements document- even if it is supported by =
one
of your coauthors.=20

If the pace of progress on technical issues remains as low as it is, - I =
bet
- you wouldn't get your codecs standardized in ten years time.

Christian

--------------------------------------------------------
Dr.-Ing. Christian Hoene
Interactive Communication Systems (ICS), University of T=FCbingen
Sand 13, 72076 T=FCbingen, Germany, Phone +49 7071 2970532
http://www.net.uni-tuebingen.de/


> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of
> Steve Underwood
> Sent: Saturday, September 05, 2009 4:34 AM
> To: stephen botzko
> Cc: bens@alum.mit.edu; codec@ietf.org
> Subject: Re: [codec] Codec guidelines and requirements drafts
>=20
> Hi Stephen,
>=20
> The subject was not resampling, but time stretching. A time stretcher
> can become trivial if it is implemented within the codec, and =
generally
> produces better sounding results.
>=20
> Steve
>=20
>=20
> On 09/05/2009 09:33 AM, stephen botzko wrote:
> > I agree that there is a need for receivers to resample (or otherwise
> > drop/add samples) in order to match the encoder's A/D clock rate =
with
> > the receiver's D/A clock rate. If normal commercial crystals are =
used
> > for clock sources, the A/D and D/A clocks may be mismatched by up to
> > about 10 samples per second (for 48 khz audio).
> >
> > I am puzzled as to why you would want to build this resamplng into =
the
> > codec however - I see no technical advantage in doing this
> >
> > The resampling is not related to the decoder operation itself, and
> > needs to be done no motter what codec is in use..Some form of fixed
> > resampling is often required anyway, in order to match (for =
instance)
> > a 32 kHz sample rate used by a codec to the 48 kHz sample rate used =
by
> > the sampling hardware.
> >
> > Stephen Botzko
> > Polycom
> >
> > On Fri, Sep 4, 2009 at 5:11 PM, Benjamin M. Schwartz
> > <bmschwar@fas.harvard.edu <mailto:bmschwar@fas.harvard.edu>> wrote:
> >
> >     Simon Perreault wrote:
> >     > On Friday 04 September 2009 16:03:47 Jean-Marc Valin wrote:
> >     >> For example, some
> >     >> people have suggested that codecs should be able to =
facilitate
time
> >     >> stretching/compressing.
> >     >
> >     > Does someone have a precise idea of what that means?
> >
> >     How precise?
> >
> >     Audio data arrives at some apparent rate, which we can measure =
in
> >     samples
> >     per second on the local system sample clock.  If the link =
latency
and
> >     buffering are constant, and the two endpoints' sample clocks =
tick
> >     at the
> >     same rate, then the audio data will arrive at the "right" speed,
> >     and can
> >     simply be decoded and thrown into the sound card's output =
buffer.
> >
> >     If the buffer has to expand or contract over some period of =
time,
> >     or if
> >     one of the endpoints ticks faster than the other, then fewer or =
more
> >     samples will be received than the sound card requires to be
> >     played.  The
> >     codec must provide some mechanism to resolve this discrepancy.
> >      Commonly
> >     suggested mechanisms include resampling (which bends pitch), =
smart
> >     time-stretching to preserve pitch, and "adjust during silence".
> >
> >     To me, the important question here is: what sort of interface =
does
the
> >     codec need to expose for time dilation? For example, it might be =
a
> >     function that says "turn the next N coded samples into M decoded
> >     samples,
> >     please".
> >
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From stewe@stewe.org  Fri Sep  4 21:42:45 2009
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Folks,
I read through this draft and found it, all in all, a well written document
covering a lot of (to me) agreeable aspects.  However, as you may expect, I
have a number of comments, mostly on certain language aspects :-)  If this
were a working group effort, I would phrase some of my suggestion below in
stronger terms.  As things stand, you are, of course, free to ignore them...
Regards,
Stephan


1. Section 1 speaks in several places of "freely available".  I would
recommend to be more precise.  There are just too many definitions, some of
which contradicting, of what constitutes "freedom" in our business.
2. Section 1: the above remark also goes for the definition of "open".  A
very large part of the standardization people (perhaps a majority)
associates "open" exclusively with "open process in the SDO" and not with
what the draft implicitly assumes ("open" use, whatever that, in turn, may
be).
3. Section 1, first bullet, "has inhibited": really?  I don't believe this
claim is true---certainly not in the form as made.  At least, I have not
seen a prove for this rather assertive statement.  Suggest softening the
language.
4. Section 1, final bullet: suggest replacing "will be freely available"
with "may be available without royalties or other undue burden", as none of
us has the means to guaranty something like "freely available".

5. Section 2.2, "preferably in the form of I-Ds".  I'm not aware of any
other form of input but I-Ds on which a WG or a group of people without WG
status in the IETF could act on (and that would allow documenting the
complexity undoubtly involved).  Therefore, I would suggest to remove
"preferably".  Also, publishing an I-D undoubtly triggers the BCP79
disclosure requirements---unspecified input may not.
6. Section 2.3: I read this section as a license to re-tune the requirements
document based on results obtained by tests (or even theoretical analysis)
of input proposals.  It may be necessary to do this.  However, you are
certainly aware of that such a procedure can lead to gaming and other evil
things (including unfair exclusion of proposals with IPR properties that are
within the scope of the IETF's IPR policy).  If this were a working group
charter, I would voice strong concerns over this vagueness.  For an informal
activity outside a WG, I only appeal to you to be careful in what you are
doing.
7. Section 2.6: to add something positive, I'm thrilled that people will at
least attempt to work cooperatively on codecs from very early on: this is
IMHO one of the really weak spots traditional codec standardization has had
in most of its instances.  I'm also very happy to see the "no
rubberstamping" sentence.
8. Section 2.7: developments of RTP payload formats happen in AVT.  No need
to word around that.
9. Section 3: I'm missing a statement on what happens if it turns out that
the requirements are NOT met.  I think there should be an end criteria of
some sort.
10: I can see value in a strategy of not requiring a bit-exact encoder (as I
see a value in having such a requirement---in classic telco applications,
for example).  However, I fail to see a value in not requiring bit-exact
decoding.  Is this a fixed/floating point issue?
The can of worms one the IPR side one is opening by not strictly defining
conformance cannot be overestimated.  Almost every IPR statement in the IETF
(and elsewhere) promises or grants licenses (or makes a non-assert covenant)
only for conformant implementations.  I would strongly recommend to require
bit-exact decoding, for that reason alone.  If you truly think that a soft
definition of conformance is necessary from a technical viewpoint, then I
would strongly recommend to at least define your version of conformance in a
normative, permanent document (standard's track RFC) which includes the
source code of the testing tool, and the test vectors against which soft
conformance is tested.
11: Section 5, first bullet, "open", see remark #2
12: Section 5, first bullet, "conspicuous gap", see remark #3
13: Section 5, second bullet "freely", see remark #1
14: Section 5, fourth bullet, "free" , see remark #1
15: Section 5, "BSD license": it's not a BSD license.  It's *currently* a
BSD-style license.  However, the IETF trust currently has control over this
thing and could, in theory, change it.  Recommend to read the relevant
documents and come up with a precise and future-proof formulation.  At this
time of writing I do not have access to the documents in question (writing
on a plane), but I can help in this.
16: Section 5, "royalty-free".  Finally, this reasonably precise term is
used.  Why not earlier?  However, just before that, the subject of open
source compatibility is mentioned as well (though not in this formulation).
Please note that it is very easy to draft a patent license that is
royalty-free and NOT open source compatible.  Similarly, I would argue that
a good percentage of the RF licensing covenants in IETF IPR disclosures (in
contrast to most non-assert covenants) are not compatible with the majority
of the open source licenses in practical use.  If you wish to express a
desire for compatibility with open source models, then you may want to
express this early on, in precise terms, and not with ambiguous language
such as "free" and "open".
17: Section 5, "20 years": as I have mentioned a number of times before: a
novel combination of 20year+ old tools may be patentable in its own right.
You may wish to add a sentence in this regard.
18: "infringing": it is in most cases unwise to use the term "infringement"
unless you truly know what you are doing---even in combination with
"unknowingly".  Suggest to replace "infringe" with "incur encumbrance" or
something like this.
19: Section 6: I would suggest to remove this section in its entirety.  Some
information presented may be seen differently in the cited SDOs.  For
example, arguably, the low delay AAC effort is intended towards interactive
work, and not storage.  Also, there may be SDOs not mentioned here which
have (or have had) viable speech/audio coding efforts that may feel insulted
by not being mentioned (3GPP2 come to mind, and some others).  As the very
minimum, I would strongly suggest to "characterize" the other SDOs efforts
using only their own (liaison-)statements or publications as a basis, and
not your interpretation.






On 9/4/09 2:40 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> FYI.
> 
> /psa
> 
> - -------- Original Message --------
> Subject: I-D Action:draft-valin-codec-guidelines-00.txt
> Date: Fri,  4 Sep 2009 11:00:01 -0700 (PDT)
> From: Internet-Drafts@ietf.org
> Reply-To: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> Title           : Guidelines for proposed CODEC working group
> Author(s)       : J. Valin, et al.
> Filename        : draft-valin-codec-guidelines-00.txt
> Pages           : 17
> Date            : 2009-09-04
> 
> This document provides general guidelines for specifying audio codecs
> within the IETF.  These guidelines cover the development process, as
> well as the evaluation and intellectual property issues.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-valin-codec-guidelines-00.txt
> 
> 
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From alexander.chemeris@gmail.com  Sat Sep  5 01:21:14 2009
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Hi,

My few notes about the draft. I'll split this into two mails - this one
is about technical considerations:

1) I wonder why support for double- or multi-talk is not mentioned.
It's usually claimed that CELP codecs have hard time encoding conferences,
where there are a lot of double- or multi-talk parts. I believe it should b=
e
stated as MUST or at least SHOULD for conferencing scenario and
included in testing set.

2) I also think we should add an ability to retieve not only signal power,
but a more complex VAD decision from the decoder, even without decoding
a frame. I believe it sohuld be stressed more, that this is a very desired
feature for conferencing and even more for gaming (where you can have
1000s of player and you can't go with decoding every stream). This would
allow codec's use in so called tandem-free bridges, where streamsare not
decoded at all.

3) It is also usually stated that VBR is bad for speech transport over
the Internet, because it leads to higher loss rates then when CBR is used.
So I would make more stress on making CBR works better, then requiring
VBR. (here I refer to source-controlled VBR, which leads to fast bitrate
changes).

On Fri, Sep 4, 2009 at 23:06, Jean-Marc
Valin<jean-marc.valin@octasic.com> wrote:
> Hi,
>
> As a follow-up to the codec BoF in Stockholm, a group of interested IETF
> participants has been working on how to move forward regarding this work =
at
> the IETF. =C2=A0As a result, today we have submitted two Internet-Drafts:
>
> (1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
> (2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt
>
> We welcome feedback on these drafts on the codec@ietf.org list.
>
> Cheers,
>
> =C2=A0 =C2=A0 =C2=A0 =C2=A0Jean-Marc
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>



--=20
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000

From alexander.chemeris@gmail.com  Sat Sep  5 01:22:19 2009
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Hi,

Here are the second part of my notes on drafts, non-technical part:

1) It should be clearly defined, that codec evaluation MUST be done for
several languages, taking into account diferent types of them - e.g. tonal
languages, caucasian languages, etc. If we aim at creating a world-wide
codec, it should work good for more then English and few european
languages.

2) Section 2, bullet 4: "Once a sufficient number of proposals has been
received...". It should be clearly defined when it this "sufficient" number
is reached. I actually don't like the word "sufficient", because it's not
clean, sufficient for what is meant here.

3) Some time frames must be defined. At least it should be clearly
stated that the effort aims at providing working codec in 1yr, 5yrs,
100yrs, etc. Taking into account the size of the goal, it might take
indefinite time to achieve and should be limited.

4) In Section 3.1:
   A more subtle aspect is the information leak that can occur when the
   codec is used over an encrypted channel (e.g.  [SRTP]).  For example,
   it was suggested [wright08] that use of source-controlled VBR may
   reveal some information about a conversation through the size of the
   compressed packets.  This would have to be investigated when
   standardizing a codec.
At least it sohuld be specified how it's planned to be investigated? What
is should be the result of such investigation? Actually I don't see why
it should be done in the scope of this effort.


On Fri, Sep 4, 2009 at 23:06, Jean-Marc
Valin<jean-marc.valin@octasic.com> wrote:
> Hi,
>
> As a follow-up to the codec BoF in Stockholm, a group of interested IETF
> participants has been working on how to move forward regarding this work =
at
> the IETF. =C2=A0As a result, today we have submitted two Internet-Drafts:
>
> (1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
> (2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt
>
> We welcome feedback on these drafts on the codec@ietf.org list.
>
> Cheers,
>
> =C2=A0 =C2=A0 =C2=A0 =C2=A0Jean-Marc
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>



--=20
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000

From alexander.chemeris@gmail.com  Sat Sep  5 01:44:48 2009
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On Sat, Sep 5, 2009 at 01:11, Benjamin M.
Schwartz<bmschwar@fas.harvard.edu> wrote:
> Audio data arrives at some apparent rate, which we can measure in samples
> per second on the local system sample clock. =C2=A0If the link latency an=
d
> buffering are constant, and the two endpoints' sample clocks tick at the
> same rate, then the audio data will arrive at the "right" speed, and can
> simply be decoded and thrown into the sound card's output buffer.
>
> If the buffer has to expand or contract over some period of time, or if
> one of the endpoints ticks faster than the other, then fewer or more
> samples will be received than the sound card requires to be played. =C2=
=A0The
> codec must provide some mechanism to resolve this discrepancy. =C2=A0Comm=
only
> suggested mechanisms include resampling (which bends pitch), smart
> time-stretching to preserve pitch, and "adjust during silence".
>
> To me, the important question here is: what sort of interface does the
> codec need to expose for time dilation? For example, it might be a
> function that says "turn the next N coded samples into M decoded samples,
> please".

I also greatly support inclusion of this feature to "SHOULD" as it is alrea=
dy
done with PLC. This will save a lot of CPU power and considerable increase
voice quality in harsh network conditions.

As it was already said, there are two kinds of time adjustment - "slow" and
"fast". Slow one reflect the need to adjust playback rate to compensate
clock skew between end points. Fast adjustment is usually done when you
need to increase/decrease jitter buffer length to adopt to network conditio=
ns.
Slow adjustments may be made with resampling, but this is not suitable for
conferencing and gaming cases, so usually slow adjustments are actually
done as a set of repeating small fast adjustments. So even in case of LAN
you will have to do time adjustments. And it's best made inside of a codec.

re: API
I've put a lot of time thinking about this when implemented sipX media engi=
ne,
and I same to conclusion that the best way to express desire of time stretc=
h
is with three values:
1) Signed number of samples you need to add/remove (e.g. +100 - I need to
add 100 samples to the stream to increase JB length; -1200 - I need to remo=
ve
1200 samples from the stream to considerably decrease JB length)
2) Some number of enumeration, showing the urgency (e.g. 0 - relax, you can
do this during silent period, 1 - hurry up, stretch actual speech, 2 - urge=
nt!
if you do not stretch, we'll have to drop frames).
3) Next frame, if available. This allows to perform interpolation instead o=
f
extrapolation, improving quality. Though if stretching is done inside the c=
odec,
this probably is not needed.

--=20
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000

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On Sep 4, 2009, at 9:41 PM, Stephan Wenger wrote:

> Folks,
> I read through this draft and found it, all in all, a well written  
> document
> covering a lot of (to me) agreeable aspects.  However, as you may  
> expect, I
> have a number of comments, mostly on certain language aspects :-)   
> If this
> were a working group effort, I would phrase some of my suggestion  
> below in
> stronger terms.  As things stand, you are, of course, free to ignore  
> them...
> Regards,
> Stephan
>
>
> 1. Section 1 speaks in several places of "freely available".  I would
> recommend to be more precise.  There are just too many definitions,  
> some of
> which contradicting, of what constitutes "freedom" in our business.
> 2. Section 1: the above remark also goes for the definition of  
> "open".  A
> very large part of the standardization people (perhaps a majority)
> associates "open" exclusively with "open process in the SDO" and not  
> with
> what the draft implicitly assumes ("open" use, whatever that, in  
> turn, may
> be).
> 3. Section 1, first bullet, "has inhibited": really?  I don't  
> believe this
> claim is true---certainly not in the form as made.  At least, I have  
> not
> seen a prove for this rather assertive statement.  Suggest softening  
> the
> language.
> 4. Section 1, final bullet: suggest replacing "will be freely  
> available"
> with "may be available without royalties or other undue burden", as  
> none of
> us has the means to guaranty something like "freely available".
>
> 5. Section 2.2, "preferably in the form of I-Ds".  I'm not aware of  
> any
> other form of input but I-Ds on which a WG or a group of people  
> without WG
> status in the IETF could act on (and that would allow documenting the
> complexity undoubtly involved).  Therefore, I would suggest to remove
> "preferably".  Also, publishing an I-D undoubtly triggers the BCP79
> disclosure requirements---unspecified input may not.

There is no way to publish an IETF stream RFC without beginning as an  
Internet-draft and
without the contributors agreeing to the IETF IPR rules. This is true  
of AD sponsored drafts
as well as WG sponsored drafts. The disclosure rules definitely apply  
to all IETF contributions.

> 6. Section 2.3: I read this section as a license to re-tune the  
> requirements
> document based on results obtained by tests (or even theoretical  
> analysis)
> of input proposals.  It may be necessary to do this.  However, you are
> certainly aware of that such a procedure can lead to gaming and  
> other evil
> things (including unfair exclusion of proposals with IPR properties  
> that are
> within the scope of the IETF's IPR policy).  If this were a working  
> group
> charter, I would voice strong concerns over this vagueness.  For an  
> informal
> activity outside a WG, I only appeal to you to be careful in what  
> you are
> doing.
> 7. Section 2.6: to add something positive, I'm thrilled that people  
> will at
> least attempt to work cooperatively on codecs from very early on:  
> this is
> IMHO one of the really weak spots traditional codec standardization  
> has had
> in most of its instances.  I'm also very happy to see the "no
> rubberstamping" sentence.
> 8. Section 2.7: developments of RTP payload formats happen in AVT.   
> No need
> to word around that.
> 9. Section 3: I'm missing a statement on what happens if it turns  
> out that
> the requirements are NOT met.  I think there should be an end  
> criteria of
> some sort.
> 10: I can see value in a strategy of not requiring a bit-exact  
> encoder (as I
> see a value in having such a requirement---in classic telco  
> applications,
> for example).  However, I fail to see a value in not requiring bit- 
> exact
> decoding.  Is this a fixed/floating point issue?
> The can of worms one the IPR side one is opening by not strictly  
> defining
> conformance cannot be overestimated.  Almost every IPR statement in  
> the IETF
> (and elsewhere) promises or grants licenses (or makes a non-assert  
> covenant)
> only for conformant implementations.  I would strongly recommend to  
> require
> bit-exact decoding, for that reason alone.  If you truly think that  
> a soft
> definition of conformance is necessary from a technical viewpoint,  
> then I
> would strongly recommend to at least define your version of  
> conformance in a
> normative, permanent document (standard's track RFC) which includes  
> the
> source code of the testing tool, and the test vectors against which  
> soft
> conformance is tested.
> 11: Section 5, first bullet, "open", see remark #2
> 12: Section 5, first bullet, "conspicuous gap", see remark #3
> 13: Section 5, second bullet "freely", see remark #1
> 14: Section 5, fourth bullet, "free" , see remark #1
> 15: Section 5, "BSD license": it's not a BSD license.  It's  
> *currently* a
> BSD-style license.  However, the IETF trust currently has control  
> over this
> thing and could, in theory, change it.

That is correct. Of course, licenses are granted to published text. I  
am not a lawyer, but my understanding is that these licenses, once  
granted, could not be made more restrictive retroactively.

>  Recommend to read the relevant
> documents and come up with a precise and future-proof formulation.   
> At this
> time of writing I do not have access to the documents in question  
> (writing
> on a plane), but I can help in this.

BCP 78 and 79 apply to IETF contributions and IETF publications. Note  
that IETF licenses
explicitly and only refer to copyright, not patent - RFC5378 :

5.5. No Patent License The licenses granted in Section 5.3 shall not  
be deemed to grant any right under any patent, patent application, or  
other similar intellectual property right disclosed by the Contributor  
under BCP 79 [RFC3979] or otherwise.
---
So, in the statement :
In accordance with BCP 78 [RFC5378], the source code for the reference  
implementation should be available under the BSD license.
I would take out "In accordance with BCP 78 [RFC5378],". If the  
reference implementation is published in an RFC, then IETF IPR rules  
will apply. If it is not, they will not.
Regards
Marshall


> 16: Section 5, "royalty-free".  Finally, this reasonably precise  
> term is
> used.  Why not earlier?  However, just before that, the subject of  
> open
> source compatibility is mentioned as well (though not in this  
> formulation).
> Please note that it is very easy to draft a patent license that is
> royalty-free and NOT open source compatible.  Similarly, I would  
> argue that
> a good percentage of the RF licensing covenants in IETF IPR  
> disclosures (in
> contrast to most non-assert covenants) are not compatible with the  
> majority
> of the open source licenses in practical use.  If you wish to  
> express a
> desire for compatibility with open source models, then you may want to
> express this early on, in precise terms, and not with ambiguous  
> language
> such as "free" and "open".
> 17: Section 5, "20 years": as I have mentioned a number of times  
> before: a
> novel combination of 20year+ old tools may be patentable in its own  
> right.
> You may wish to add a sentence in this regard.
> 18: "infringing": it is in most cases unwise to use the term  
> "infringement"
> unless you truly know what you are doing---even in combination with
> "unknowingly".  Suggest to replace "infringe" with "incur  
> encumbrance" or
> something like this.
> 19: Section 6: I would suggest to remove this section in its  
> entirety.  Some
> information presented may be seen differently in the cited SDOs.  For
> example, arguably, the low delay AAC effort is intended towards  
> interactive
> work, and not storage.  Also, there may be SDOs not mentioned here  
> which
> have (or have had) viable speech/audio coding efforts that may feel  
> insulted
> by not being mentioned (3GPP2 come to mind, and some others).  As  
> the very
> minimum, I would strongly suggest to "characterize" the other SDOs  
> efforts
> using only their own (liaison-)statements or publications as a  
> basis, and
> not your interpretation.
>
>
>
>
>
>
> On 9/4/09 2:40 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> FYI.
>>
>> /psa
>>
>> - -------- Original Message --------
>> Subject: I-D Action:draft-valin-codec-guidelines-00.txt
>> Date: Fri,  4 Sep 2009 11:00:01 -0700 (PDT)
>> From: Internet-Drafts@ietf.org
>> Reply-To: internet-drafts@ietf.org
>> To: i-d-announce@ietf.org
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>> Title           : Guidelines for proposed CODEC working group
>> Author(s)       : J. Valin, et al.
>> Filename        : draft-valin-codec-guidelines-00.txt
>> Pages           : 17
>> Date            : 2009-09-04
>>
>> This document provides general guidelines for specifying audio codecs
>> within the IETF.  These guidelines cover the development process, as
>> well as the evaluation and intellectual property issues.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-valin-codec-guidelines-00.txt
>>
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.8 (Darwin)
>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>>
>> iEYEARECAAYFAkqhXxwACgkQNL8k5A2w/vw3KQCbBDmV7SZPXLaiHIDKN4UW1sZO
>> ZssAoJMDkNDNGGOXXscj4PoaB0HFWhtG
>> =Al+W
>> -----END PGP SIGNATURE-----
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>


From jean-marc.valin@usherbrooke.ca  Sat Sep  5 06:42:12 2009
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Alexander Chemeris a écrit :
> 1) I wonder why support for double- or multi-talk is not mentioned.
> It's usually claimed that CELP codecs have hard time encoding conferences,
> where there are a lot of double- or multi-talk parts. I believe it should be
> stated as MUST or at least SHOULD for conferencing scenario and
> included in testing set.

Indeed, this is a very good point that we forgot for the
conferencing-type applications. Will add that.

> 2) I also think we should add an ability to retieve not only signal power,
> but a more complex VAD decision from the decoder, even without decoding
> a frame. I believe it sohuld be stressed more, that this is a very desired
> feature for conferencing and even more for gaming (where you can have
> 1000s of player and you can't go with decoding every stream). This would
> allow codec's use in so called tandem-free bridges, where streamsare not
> decoded at all.

Well, the current text reads like this: "the ability to derive
sufficient parameters, such as loudness and/or spectral envelope, for
estimating voice activity of a compressed frame without fully decoding
that frame". I believe that the spectral envelope contains pretty much
what you need here. Any other info that you think is important?

> 3) It is also usually stated that VBR is bad for speech transport over
> the Internet, because it leads to higher loss rates then when CBR is used.
> So I would make more stress on making CBR works better, then requiring
> VBR. (here I refer to source-controlled VBR, which leads to fast bitrate
> changes).

Do you have any reference regarding VBR and loss rate? In any case, I
think it's clear that both VBR and CBR have valid use cases.

	Jean-Marc

> On Fri, Sep 4, 2009 at 23:06, Jean-Marc
> Valin<jean-marc.valin@octasic.com> wrote:
>> Hi,
>>
>> As a follow-up to the codec BoF in Stockholm, a group of interested IETF
>> participants has been working on how to move forward regarding this work at
>> the IETF.  As a result, today we have submitted two Internet-Drafts:
>>
>> (1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
>> (2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt
>>
>> We welcome feedback on these drafts on the codec@ietf.org list.
>>
>> Cheers,
>>
>>        Jean-Marc
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
> 
> 
> 

From simon.perreault@viagenie.ca  Sat Sep  5 07:15:43 2009
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On Saturday 05 September 2009 04:44:33 Alexander Chemeris wrote:
> As it was already said, there are two kinds of time adjustment - "slow" and
> "fast". Slow one reflect the need to adjust playback rate to compensate
> clock skew between end points. Fast adjustment is usually done when you
> need to increase/decrease jitter buffer length to adopt to network
> conditions. Slow adjustments may be made with resampling, but this is not
> suitable for conferencing and gaming cases, so usually slow adjustments are
> actually done as a set of repeating small fast adjustments. So even in case
> of LAN you will have to do time adjustments. And it's best made inside of a
> codec.

This last sentence puzzles me. Care to explain it?

Thanks,
Simon
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Alexander Chemeris a écrit :
> Here are the second part of my notes on drafts, non-technical part:
> 
> 1) It should be clearly defined, that codec evaluation MUST be done for
> several languages, taking into account diferent types of them - e.g. tonal
> languages, caucasian languages, etc. If we aim at creating a world-wide
> codec, it should work good for more then English and few european
> languages.

I think this actually belongs to the requirements, but otherwise I
totally agree that this is currently missing.

> 2) Section 2, bullet 4: "Once a sufficient number of proposals has been
> received...". It should be clearly defined when it this "sufficient" number
> is reached. I actually don't like the word "sufficient", because it's not
> clean, sufficient for what is meant here.

Will try and come up with something clearer. In any case, it's not that
much about numbers as it is about having all the "building blocks"
required. Any thoughts?

> 3) Some time frames must be defined. At least it should be clearly
> stated that the effort aims at providing working codec in 1yr, 5yrs,
> 100yrs, etc. Taking into account the size of the goal, it might take
> indefinite time to achieve and should be limited.

Someone more knowledgeable of the IETF process can correct me if I'm
wrong here, but my understanding is that the time frame belongs to the
charter "milestones" section.

> 4) In Section 3.1:
>    A more subtle aspect is the information leak that can occur when the
>    codec is used over an encrypted channel (e.g.  [SRTP]).  For example,
>    it was suggested [wright08] that use of source-controlled VBR may
>    reveal some information about a conversation through the size of the
>    compressed packets.  This would have to be investigated when
>    standardizing a codec.
> At least it sohuld be specified how it's planned to be investigated? What
> is should be the result of such investigation? Actually I don't see why
> it should be done in the scope of this effort.

I think this is where cross area review is useful. I assume that in the
worst case, we would end up with using only CBR for SRTP.

Cheers,

	Jean-Marc

> 
> On Fri, Sep 4, 2009 at 23:06, Jean-Marc
> Valin<jean-marc.valin@octasic.com> wrote:
>> Hi,
>>
>> As a follow-up to the codec BoF in Stockholm, a group of interested IETF
>> participants has been working on how to move forward regarding this work at
>> the IETF.  As a result, today we have submitted two Internet-Drafts:
>>
>> (1) http://www.ietf.org/id/draft-valin-codec-guidelines-00.txt
>> (2) http://www.ietf.org/id/draft-valin-codec-requirements-00.txt
>>
>> We welcome feedback on these drafts on the codec@ietf.org list.
>>
>> Cheers,
>>
>>        Jean-Marc
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
> 
> 
> 

From jason.fischl@skypelabs.com  Sat Sep  5 11:15:56 2009
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The minutes have been uploaded now:
http://www.ietf.org/proceedings/75/minutes/codec.txt

Many thanks to Peter for editing them.
Jason

On Fri, Sep 4, 2009 at 12:42 PM, Peter Saint-Andre<stpeter@stpeter.im> wrote:
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Quoting Jean-Marc Valin:

>> 3) It is also usually stated that VBR is bad for speech transport over
>> the Internet, because it leads to higher loss rates then when CBR is used.
>> [...]
>
> Do you have any reference regarding VBR and loss rate? In any case, I
> think it's clear that both VBR and CBR have valid use cases.

I'm only aware of the opposite relationship between VBR and packet  
loss: short peaks at higher rate are easily handled by the packet  
buffer in the bottleneck (e.g. DSL modem). Even temporarily exceeding  
the network bandwidth only results in a small amount of jitter.
On the other hand, constantly exceeding the network bandwidth not only  
guarantees packet loss, but also keeps the packet buffer filled up,  
easily adding hundreds of milliseconds of extra delay to the call.

Other arguments against using CBR:
- At a given quality level, VBR results in fewer total bits (~ half)  
being sent over the network over the duration of a call.
- VBR helps on the receiving end to estimate the available network  
bandwidth, so that the codec can adjust to it (using a feedback  
channel).

VBR seems better suited for the Internet than CBR.

best,
koen.

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Quoting Jean-Marc Valin:
>> 4) In Section 3.1:
>>    A more subtle aspect is the information leak that can occur when the
>>    codec is used over an encrypted channel (e.g.  [SRTP]).  For example,
>>    it was suggested [wright08] that use of source-controlled VBR may
>>    reveal some information about a conversation through the size of the
>>    compressed packets.  This would have to be investigated when
>>    standardizing a codec.
>> At least it sohuld be specified how it's planned to be investigated? What
>> is should be the result of such investigation? Actually I don't see why
>> it should be done in the scope of this effort.
>
> I think this is where cross area review is useful. I assume that in the
> worst case, we would end up with using only CBR for SRTP.

There's already a draft suggesting that:
http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt

The [wright08] reference talks about recognizing whole sentences, from  
a small "dictionary" of 122 sentences, all read in a controlled  
environment, with the same TIMIT database being used for training and  
evaluation.  I'm sure that just the *length* of the sentence reveals  
information about which sentence is being read.  So should we also  
look into hiding the duration of calls?

The [wright08] results do not provide any evidence that "free" speech  
can be recognized by looking at packet sizes.  Speech recognition is  
hard even when the speech signal is available.

koen.

From jean-marc.valin@usherbrooke.ca  Sat Sep  5 18:57:36 2009
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Koen Vos a écrit :
> The [wright08] reference talks about recognizing whole sentences, from a
> small "dictionary" of 122 sentences, all read in a controlled
> environment, with the same TIMIT database being used for training and
> evaluation.  I'm sure that just the *length* of the sentence reveals
> information about which sentence is being read.  So should we also look
> into hiding the duration of calls?

I think [wright08] is interesting in that it raises awareness about a
*potential* issue with VBR. However, I think its main limitation is that
is that it shows a potential information leak in the *Speex* VBR. At
this point, I'm not sure whether the results would generalise to a codec
other than Speex, which decides on the rate based on very simple
heuristics (very unlike SILK). So while I think we need to be careful,
I'm not yet ready to say that encrypted VBR is evil. OTOH, I think pure
CBR has its advantages as well and see no reason we can't have both.

> The [wright08] results do not provide any evidence that "free" speech
> can be recognized by looking at packet sizes.  Speech recognition is
> hard even when the speech signal is available.

I can confirm this. The Speex VBR leaks in the order of 50 bits/second.
That would make a pretty bad recogniser -- even if the VBR code was
specifically optimised for recognition rate.

Cheers,

	Jean-Marc

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Quoting Jean-Marc Valin:

> OTOH, I think pure CBR has its advantages as well and see no reason  
> we can't have both.

Agree that we can have both. But just out of curiosity, what are  
specific advantages of CBR (besides the irrational fear that VBR would  
enable eavesdropping)?

I think Internet codecs should be optimized for low average bitrate  
more than low peak bitrate, because data channels have become shared.  
Similarly, average complexity is more important than peak complexity,  
because there are fewer hard real-time requirements in the system  
(especially on the encoder side).

best,
koen.

From jean-marc.valin@usherbrooke.ca  Sun Sep  6 18:06:49 2009
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Koen Vos a écrit :
> Quoting Jean-Marc Valin:
>> OTOH, I think pure CBR has its advantages as well and see no reason we
>> can't have both.
> 
> Agree that we can have both. But just out of curiosity, what are
> specific advantages of CBR (besides the irrational fear that VBR would
> enable eavesdropping)?

I wouldn't call the fear of encrypted VBR irrational, even though I
believe it is likely blown out of proportion. Actually, to me the main
leak involved in VBR is letting an eavesdroper know when there is active
speech. This is pretty much the same leak as that caused by a VAD, or
even knowing someone is present from the increased video bandwidth.

In any case, the fact that some people want CBR justifies including it
as a requirement. This is not to say that we shouldn't also have VBR, or
even that it shouldn't be optimised for VBR. If we have a very good VBR
codec, I don't think it would be that hard to do CBR with it (even if it
means doing a bit of padding).

> I think Internet codecs should be optimized for low average bitrate more
> than low peak bitrate, because data channels have become shared.

I agree that for most applications, average bit-rate is probably more
important than peak.

> Similarly, average complexity is more important than peak complexity,
> because there are fewer hard real-time requirements in the system
> (especially on the encoder side).

I actually disagree about average complexity. Even though average
complexity matters, peak complexity is what determines whether (e.g.) an
embedded DSP is powerful enough to run the codec in real-time.

Cheers,

	Jean-Marc

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Koen Vos wrote:
> Quoting Jean-Marc Valin:
>=20
>> OTOH, I think pure CBR has its advantages as well and see no reason we=

>> can't have both.
>=20
> Agree that we can have both. But just out of curiosity, what are
> specific advantages of CBR (besides the irrational fear that VBR would
> enable eavesdropping)?

I can think of many advantages to a codec whose peak bandwidth (in a
window of N milliseconds) can easily be limited.  For example, there are
many places where telephone modems are still the most common form of
internet connection.  A good mono voice codec capped at 32kbps could
easily outperform the underlying telephone system for voice quality, in
addition to providing the connectivity advantages of VoIP.

If you can limit peak bandwidth, then CBR is trivial.


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Subject: Re: [codec] Codec guidelines and requirements drafts: complexity
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Quoting Jean-Marc Valin:

> I actually disagree about average complexity. Even though average
> complexity matters, peak complexity is what determines whether (e.g.) an
> embedded DSP is powerful enough to run the codec in real-time.

The hard real-time aspect is part of circuit-switched calls, where  
payloads have to be ready at guaranteed times in order to be sent in  
the allocated circuit-switched time slot. For VoIP applications it's  
ok if a packet is sent a few microseconds, or even milliseconds,  
behind schedule. More importantly, in the vast majority of cases, the  
codec will run on a processor with soft real-time constraints, where  
the processing power is shared with other applications.

Minimizing average complexity:
- Increases battery life on mobile devices
- Enables less powerful devices to run the codec (as long as those  
devices have soft real-time constraints)
- Reduces global energy consumption

For voice communications, the trend is away from fixed allocated  
resources to flexible, shared resources. It's not just about a network  
protocol. My hope is that IETF will be faster to recognize this than  
other SDOs.

koen.

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting Jean-Marc Valin:

> I wouldn't call the fear of encrypted VBR irrational, even though I
> believe it is likely blown out of proportion. Actually, to me the main
> leak involved in VBR is letting an eavesdroper know when there is active
> speech. This is pretty much the same leak as that caused by a VAD, or
> even knowing someone is present from the increased video bandwidth.

How would people take advantage of that kind of information? Surely  
the fact there was a call to begin with (and the sending and receiving  
IP addresses) are much more revealing and incriminating? Does SRTP  
provide mechanisms to obfuscate those?

I'll change my mind when there's an application that transcribes  
(encrypted, VBR) Skype calls by sniffing the packets. (Writing such an  
app must be a sure was to get rich and famous.)


> In any case, the fact that some people want CBR justifies including it
> as a requirement.

Some people on this list want steep royalties to be part of the  
codec.. Don't there need to be rational arguments?

The problem is really that eavesdropping (1) was never shown to work  
and (2) has strong arguments against it based on information theory,  
yet is almost impossible to disprove.

FWIW, Skype cares *a lot* about preventing eavesdropping, and will  
continue to use VBR.

koen.

From jean-marc.valin@usherbrooke.ca  Sun Sep  6 21:57:45 2009
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Subject: Re: [codec] Codec guidelines and requirements drafts: complexity
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Hi,

OK, there's actually a few different issues here. For a decoder, I would
argue that the worst-case complexity is the most important because when
you're waiting for a packet that hasn't arrived yet, you want to wait as
long as possible before choosing to consider it lost. For that, you need
to consider the worst-case complexity of the decoder, otherwise you risk
an audio underrun.

On the encoder side I agree that you can tolerate a bit more "complexity
jitter" and I'm certainly in favour of *also* minimising the average
complexity. But even then, you have to be careful about two things:
1) How bad the worst case is. This has to be bounded. If it takes an
encoder on average 10 ms to encode a 20 ms frame, it's obviously not
acceptable to have the worst-case take 200 ms!
2) How correlated the complexity is to the input signal. If the
complexity looks random, then it may be OK. However, if speech frames
take a lot more CPU than silence/noise frames, your DSP still needs to
be able to encode the speech periods in real-time without lagging.

So I'm not at all against reducing the average complexity of the codec.
I'm just saying that we still have to be very careful about the worst
case complexity. Fortunately, I think most algorithms tend to have a
worst case that isn't too far from the average. And of course, if we can
reduce the average without making the worst case much worse, then we
should definitely do it.

Cheers,

	Jean-Marc

Koen Vos a écrit :
> Quoting Jean-Marc Valin:
> The hard real-time aspect is part of circuit-switched calls, where
> payloads have to be ready at guaranteed times in order to be sent in the
> allocated circuit-switched time slot. For VoIP applications it's ok if a
> packet is sent a few microseconds, or even milliseconds, behind
> schedule. More importantly, in the vast majority of cases, the codec
> will run on a processor with soft real-time constraints, where the
> processing power is shared with other applications.
> 
> Minimizing average complexity:
> - Increases battery life on mobile devices
> - Enables less powerful devices to run the codec (as long as those
> devices have soft real-time constraints)
> - Reduces global energy consumption
> 
> For voice communications, the trend is away from fixed allocated
> resources to flexible, shared resources. It's not just about a network
> protocol. My hope is that IETF will be faster to recognize this than
> other SDOs.
> 
> koen.
> 
> 

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Cc: "codec@ietf.org" <codec@ietf.org>, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
Subject: Re: [codec] Codec guidelines and requirements drafts
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Quoting "Benjamin M. Schwartz":

> I can think of many advantages to a codec whose peak bandwidth (in a
> window of N milliseconds) can easily be limited.  For example, there are
> many places where telephone modems are still the most common form of
> internet connection.  A good mono voice codec capped at 32kbps could
> easily outperform the underlying telephone system for voice quality, in
> addition to providing the connectivity advantages of VoIP.
>
> If you can limit peak bandwidth, then CBR is trivial.

I agree.

When a 40 kbps, 20 ms packet is sent over a 32 kbps channel, it will  
take 25 ms to transport that packet. The packet will thus arrive 5 ms  
"late", but won't be lost. And the late arrival may not even be  
noticed because of other jitter from the network.

SILK deals with this by defining a maximum delay that can be built up  
in the network buffer. So if the bitrate is set to 32 kbps, and the  
maximum delay is 10 ms, that means that one packet may be sent at 48  
kbps, or 10 consecutive packets may be sent at 33.6 kbps, but after  
that the bitrate is no longer allowed to exceed 32 kbps.

koen.

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Koen Vos a écrit :
> Quoting Jean-Marc Valin:
> 
>> I wouldn't call the fear of encrypted VBR irrational, even though I
>> believe it is likely blown out of proportion. Actually, to me the main
>> leak involved in VBR is letting an eavesdroper know when there is active
>> speech. This is pretty much the same leak as that caused by a VAD, or
>> even knowing someone is present from the increased video bandwidth.
> 
> How would people take advantage of that kind of information? Surely the
> fact there was a call to begin with (and the sending and receiving IP
> addresses) are much more revealing and incriminating? Does SRTP provide
> mechanisms to obfuscate those?

The only case I can think of is when you have an audio link running all
day long, with most of the time nobody talking. Observing the rate would
tell you when a meeting starts. I don't mean to say that this is really
very bad. I'm just saying it's the only real "threat" I see.

> I'll change my mind when there's an application that transcribes
> (encrypted, VBR) Skype calls by sniffing the packets. (Writing such an
> app must be a sure was to get rich and famous.)

Oh, I totally agree with you on this point. I'm not at all concerned
about ever having a continuous speech recognizer running from VBR packet
sizes -- especially considering that most VBR implementations probably
leak much less information than the Speex one. OTOH, I wouldn't be too
surprised if you could get a yes/no recognizer to work, although I'm not
sure what the use of that would be.

> Some people on this list want steep royalties to be part of the codec..
> Don't there need to be rational arguments?
> 
> The problem is really that eavesdropping (1) was never shown to work and
> (2) has strong arguments against it based on information theory, yet is
> almost impossible to disprove.
> 
> FWIW, Skype cares *a lot* about preventing eavesdropping, and will
> continue to use VBR.

You don't need to convince my that VBR is good and useful. I'm just
trying to understand what would be so bad with *also* supporting CBR.

	Jean-Marc

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From: Ron <ron@debian.org>
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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On Mon, Sep 07, 2009 at 01:10:53AM -0400, Jean-Marc Valin wrote:
> Koen Vos a écrit :
> > Quoting Jean-Marc Valin:
> > 
> >> I wouldn't call the fear of encrypted VBR irrational, even though I
> >> believe it is likely blown out of proportion. Actually, to me the main
> >> leak involved in VBR is letting an eavesdroper know when there is active
> >> speech. This is pretty much the same leak as that caused by a VAD, or
> >> even knowing someone is present from the increased video bandwidth.
> > 
> > How would people take advantage of that kind of information? Surely the
> > fact there was a call to begin with (and the sending and receiving IP
> > addresses) are much more revealing and incriminating? Does SRTP provide
> > mechanisms to obfuscate those?
> 
> The only case I can think of is when you have an audio link running all
> day long, with most of the time nobody talking. Observing the rate would
> tell you when a meeting starts. I don't mean to say that this is really
> very bad. I'm just saying it's the only real "threat" I see.

You can also tell who is talking.  Which combined with other sources
of information leakage may result in some interesting correlations.

If for example you were monitoring a call between me and my stockbroker,
you might make a fairly reasonably guess about whether they were giving
me tips, or I was giving them instructions, based on who was doing most
of the talking.  Combine that with other information a MITM might observe
about my online or other activity, and it may be possible to tip the
balances of probability in the sort of ways that are interesting (in a
probably illegally profitable sort of way) in that domain.
</tinfoil hat>

I'm likewise not utterly paranoid about the conversation itself being
decoded precisely, but information leakage has a way of being useful
when you least want it to.  For the majority of people it may never
be an issue, but if the people who do (have to) care have to use a
different codec, that can be a form of information leakage too...

That may not be a terribly compelling argument, but saying "I can't
really see a way to use this either" doesn't really mean a lot unless
the person saying that is reporting on the consensus of a group of
eminent cryptographers who have made a determined effort to use it :)

Cheers,
Ron



From alexander.chemeris@gmail.com  Mon Sep  7 01:08:59 2009
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From: Alexander Chemeris <Alexander.Chemeris@sipez.com>
Date: Mon, 7 Sep 2009 12:09:01 +0400
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Subject: Re: [codec] Codec guidelines and requirements drafts
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On Sat, Sep 5, 2009 at 18:16, Simon
Perreault<simon.perreault@viagenie.ca> wrote:
> On Saturday 05 September 2009 04:44:33 Alexander Chemeris wrote:
>> As it was already said, there are two kinds of time adjustment - "slow" and
>> "fast". Slow one reflect the need to adjust playback rate to compensate
>> clock skew between end points. Fast adjustment is usually done when you
>> need to increase/decrease jitter buffer length to adopt to network
>> conditions. Slow adjustments may be made with resampling, but this is not
>> suitable for conferencing and gaming cases, so usually slow adjustments are
>> actually done as a set of repeating small fast adjustments. So even in case
>> of LAN you will have to do time adjustments. And it's best made inside of a
>> codec.
>
> This last sentence puzzles me. Care to explain it?

I meant that codec usually have all (or most) of the information,
needed to perform
time stretching - like pitch, envelop, voice activity, etc.


-- 
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000

From stephen.botzko@gmail.com  Mon Sep  7 04:11:46 2009
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Date: Mon, 7 Sep 2009 07:12:07 -0400
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From: stephen botzko <stephen.botzko@gmail.com>
To: Koen Vos <koen.vos@skype.net>
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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--00151747bf2ca6177a0472faeb32
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>>>
How would people take advantage of that kind of information?
>>>

My understanding is that the security concern is that a third party can
determine *when* someone is speaking simply by observing the audio bit
rate.  .

In any event, other VBR codecs (for instance RFC 5404 for G.719) have the
same "vulnerability".  However, that RFC does not mention VBR as a security
issue in the "security considerations" section (or anywhere else), so i
don't think it was a signifiant concern to AVT.

Providing for a CBR mode would allow the codec to pass through some gateways
that require fixed rate transports without transcoding.  For instance,G.719
can be carried on H.320 connections, and passed through SIP/H.320 gateways.
H.320 happens to require fixed rate audio.  In order to allow this scenario
to be supported w/o audio transcoding, RFC 5404 includes signaling to
negotiate CBR.

Of course, any VBR stream can be converted to CBR simply by padding the
bitstream.  So it would be easy to support it if someone needs it. (I am not
proposing that it be added, just observing that it is easy to do).

Regards
Stephen Botzko
Polycom



On Mon, Sep 7, 2009 at 12:48 AM, Koen Vos <koen.vos@skype.net> wrote:

> Quoting Jean-Marc Valin:
>
>  I wouldn't call the fear of encrypted VBR irrational, even though I
>> believe it is likely blown out of proportion. Actually, to me the main
>> leak involved in VBR is letting an eavesdroper know when there is active
>> speech. This is pretty much the same leak as that caused by a VAD, or
>> even knowing someone is present from the increased video bandwidth.
>>
>
> How would people take advantage of that kind of information? Surely the
> fact there was a call to begin with (and the sending and receiving IP
> addresses) are much more revealing and incriminating? Does SRTP provide
> mechanisms to obfuscate those?
>
> I'll change my mind when there's an application that transcribes
> (encrypted, VBR) Skype calls by sniffing the packets. (Writing such an app
> must be a sure was to get rich and famous.)
>
>
>  In any case, the fact that some people want CBR justifies including it
>> as a requirement.
>>
>
> Some people on this list want steep royalties to be part of the codec..
> Don't there need to be rational arguments?
>
> The problem is really that eavesdropping (1) was never shown to work and
> (2) has strong arguments against it based on information theory, yet is
> almost impossible to disprove.
>
> FWIW, Skype cares *a lot* about preventing eavesdropping, and will continue
> to use VBR.
>
> koen.
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--00151747bf2ca6177a0472faeb32
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

&gt;&gt;&gt;<br>How would people take advantage of that kind of information=
?<br>&gt;&gt;&gt;<br><br>My understanding is that the security concern is t=
hat a third party can determine <u>when</u> someone is speaking simply by o=
bserving the audio bit rate.=A0 .=A0 <br>
<br>In any event, other VBR codecs (for instance RFC 5404 for G.719) have t=
he same &quot;vulnerability&quot;.=A0 However, that RFC does not mention VB=
R as a security issue in the &quot;security considerations&quot; section (o=
r anywhere else), so i don&#39;t think it was a signifiant concern to AVT.<=
br>
<br>Providing for a CBR mode would allow the codec to pass through some gat=
eways that require fixed rate transports without transcoding.=A0 For instan=
ce,G.719 can be carried on H.320 connections, and passed through SIP/H.320 =
gateways.=A0 H.320 happens to require fixed rate audio.=A0 In order to allo=
w this scenario to be supported w/o audio transcoding, RFC 5404 includes si=
gnaling to negotiate CBR.=A0 <br>
<br>Of course, any VBR stream can be converted to CBR simply by padding the=
 bitstream.=A0 So it would be easy to support it if someone needs it. (I am=
 not proposing that it be added, just observing that it is easy to do).<br>
<br>Regards<br>Stephen Botzko<br>Polycom<br><br><br><br><div class=3D"gmail=
_quote">On Mon, Sep 7, 2009 at 12:48 AM, Koen Vos <span dir=3D"ltr">&lt;<a =
href=3D"mailto:koen.vos@skype.net">koen.vos@skype.net</a>&gt;</span> wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Quoting Jean-Marc=
 Valin:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I wouldn&#39;t call the fear of encrypted VBR irrational, even though I<br>
believe it is likely blown out of proportion. Actually, to me the main<br>
leak involved in VBR is letting an eavesdroper know when there is active<br=
>
speech. This is pretty much the same leak as that caused by a VAD, or<br>
even knowing someone is present from the increased video bandwidth.<br>
</blockquote>
<br>
How would people take advantage of that kind of information? Surely the fac=
t there was a call to begin with (and the sending and receiving IP addresse=
s) are much more revealing and incriminating? Does SRTP provide mechanisms =
to obfuscate those?<br>

<br>
I&#39;ll change my mind when there&#39;s an application that transcribes (e=
ncrypted, VBR) Skype calls by sniffing the packets. (Writing such an app mu=
st be a sure was to get rich and famous.)<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
In any case, the fact that some people want CBR justifies including it<br>
as a requirement.<br>
</blockquote>
<br>
Some people on this list want steep royalties to be part of the codec.. Don=
&#39;t there need to be rational arguments?<br>
<br>
The problem is really that eavesdropping (1) was never shown to work and (2=
) has strong arguments against it based on information theory, yet is almos=
t impossible to disprove.<br>
<br>
FWIW, Skype cares *a lot* about preventing eavesdropping, and will continue=
 to use VBR.<br>
<br>
koen.<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</blockquote></div><br>

--00151747bf2ca6177a0472faeb32--

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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: 'Ron' <ron@debian.org>, "codec@ietf.org" <codec@ietf.org>
Date: Mon, 7 Sep 2009 14:14:36 -0400
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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I agree.  Assuming something can't be done because one can't think of a way=
 today is *very* dangerous when it comes to security.  The assumption must =
be that if information is leaked, there will be ways to exploit it.  There =
are many cases where the call content is highly structured, such as your ex=
ample of a call to a stockbroker.  Many other types of service or IVR calls=
 tend to be that way.  If an attacker knows at what point during a call a p=
arty likely speaks their credit card or social security number, the potenti=
al search space is greatly reduced.  =20

I thus think it is a MUST requirement that the encoder can be run in a mode=
 where there is no correlation between packet size, packet rate and timing,=
 and the short and long-term audio being encoded. =20

--Felix

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
> Ron
> Sent: Monday, September 07, 2009 04:02
> To: codec@ietf.org
> Subject: Re: [codec] Codec guidelines and requirements drafts:
> eavesdropping
>=20
> On Mon, Sep 07, 2009 at 01:10:53AM -0400, Jean-Marc Valin wrote:
> > Koen Vos a =E9crit :
> > > Quoting Jean-Marc Valin:
> > >
> > >> I wouldn't call the fear of encrypted VBR irrational, even though I
> > >> believe it is likely blown out of proportion. Actually, to me the
> main
> > >> leak involved in VBR is letting an eavesdroper know when there is
> active
> > >> speech. This is pretty much the same leak as that caused by a VAD, o=
r
> > >> even knowing someone is present from the increased video bandwidth.
> > >
> > > How would people take advantage of that kind of information? Surely
> the
> > > fact there was a call to begin with (and the sending and receiving IP
> > > addresses) are much more revealing and incriminating? Does SRTP
> provide
> > > mechanisms to obfuscate those?
> >
> > The only case I can think of is when you have an audio link running all
> > day long, with most of the time nobody talking. Observing the rate woul=
d
> > tell you when a meeting starts. I don't mean to say that this is really
> > very bad. I'm just saying it's the only real "threat" I see.
>=20
> You can also tell who is talking.  Which combined with other sources
> of information leakage may result in some interesting correlations.
>=20
> If for example you were monitoring a call between me and my stockbroker,
> you might make a fairly reasonably guess about whether they were giving
> me tips, or I was giving them instructions, based on who was doing most
> of the talking.  Combine that with other information a MITM might observe
> about my online or other activity, and it may be possible to tip the
> balances of probability in the sort of ways that are interesting (in a
> probably illegally profitable sort of way) in that domain.
> </tinfoil hat>
>=20
> I'm likewise not utterly paranoid about the conversation itself being
> decoded precisely, but information leakage has a way of being useful
> when you least want it to.  For the majority of people it may never
> be an issue, but if the people who do (have to) care have to use a
> different codec, that can be a form of information leakage too...
>=20
> That may not be a terribly compelling argument, but saying "I can't
> really see a way to use this either" doesn't really mean a lot unless
> the person saying that is reporting on the consensus of a group of
> eminent cryptographers who have made a determined effort to use it :)
>=20
> Cheers,
> Ron



From koen.vos@skype.net  Mon Sep  7 18:47:12 2009
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From: Koen Vos <koen.vos@skype.net>
To: Ron <ron@debian.org>
References: <4AA16546.8050700@octasic.com> <3d032e5d0909050122l7d0163ecyd9f2683c27f1e5eb@mail.gmail.com> <4AA26C95.8090504@usherbrooke.ca> <20090905161909.77713cwiggjt4cml@mail.skype.net> <4AA31726.3040904@usherbrooke.ca> <20090906164218.85586qpxwpiv2guy@mail.skype.net> <4AA45CC1.7050500@usherbrooke.ca> <20090906214819.191431jmd444jtlf@mail.skype.net> <4AA495DD.3060606@usherbrooke.ca> <20090907080140.GR4137@audi.shelbyville.oz>
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Cc: codec@ietf.org
Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting Ron:

> If for example you were monitoring a call between me and my stockbroker,
> you might make a fairly reasonably guess about whether they were giving
> me tips, or I was giving them instructions, based on who was doing most
> of the talking.

Why not just look at who initiated the call?

koen.

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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On Mon, Sep 07, 2009 at 06:47:38PM -0700, Koen Vos wrote:
> Quoting Ron:
> 
> >If for example you were monitoring a call between me and my stockbroker,
> >you might make a fairly reasonably guess about whether they were giving
> >me tips, or I was giving them instructions, based on who was doing most
> >of the talking.
> 
> Why not just look at who initiated the call?

Obviously you'd look at that too, but either of us may be returning a
missed call, or initiating the call based on an out of band signal,
or some prearranged schedule.  The originator alone doesn't tell you
which party has important information for the other, but the extra
information you could obtain in this manner from the _content_ of the
call about the direction of information flow might well hint that better.

Anyhow, we shouldn't get bogged down in this, that's a job for crypto
people and I'm sure they could come up with better examples than I
did.  I'm pretty sure they have similar correlation techniques already
deployed.  I don't think anyone disputes there _is_ information leakage
here, so if privacy is a factor in the codec requirements, then there
should be a way to cork that leakage.  We aren't the right people to
be judging its importance.

Whether that is done in the codec itself or in the security wrapper
that transports its packets is an open question to consider.  All I'm
saying is that we shouldn't handwave it away because _we_ have no use
for the information we see being leaked.

Recognising that CBR vs VBR is a tradeoff is good engineering, one may
be better than the other in more of the cases we are considering, but
neither is universally better than the other for all use cases.

 Ron



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To: "Wyss, Felix" <Felix.Wyss@inin.com>
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting "Wyss, Felix":

> If an attacker knows at what point during a call a party likely  
> speaks their credit card or social security number, the potential  
> search space is greatly reduced.

Compelling example. Recognizing spoken digits from packet sizes would  
be an interesting study.

> I thus think it is a MUST requirement that the encoder can be run in  
> a mode where there is no correlation between packet size, packet  
> rate and timing, and the short and long-term audio being encoded.

Calls of an agreeing nature are on average shorter than calls with  
lots of discussion.  And without trying to be sexist, I have a  
suspicion there is a correlation between call duration and gender.   
Should a signaling stack have a mode without termination?

Anyway, I absolutely agree there is an information leak with VBR. I am  
not aware of any evidence that it is practically exploitable, but  
concede it's dangerous to just rule that out. But I also think the  
"market" for a codec without such a leak is small and unproven. If we  
make a MUST out of every niche market requirement, we will have a hard  
time getting things done. And anyone capable of implementing a  
reasonably secure communications tool will know how to pad a VBR codec.

In this case there's no problem as most proposals so far have a CBR  
mode anyway. I do however object to the basis for the requirement.

koen.

From Felix.Wyss@inin.com  Wed Sep  9 06:29:44 2009
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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: 'Koen Vos' <koen.vos@skype.net>
Date: Wed, 9 Sep 2009 09:30:15 -0400
Thread-Topic: [codec] Codec guidelines and requirements drafts: eavesdropping
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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> > I thus think it is a MUST requirement that the encoder can be run in
> > a mode where there is no correlation between packet size, packet
> > rate and timing, and the short and long-term audio being encoded.
>
> Calls of an agreeing nature are on average shorter than calls with
> lots of discussion.  And without trying to be sexist, I have a
> suspicion there is a correlation between call duration and gender.
> Should a signaling stack have a mode without termination?

This is not relevant to the codec, though.  If the parties see the need to =
disguise the duration of the interaction, they can keep the streams up long=
er, for example to some fixed length.


> Anyway, I absolutely agree there is an information leak with VBR. I am
> not aware of any evidence that it is practically exploitable, but
> concede it's dangerous to just rule that out. But I also think the
> "market" for a codec without such a leak is small and unproven. If we
> make a MUST out of every niche market requirement, we will have a hard
> time getting things done. And anyone capable of implementing a
> reasonably secure communications tool will know how to pad a VBR codec.

I believe content privacy is pretty fundamental and not just some "niche ma=
rket requirement".

Isn't the goal of this Internet audio codec to avoid having proprietary and=
 likely incompatible extensions?  Anything not clearly spelled out in the R=
FC is basically fair game and good luck getting devices from different vend=
ors to play well together.  In our experience, it's hard enough to get devi=
ces that implement what's in the RFCs correctly and anything slightly unusu=
al greatly increases the likelihood of incompatibilities and problems.  For=
 example, I'd be reluctant to use RTP padding here for this very reason and=
 because it's up to the sender and not part of the media type negotiation.

--Felix



From stephen.botzko@gmail.com  Wed Sep  9 07:56:35 2009
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To: "Wyss, Felix" <Felix.Wyss@inin.com>
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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--0015174769c89e83940473264bca
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All VBR codecs are potentially vulnerable to traffic analysis.  There is at
least one active draft in AVT on the subject already (
http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt).

IMHO, the best way to handle this right now is to let the AVT discussions
develop.

It might also be wise to limit the use of VBR to adaptation to network
conditions.  Alternatively, the VAD guidelines in the Perkin's draft
(section 3) might also be relevant (and could be adapted to the use of VBR
to minimize average bandwidth use).  Though I think those guidelines need
more scrutiny.

Either way, this is a reason for standardizing both the encoder and the
decoder - as this particular vulnerability is in the encoder.

Stephen Botzko
Polycom

On Wed, Sep 9, 2009 at 9:30 AM, Wyss, Felix <Felix.Wyss@inin.com> wrote:

> > > I thus think it is a MUST requirement that the encoder can be run in
> > > a mode where there is no correlation between packet size, packet
> > > rate and timing, and the short and long-term audio being encoded.
> >
> > Calls of an agreeing nature are on average shorter than calls with
> > lots of discussion.  And without trying to be sexist, I have a
> > suspicion there is a correlation between call duration and gender.
> > Should a signaling stack have a mode without termination?
>
> This is not relevant to the codec, though.  If the parties see the need to
> disguise the duration of the interaction, they can keep the streams up
> longer, for example to some fixed length.
>
>
> > Anyway, I absolutely agree there is an information leak with VBR. I am
> > not aware of any evidence that it is practically exploitable, but
> > concede it's dangerous to just rule that out. But I also think the
> > "market" for a codec without such a leak is small and unproven. If we
> > make a MUST out of every niche market requirement, we will have a hard
> > time getting things done. And anyone capable of implementing a
> > reasonably secure communications tool will know how to pad a VBR codec.
>
> I believe content privacy is pretty fundamental and not just some "niche
> market requirement".
>
> Isn't the goal of this Internet audio codec to avoid having proprietary and
> likely incompatible extensions?  Anything not clearly spelled out in the RFC
> is basically fair game and good luck getting devices from different vendors
> to play well together.  In our experience, it's hard enough to get devices
> that implement what's in the RFCs correctly and anything slightly unusual
> greatly increases the likelihood of incompatibilities and problems.  For
> example, I'd be reluctant to use RTP padding here for this very reason and
> because it's up to the sender and not part of the media type negotiation.
>
> --Felix
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--0015174769c89e83940473264bca
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

All VBR codecs are potentially vulnerable to traffic analysis.=A0 There is =
at least one active draft in AVT on the subject already (<a href=3D"http://=
www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt">http://www.ietf.or=
g/id/draft-perkins-avt-srtp-vbr-audio-01.txt</a>).<br>
<br>IMHO, the best way to handle this right now is to let the AVT discussio=
ns develop.=A0 <br><br>It might also be wise to limit the use of VBR to ada=
ptation to network conditions.=A0 Alternatively, the VAD guidelines in the =
Perkin&#39;s draft (section 3) might also be relevant (and could be adapted=
 to the use of VBR to minimize average bandwidth use).=A0 Though I think th=
ose guidelines need more scrutiny.<br>
<br>Either way, this is a reason for standardizing both the encoder and the=
 decoder - as this particular vulnerability is in the encoder.<br><br>Steph=
en Botzko<br>Polycom<br><br><div class=3D"gmail_quote">On Wed, Sep 9, 2009 =
at 9:30 AM, Wyss, Felix <span dir=3D"ltr">&lt;<a href=3D"mailto:Felix.Wyss@=
inin.com">Felix.Wyss@inin.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class=3D"im"=
>&gt; &gt; I thus think it is a MUST requirement that the encoder can be ru=
n in<br>

&gt; &gt; a mode where there is no correlation between packet size, packet<=
br>
&gt; &gt; rate and timing, and the short and long-term audio being encoded.=
<br>
&gt;<br>
&gt; Calls of an agreeing nature are on average shorter than calls with<br>
&gt; lots of discussion. =A0And without trying to be sexist, I have a<br>
&gt; suspicion there is a correlation between call duration and gender.<br>
&gt; Should a signaling stack have a mode without termination?<br>
<br>
</div>This is not relevant to the codec, though. =A0If the parties see the =
need to disguise the duration of the interaction, they can keep the streams=
 up longer, for example to some fixed length.<br>
<div class=3D"im"><br>
<br>
&gt; Anyway, I absolutely agree there is an information leak with VBR. I am=
<br>
&gt; not aware of any evidence that it is practically exploitable, but<br>
&gt; concede it&#39;s dangerous to just rule that out. But I also think the=
<br>
&gt; &quot;market&quot; for a codec without such a leak is small and unprov=
en. If we<br>
&gt; make a MUST out of every niche market requirement, we will have a hard=
<br>
&gt; time getting things done. And anyone capable of implementing a<br>
&gt; reasonably secure communications tool will know how to pad a VBR codec=
.<br>
<br>
</div>I believe content privacy is pretty fundamental and not just some &qu=
ot;niche market requirement&quot;.<br>
<br>
Isn&#39;t the goal of this Internet audio codec to avoid having proprietary=
 and likely incompatible extensions? =A0Anything not clearly spelled out in=
 the RFC is basically fair game and good luck getting devices from differen=
t vendors to play well together. =A0In our experience, it&#39;s hard enough=
 to get devices that implement what&#39;s in the RFCs correctly and anythin=
g slightly unusual greatly increases the likelihood of incompatibilities an=
d problems. =A0For example, I&#39;d be reluctant to use RTP padding here fo=
r this very reason and because it&#39;s up to the sender and not part of th=
e media type negotiation.<br>

<font color=3D"#888888"><br>
--Felix<br>
</font><div><div></div><div class=3D"h5"><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>

--0015174769c89e83940473264bca--

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Re: "It might also be wise to limit the use of VBR to adaptation to
network conditions."

=20

Huh? I think I might understand if you equated VAD/DTX (of a CBR codec)
or layered codec with VBR ... but in general?

=20

For example, lossless codecs (data compression algorithms) have the
property that their output entropy is a function of the input signal
entropy. In other words, they are VBR by design.

=20

To "pad" (or obfuscate) such codec operation is tantamount to wasting
information for a criteria (lack of information leakage) that might not
be important to the particular session.

=20

I fail to see that this is a MUST requirement. The people looking for
exploitable and classifiable differences in the output (i.e.,
information leakage) will just have to look harder/longer (just ask
NSA).

=20

Michael Ramalho

=20

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of stephen botzko
Sent: Wednesday, September 09, 2009 10:57 AM
To: Wyss, Felix
Cc: codec@ietf.org
Subject: Re: [codec] Codec guidelines and requirements drafts:
eavesdropping

=20

All VBR codecs are potentially vulnerable to traffic analysis.  There is
at least one active draft in AVT on the subject already
(http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt).

IMHO, the best way to handle this right now is to let the AVT
discussions develop. =20

It might also be wise to limit the use of VBR to adaptation to network
conditions.  Alternatively, the VAD guidelines in the Perkin's draft
(section 3) might also be relevant (and could be adapted to the use of
VBR to minimize average bandwidth use).  Though I think those guidelines
need more scrutiny.

Either way, this is a reason for standardizing both the encoder and the
decoder - as this particular vulnerability is in the encoder.

Stephen Botzko
Polycom

On Wed, Sep 9, 2009 at 9:30 AM, Wyss, Felix <Felix.Wyss@inin.com> wrote:

> > I thus think it is a MUST requirement that the encoder can be run in
> > a mode where there is no correlation between packet size, packet
> > rate and timing, and the short and long-term audio being encoded.
>
> Calls of an agreeing nature are on average shorter than calls with
> lots of discussion.  And without trying to be sexist, I have a
> suspicion there is a correlation between call duration and gender.
> Should a signaling stack have a mode without termination?

This is not relevant to the codec, though.  If the parties see the need
to disguise the duration of the interaction, they can keep the streams
up longer, for example to some fixed length.



> Anyway, I absolutely agree there is an information leak with VBR. I am
> not aware of any evidence that it is practically exploitable, but
> concede it's dangerous to just rule that out. But I also think the
> "market" for a codec without such a leak is small and unproven. If we
> make a MUST out of every niche market requirement, we will have a hard
> time getting things done. And anyone capable of implementing a
> reasonably secure communications tool will know how to pad a VBR
codec.

I believe content privacy is pretty fundamental and not just some "niche
market requirement".

Isn't the goal of this Internet audio codec to avoid having proprietary
and likely incompatible extensions?  Anything not clearly spelled out in
the RFC is basically fair game and good luck getting devices from
different vendors to play well together.  In our experience, it's hard
enough to get devices that implement what's in the RFCs correctly and
anything slightly unusual greatly increases the likelihood of
incompatibilities and problems.  For example, I'd be reluctant to use
RTP padding here for this very reason and because it's up to the sender
and not part of the media type negotiation.

--Felix



_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Re: &#8220;</span>It might also be wise to limit the use =
of VBR
to adaptation to network conditions.&#8221;<o:p></o:p></p>

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

<p class=3DMsoNormal>Huh? I think I might understand if you equated =
VAD/DTX (of a
CBR codec) or layered codec with VBR &#8230; but in =
general?<o:p></o:p></p>

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

<p class=3DMsoNormal>For example, lossless codecs (data compression =
algorithms)
have the property that their output entropy is a function of the input =
signal
entropy. In other words, they are VBR by design.<o:p></o:p></p>

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

<p class=3DMsoNormal>To &#8220;pad&#8221; (or obfuscate) such codec =
operation is
tantamount to wasting information for a criteria (lack of information =
leakage) that
might not be important to the particular session.<o:p></o:p></p>

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

<p class=3DMsoNormal>I fail to see that this is a MUST requirement. The =
people
looking for exploitable and classifiable differences in the output =
(i.e.,
information leakage) will just have to look harder/longer (just ask =
NSA).<o:p></o:p></p>

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

<p class=3DMsoNormal>Michael Ramalho<span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b>On Behalf Of =
</b>stephen
botzko<br>
<b>Sent:</b> Wednesday, September 09, 2009 10:57 AM<br>
<b>To:</b> Wyss, Felix<br>
<b>Cc:</b> codec@ietf.org<br>
<b>Subject:</b> Re: [codec] Codec guidelines and requirements drafts:
eavesdropping<o:p></o:p></span></p>

</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>All VBR codecs are =
potentially
vulnerable to traffic analysis.&nbsp; There is at least one active draft =
in AVT
on the subject already (<a
href=3D"http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt">h=
ttp://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt</a>).<br>
<br>
IMHO, the best way to handle this right now is to let the AVT =
discussions
develop.&nbsp; <br>
<br>
It might also be wise to limit the use of VBR to adaptation to network
conditions.&nbsp; Alternatively, the VAD guidelines in the Perkin's =
draft
(section 3) might also be relevant (and could be adapted to the use of =
VBR to
minimize average bandwidth use).&nbsp; Though I think those guidelines =
need
more scrutiny.<br>
<br>
Either way, this is a reason for standardizing both the encoder and the =
decoder
- as this particular vulnerability is in the encoder.<br>
<br>
Stephen Botzko<br>
Polycom<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Wed, Sep 9, 2009 at 9:30 AM, Wyss, Felix &lt;<a
href=3D"mailto:Felix.Wyss@inin.com">Felix.Wyss@inin.com</a>&gt; =
wrote:<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&gt; &gt; I thus =
think it is a
MUST requirement that the encoder can be run in<br>
&gt; &gt; a mode where there is no correlation between packet size, =
packet<br>
&gt; &gt; rate and timing, and the short and long-term audio being =
encoded.<br>
&gt;<br>
&gt; Calls of an agreeing nature are on average shorter than calls =
with<br>
&gt; lots of discussion. &nbsp;And without trying to be sexist, I have =
a<br>
&gt; suspicion there is a correlation between call duration and =
gender.<br>
&gt; Should a signaling stack have a mode without =
termination?<o:p></o:p></p>

</div>

<p class=3DMsoNormal>This is not relevant to the codec, though. &nbsp;If =
the
parties see the need to disguise the duration of the interaction, they =
can keep
the streams up longer, for example to some fixed length.<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
<br>
&gt; Anyway, I absolutely agree there is an information leak with VBR. I =
am<br>
&gt; not aware of any evidence that it is practically exploitable, =
but<br>
&gt; concede it's dangerous to just rule that out. But I also think =
the<br>
&gt; &quot;market&quot; for a codec without such a leak is small and =
unproven.
If we<br>
&gt; make a MUST out of every niche market requirement, we will have a =
hard<br>
&gt; time getting things done. And anyone capable of implementing a<br>
&gt; reasonably secure communications tool will know how to pad a VBR =
codec.<o:p></o:p></p>

</div>

<p class=3DMsoNormal>I believe content privacy is pretty fundamental and =
not just
some &quot;niche market requirement&quot;.<br>
<br>
Isn't the goal of this Internet audio codec to avoid having proprietary =
and
likely incompatible extensions? &nbsp;Anything not clearly spelled out =
in the
RFC is basically fair game and good luck getting devices from different =
vendors
to play well together. &nbsp;In our experience, it's hard enough to get =
devices
that implement what's in the RFCs correctly and anything slightly =
unusual
greatly increases the likelihood of incompatibilities and problems. =
&nbsp;For
example, I'd be reluctant to use RTP padding here for this very reason =
and
because it's up to the sender and not part of the media type =
negotiation.<br>
<span style=3D'color:#888888'><br>
--Felix</span><o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a><o:p></o=
:p></p>

</div>

</div>

</div>

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

</div>

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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>>>
   It might also be wise to limit the use of VBR to adaptation to network
conditions.
>>>

The Perkins draft (section 2) is suggesting that

  To avoid the potential information leaks that might enable traffic
   analysis, VBR audio codecs that alter the size or spacing of their
   output according to the characteristics of the input speech signal
   SHOULD NOT be used with encrypted SRTP sessions.

My comment was not intended to be general (for all time), just that it *mig=
ht
*be wise to limit the use of VBR in the requirements for the *first* codec.
Although this is perhaps needlessly conservative from a bandwidth
perspective, it does prevent traffic analysis, and also ensures that the
codec does not need to have a distinct operating mode for SRTP.

Again, my view is that the AVT should take the lead on this, and I also not=
e
that the Perkins draft is a recent submission.

MUSTs (and SHOULD NOTs) related to security are important, and in my view
any codec work in the IETF should conform to best practices as far as
security is concerned.  It would be particularly embarrassing if the AVT
published a BCP that the IETF codec group did not follow.

Regards,
Stephen Botzko
Polycom

On Wed, Sep 9, 2009 at 11:12 AM, Michael Ramalho (mramalho) <
mramalho@cisco.com> wrote:

>  Re: =93It might also be wise to limit the use of VBR to adaptation to
> network conditions.=94
>
>
>
> Huh? I think I might understand if you equated VAD/DTX (of a CBR codec) o=
r
> layered codec with VBR =85 but in general?
>
>
>
> For example, lossless codecs (data compression algorithms) have the
> property that their output entropy is a function of the input signal
> entropy. In other words, they are VBR by design.
>
>
>
> To =93pad=94 (or obfuscate) such codec operation is tantamount to wasting
> information for a criteria (lack of information leakage) that might not b=
e
> important to the particular session.
>
>
>
> I fail to see that this is a MUST requirement. The people looking for
> exploitable and classifiable differences in the output (i.e., information
> leakage) will just have to look harder/longer (just ask NSA).
>
>
>
> Michael Ramalho
>
>
>
> *From:* codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] *On Behalf
> Of *stephen botzko
> *Sent:* Wednesday, September 09, 2009 10:57 AM
> *To:* Wyss, Felix
> *Cc:* codec@ietf.org
> *Subject:* Re: [codec] Codec guidelines and requirements drafts:
> eavesdropping
>
>
>
> All VBR codecs are potentially vulnerable to traffic analysis.  There is =
at
> least one active draft in AVT on the subject already (
> http://www.ietf.org/id/draft-perkins-avt-srtp-vbr-audio-01.txt).
>
>
> IMHO, the best way to handle this right now is to let the AVT discussions
> develop.
>
> It might also be wise to limit the use of VBR to adaptation to network
> conditions.  Alternatively, the VAD guidelines in the Perkin's draft
> (section 3) might also be relevant (and could be adapted to the use of VB=
R
> to minimize average bandwidth use).  Though I think those guidelines need
> more scrutiny.
>
> Either way, this is a reason for standardizing both the encoder and the
> decoder - as this particular vulnerability is in the encoder.
>
> Stephen Botzko
> Polycom
>
> On Wed, Sep 9, 2009 at 9:30 AM, Wyss, Felix <Felix.Wyss@inin.com> wrote:
>
> > > I thus think it is a MUST requirement that the encoder can be run in
> > > a mode where there is no correlation between packet size, packet
> > > rate and timing, and the short and long-term audio being encoded.
> >
> > Calls of an agreeing nature are on average shorter than calls with
> > lots of discussion.  And without trying to be sexist, I have a
> > suspicion there is a correlation between call duration and gender.
> > Should a signaling stack have a mode without termination?
>
> This is not relevant to the codec, though.  If the parties see the need t=
o
> disguise the duration of the interaction, they can keep the streams up
> longer, for example to some fixed length.
>
>
>
> > Anyway, I absolutely agree there is an information leak with VBR. I am
> > not aware of any evidence that it is practically exploitable, but
> > concede it's dangerous to just rule that out. But I also think the
> > "market" for a codec without such a leak is small and unproven. If we
> > make a MUST out of every niche market requirement, we will have a hard
> > time getting things done. And anyone capable of implementing a
> > reasonably secure communications tool will know how to pad a VBR codec.
>
> I believe content privacy is pretty fundamental and not just some "niche
> market requirement".
>
> Isn't the goal of this Internet audio codec to avoid having proprietary a=
nd
> likely incompatible extensions?  Anything not clearly spelled out in the =
RFC
> is basically fair game and good luck getting devices from different vendo=
rs
> to play well together.  In our experience, it's hard enough to get device=
s
> that implement what's in the RFCs correctly and anything slightly unusual
> greatly increases the likelihood of incompatibilities and problems.  For
> example, I'd be reluctant to use RTP padding here for this very reason an=
d
> because it's up to the sender and not part of the media type negotiation.
>
> --Felix
>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>
>
>

--0023545bdb80c76c6b047326d87f
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

&gt;&gt;&gt;<br><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);"><=
/span>=A0=A0 It might also be wise to limit the use of VBR
to adaptation to network conditions.<br>&gt;&gt;&gt;<br><br>The Perkins dra=
ft (section 2) is suggesting that=A0 <br><pre style=3D"margin-left: 40px;">=
<font style=3D"font-family: arial,helvetica,sans-serif;" size=3D"2">  To av=
oid the potential information leaks that might enable traffic<br>
   analysis, VBR audio codecs that alter the size or spacing of their<br>  =
 output according to the characteristics of the input speech signal<br>   S=
HOULD NOT be used with encrypted SRTP sessions</font>.<br></pre>My comment =
was not intended to be general (for all time), just that it <u>might </u>be=
 wise to limit the use of VBR in the requirements for the <u>first</u> code=
c.=A0 Although this is perhaps needlessly conservative from a bandwidth per=
spective, it does prevent traffic analysis, and also ensures that the codec=
 does not need to have a distinct operating mode for SRTP.<br>
<br>Again, my view is that the AVT should take the lead on this, and I also=
 note that the Perkins draft is a recent submission. <br><br>MUSTs (and SHO=
ULD NOTs) related to security are important, and in my view any codec work =
in the IETF should conform to best practices as far as security is concerne=
d.=A0 It would be particularly embarrassing if the AVT published a BCP that=
 the IETF codec group did not follow.<br>
<br>Regards,<br>Stephen Botzko <br>Polycom<br><br><div class=3D"gmail_quote=
">On Wed, Sep 9, 2009 at 11:12 AM, Michael Ramalho (mramalho) <span dir=3D"=
ltr">&lt;<a href=3D"mailto:mramalho@cisco.com">mramalho@cisco.com</a>&gt;</=
span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">








<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">

<div>

<p><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);">Re: =93</span>=
It might also be wise to limit the use of VBR
to adaptation to network conditions.=94</p>

<p>=A0</p>

<p>Huh? I think I might understand if you equated VAD/DTX (of a
CBR codec) or layered codec with VBR =85 but in general?</p>

<p>=A0</p>

<p>For example, lossless codecs (data compression algorithms)
have the property that their output entropy is a function of the input sign=
al
entropy. In other words, they are VBR by design.</p>

<p>=A0</p>

<p>To =93pad=94 (or obfuscate) such codec operation is
tantamount to wasting information for a criteria (lack of information leaka=
ge) that
might not be important to the particular session.</p>

<p>=A0</p>

<p>I fail to see that this is a MUST requirement. The people
looking for exploitable and classifiable differences in the output (i.e.,
information leakage) will just have to look harder/longer (just ask NSA).</=
p>

<p>=A0</p>

<p>Michael Ramalho<span style=3D"font-size: 11pt; color: rgb(31, 73, 125);"=
></span></p>

<p><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);">=A0</span></p>

<div style=3D"border-style: solid none none; border-color: rgb(181, 196, 22=
3) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium=
; padding: 3pt 0in 0in;">

<p><b><span style=3D"font-size: 10pt;">From:</span></b><span style=3D"font-=
size: 10pt;">
<a href=3D"mailto:codec-bounces@ietf.org" target=3D"_blank">codec-bounces@i=
etf.org</a> [mailto:<a href=3D"mailto:codec-bounces@ietf.org" target=3D"_bl=
ank">codec-bounces@ietf.org</a>] <b>On Behalf Of </b>stephen
botzko<br>
<b>Sent:</b> Wednesday, September 09, 2009 10:57 AM<br>
<b>To:</b> Wyss, Felix<br>
<b>Cc:</b> <a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.o=
rg</a><div class=3D"im"><br>
<b>Subject:</b> Re: [codec] Codec guidelines and requirements drafts:
eavesdropping</div></span></p>

</div>

<p>=A0</p>

<p style=3D"margin-bottom: 12pt;">All VBR codecs are potentially
vulnerable to traffic analysis.=A0 There is at least one active draft in AV=
T
on the subject already (<a href=3D"http://www.ietf.org/id/draft-perkins-avt=
-srtp-vbr-audio-01.txt" target=3D"_blank">http://www.ietf.org/id/draft-perk=
ins-avt-srtp-vbr-audio-01.txt</a>).</p><div><div></div><div class=3D"h5"><b=
r>

<br>
IMHO, the best way to handle this right now is to let the AVT discussions
develop.=A0 <br>
<br>
It might also be wise to limit the use of VBR to adaptation to network
conditions.=A0 Alternatively, the VAD guidelines in the Perkin&#39;s draft
(section 3) might also be relevant (and could be adapted to the use of VBR =
to
minimize average bandwidth use).=A0 Though I think those guidelines need
more scrutiny.<br>
<br>
Either way, this is a reason for standardizing both the encoder and the dec=
oder
- as this particular vulnerability is in the encoder.<br>
<br>
Stephen Botzko<br>
Polycom</div></div><div><div></div><div class=3D"h5">

<div>

<p>On Wed, Sep 9, 2009 at 9:30 AM, Wyss, Felix &lt;<a href=3D"mailto:Felix.=
Wyss@inin.com" target=3D"_blank">Felix.Wyss@inin.com</a>&gt; wrote:</p>

<div>

<p style=3D"margin-bottom: 12pt;">&gt; &gt; I thus think it is a
MUST requirement that the encoder can be run in<br>
&gt; &gt; a mode where there is no correlation between packet size, packet<=
br>
&gt; &gt; rate and timing, and the short and long-term audio being encoded.=
<br>
&gt;<br>
&gt; Calls of an agreeing nature are on average shorter than calls with<br>
&gt; lots of discussion. =A0And without trying to be sexist, I have a<br>
&gt; suspicion there is a correlation between call duration and gender.<br>
&gt; Should a signaling stack have a mode without termination?</p>

</div>

<p>This is not relevant to the codec, though. =A0If the
parties see the need to disguise the duration of the interaction, they can =
keep
the streams up longer, for example to some fixed length.</p>

<div>

<p style=3D"margin-bottom: 12pt;"><br>
<br>
&gt; Anyway, I absolutely agree there is an information leak with VBR. I am=
<br>
&gt; not aware of any evidence that it is practically exploitable, but<br>
&gt; concede it&#39;s dangerous to just rule that out. But I also think the=
<br>
&gt; &quot;market&quot; for a codec without such a leak is small and unprov=
en.
If we<br>
&gt; make a MUST out of every niche market requirement, we will have a hard=
<br>
&gt; time getting things done. And anyone capable of implementing a<br>
&gt; reasonably secure communications tool will know how to pad a VBR codec=
.</p>

</div>

<p>I believe content privacy is pretty fundamental and not just
some &quot;niche market requirement&quot;.<br>
<br>
Isn&#39;t the goal of this Internet audio codec to avoid having proprietary=
 and
likely incompatible extensions? =A0Anything not clearly spelled out in the
RFC is basically fair game and good luck getting devices from different ven=
dors
to play well together. =A0In our experience, it&#39;s hard enough to get de=
vices
that implement what&#39;s in the RFCs correctly and anything slightly unusu=
al
greatly increases the likelihood of incompatibilities and problems. =A0For
example, I&#39;d be reluctant to use RTP padding here for this very reason =
and
because it&#39;s up to the sender and not part of the media type negotiatio=
n.<br>
<span style=3D"color: rgb(136, 136, 136);"><br>
--Felix</span></p>

<div>

<div>

<p><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a></p>

</div>

</div>

</div>

<p>=A0</p>

</div></div></div>

</div>


</blockquote></div><br>

--0023545bdb80c76c6b047326d87f--

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Date: Wed,  9 Sep 2009 11:47:09 -0400
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Cc: codec@ietf.org, "Wyss, Felix" <Felix.Wyss@inin.com>, stephen botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting "Michael Ramalho (mramalho)" <mramalho@cisco.com>:
> For example, lossless codecs (data compression algorithms) have the
> property that their output entropy is a function of the input signal
> entropy. In other words, they are VBR by design.

I think (well, I hope!) we all agree that asking a lossless codec to be CBR
doesn't make any sense.

> To "pad" (or obfuscate) such codec operation is tantamount to wasting
> information for a criteria (lack of information leakage) that might not
> be important to the particular session.
>
> I fail to see that this is a MUST requirement. The people looking for
> exploitable and classifiable differences in the output (i.e.,
> information leakage) will just have to look harder/longer (just ask
> NSA).

I don't think anyone said that VBR should be ruled out. The only thing that was
mentioned was having a CBR option in the codec for the cases where the user
*does* care. Of course, all of this only applies to lossy codecs (I'm still
waiting for a lossless codec that is guaranteed to always reduce the size!).

Cheers,

   Jean-Marc


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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Re:" I thus think it is a MUST requirement that the encoder can be run =
in a mode where there is no correlation between packet size, packet rate =
and timing, and the short and long-term audio being encoded."

What is your definition of "no correlation"?

I can more or less guarantee you that humans can design a system that =
they believe has "no correlations" .... but if you train a classifier =
with enough training data with similar sessions .... that you can with =
high probability ascertain the use (or uses) of the channel.

As we speak, bittorrent (and other applications) are being classified =
(i.e., accurately identified) inside of IPSec tunnels (or TLS sessions) =
as I type this. We may not know what is going on inside those channels =
at any given instant - but we have some information (i.e., some =
information leakage!).

So this whole discussion is a matter of degree. Again, ask NSA or people =
applying decades-old classification techniques (VQ, HMMs, etc.) to =
unstructured data communications.

So while I may agree that *some effort* should/may be made to obfuscate =
what is going on (just to make it a little harder for the bad guys), it =
is asymptotically hard to design a system with "no correlations" .... as =
classification machines can see very small differences in channel =
behavior characteristics. I think voice and video transmissions are =
pretty easy to identify (as their entropy ranges and rates of change are =
well known).

I wouldn't like to see us go down this path only to find that some =
relatively unintelligent undergraduate student uses a MATLAB =
classification function to successfully identify our session with less =
than a few hours of effort.

Michael Ramalho

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Wyss, Felix
Sent: Monday, September 07, 2009 2:15 PM
To: 'Ron'; codec@ietf.org
Subject: Re: [codec] Codec guidelines and requirements drafts: =
eavesdropping

I agree.  Assuming something can't be done because one can't think of a =
way today is *very* dangerous when it comes to security.  The assumption =
must be that if information is leaked, there will be ways to exploit it. =
 There are many cases where the call content is highly structured, such =
as your example of a call to a stockbroker.  Many other types of service =
or IVR calls tend to be that way.  If an attacker knows at what point =
during a call a party likely speaks their credit card or social security =
number, the potential search space is greatly reduced.  =20

I thus think it is a MUST requirement that the encoder can be run in a =
mode where there is no correlation between packet size, packet rate and =
timing, and the short and long-term audio being encoded. =20

--Felix

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of
> Ron
> Sent: Monday, September 07, 2009 04:02
> To: codec@ietf.org
> Subject: Re: [codec] Codec guidelines and requirements drafts:
> eavesdropping
>=20
> On Mon, Sep 07, 2009 at 01:10:53AM -0400, Jean-Marc Valin wrote:
> > Koen Vos a =E9crit :
> > > Quoting Jean-Marc Valin:
> > >
> > >> I wouldn't call the fear of encrypted VBR irrational, even though =
I
> > >> believe it is likely blown out of proportion. Actually, to me the
> main
> > >> leak involved in VBR is letting an eavesdroper know when there is
> active
> > >> speech. This is pretty much the same leak as that caused by a =
VAD, or
> > >> even knowing someone is present from the increased video =
bandwidth.
> > >
> > > How would people take advantage of that kind of information? =
Surely
> the
> > > fact there was a call to begin with (and the sending and receiving =
IP
> > > addresses) are much more revealing and incriminating? Does SRTP
> provide
> > > mechanisms to obfuscate those?
> >
> > The only case I can think of is when you have an audio link running =
all
> > day long, with most of the time nobody talking. Observing the rate =
would
> > tell you when a meeting starts. I don't mean to say that this is =
really
> > very bad. I'm just saying it's the only real "threat" I see.
>=20
> You can also tell who is talking.  Which combined with other sources
> of information leakage may result in some interesting correlations.
>=20
> If for example you were monitoring a call between me and my =
stockbroker,
> you might make a fairly reasonably guess about whether they were =
giving
> me tips, or I was giving them instructions, based on who was doing =
most
> of the talking.  Combine that with other information a MITM might =
observe
> about my online or other activity, and it may be possible to tip the
> balances of probability in the sort of ways that are interesting (in a
> probably illegally profitable sort of way) in that domain.
> </tinfoil hat>
>=20
> I'm likewise not utterly paranoid about the conversation itself being
> decoded precisely, but information leakage has a way of being useful
> when you least want it to.  For the majority of people it may never
> be an issue, but if the people who do (have to) care have to use a
> different codec, that can be a form of information leakage too...
>=20
> That may not be a terribly compelling argument, but saying "I can't
> really see a way to use this either" doesn't really mean a lot unless
> the person saying that is reporting on the consensus of a group of
> eminent cryptographers who have made a determined effort to use it :)
>=20
> Cheers,
> Ron


_______________________________________________
codec mailing list
codec@ietf.org
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From: Gregory Maxwell <gmaxwell@juniper.net>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "Wyss, Felix" <Felix.Wyss@inin.com>, Ron <ron@debian.org>, "codec@ietf.org" <codec@ietf.org>
Date: Wed, 9 Sep 2009 13:57:52 -0700
Thread-Topic: [codec] Codec guidelines and requirements drafts: eavesdropping
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Michael Ramalho (mramalho) [mramalho@cisco.com] wrote:
> What is your definition of "no correlation"?
> I can more or less guarantee you that humans can design a system that the=
y believe has "no correlations" .
> ... but if you train a classifier with enough training data with similar =
sessions .... that you can with high probability ascertain the use (or uses=
) of the channel.
[snip]

No information =3D no correlation.  I'm a little surprised to see this CBR =
issue being debated.

First please take a moment to look at this:

http://www.cs.jhu.edu/~cwright/voip-vbr.pdf

In this paper they perform blind language identification using only VBR pac=
ket size sequences from speex.  While many people don't care if an evesdrop=
per could identify their language from the coded packets, I think we'd all =
be surprised to have this happen (*but it's encrypted!*) and I hope we can =
understand that it might be a real security problem for some people.

Of course, it may be that other codecs don't suffer from this as badly as s=
peex does=97 but we have no way to tell. Like many things in security we ca=
n try and fail, only to have an attacker show up later with a more powerful=
 tool.  The ability to use a constant frame size kills that leak dead.  You=
 are either leaking information about the interior channel or you are not, =
and if you are there is a great big uncertainty about how much of a leak th=
e user can tolerate and how strong the attacker's statistical analysis is.=
=20

True=97 there are other risks: Third parties are able to analyize the times=
 and durations, look at who talks to who.  These leak information too, but =
these leaks are very difficult to cure even at an enormous cost and they ex=
ist for most secure communication systems. They are obvious and intuitive w=
eaknesses which should not catch users of secure systems by surprise.  The =
same is not true of VBR: For VBR the possible leak is surprising and the le=
ak can be completely closed at a low cost.

We do not reject SSL and SSH because it's possible that an attacker has ins=
talled a keylogger or because of the prevalence of malware may sometimes ma=
ke them moot.

The magnitude of the leak presented by VBR, if the paper on speex is anythi=
ng to go by, is significantly greater than the magnitude of weaknesses in s=
ome cryptosystems which were considered absolutely fatal.  If you ask a cry=
pto/security focused person to build a secure system out of a data-dependan=
t VBR codec they are going to pad it to a constant size. It's so inexpensiv=
e to close that weakness (even just using transport padding to a constant s=
ize) that even with the most conservative estimate of the leak amount it is=
 obviously worth doing if your goal is security.

So *given* that some users are going to demand constant length output for s=
ecurity reasons or for frame-filling reasons wouldn't it be a good engineer=
ing criteria for the codec to be able fill constant size buffers exactly an=
d have a chance to make some good use of the space provided rather than was=
te it with useless zeros? Wouldn't it be a win for interoperability to not =
have to contend with a multitude of external padding systems?




From alexander.chemeris@gmail.com  Wed Sep  9 14:52:10 2009
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Subject: Re: [codec] Codec guidelines and requirements drafts
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Hi,

A few afterthoughts to add to my previous notes:

On Sat, Sep 5, 2009 at 12:21, Alexander
Chemeris<Alexander.Chemeris@sipez.com> wrote:
> 1) I wonder why support for double- or multi-talk is not mentioned.
> It's usually claimed that CELP codecs have hard time encoding conferences,
> where there are a lot of double- or multi-talk parts. I believe it should be
> stated as MUST or at least SHOULD for conferencing scenario and
> included in testing set.

Also test set should include double-encoding for the same purpose
of conferencing. It's likely that conference bridge will decode speech,
and then encode it again and this should not introduce big degradation
of quality.

Also re: CBR support.
>From what I know, CBR mode is usually less CPU intensive and less
memory hungry, and thus is much better suited for very small and dumb
chips. So ability to support true CBR (not the one with padding, but the
one with just one mode) is very much wanted. For such devices it's usually
required to have low-complexity codec with *acceptable* qulity, i.e. it
is not required to be perfect. I think we should add this point of low CPU
and memory consumption to the wanted list of CBR features in
requirements I-D.

-- 
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000

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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting Jean-Marc Valin:

> I don't think anyone said that VBR should be ruled out. The only  
> thing that was mentioned was having a CBR option in the codec for  
> the cases where the user *does* care.

The privacy problem can easily be solved at a higher layer than the  
codec, by padding a VBR codec in conjunction with encryption. We can  
definitely help the SRTP guys figure out how to do that (padding with  
zeros would obviously be bad for the encryption). And I also have no  
problem with providing a CBR mode that they can prescribe.

Most users and applications will be fine with VBR however, because the  
information leak from VBR is small compared to other leaks (including  
IP addresses, which are easy to extract, can be used as evidence in  
court, etc). PSTN and GSM phones are far more "leaky" than  
well-encrypted VBR, and few people shun these because of privacy  
concerns.

Above all, using CBR for all modes would come at a high cost in  
bitrate, which would roughly double. That would hurt adoption of the  
codec, as many developers would simply choose a more efficient codec,  
and IWAC risks becoming irrelevant.

koen.

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Date: Wed, 09 Sep 2009 15:24:32 -0700
From: Koen Vos <koen.vos@skype.net>
To: Alexander Chemeris <Alexander.Chemeris@sipez.com>
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] Codec guidelines and requirements drafts
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Quoting Alexander Chemeris:

> Also re: CBR support.
> From what I know, CBR mode is usually less CPU intensive and less
> memory hungry, and thus is much better suited for very small and dumb
> chips.

Haven't heard that before, any references?

I would say the CPU and memory consumption depend mostly on coding  
efficiency (ie, quality/bitrate). If anything, VBR has an advantage by  
not spending as many cycles encoding background noise.

koen.



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To: Koen Vos <koen.vos@skype.net>
References: <4AA16546.8050700@octasic.com>	<4AA31726.3040904@usherbrooke.ca><20090906164218.85586qpxwpiv2guy@mail.skype.net><4AA45CC1.7050500@usherbrooke.ca><20090906214819.191431jmd444jtlf@mail.skype.net><4AA495DD.3060606@usherbrooke.ca><20090907080140.GR4137@audi.shelbyville.oz><B043FD61A001424599674F50FC89C2D754F65B32AC@ININMAIL.i3domain.inin.com><20090907194908.161459bhbyrnn0o4@mail.skype.net><B043FD61A001424599674F50FC89C2D754F65B32C3@ININMAIL.i3domain.inin.com>	<6e9223710909090757k2e3afdf1r7bb1baa33aa42a86@mail.gmail.com>	<AA847E176042A54CBB8BA283835E7BCE0166AAFE@xmb-rtp-219.amer.cisco.com>	<1252511229.4aa7cdfda646d@www.usherbrooke.ca> <20090909151546.17786udb93pizklu@mail.skype.net>
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Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Koen Vos wrote:
> Above all, using CBR for all modes would come at a high cost in bitrate=
,
> which would roughly double. That would hurt adoption of the codec, as
> many developers would simply choose a more efficient codec, and IWAC
> risks becoming irrelevant.

No one is suggesting that CBR be the default.  In fact, the requirements
list VBR specifically as a required codec feature, with the implication
that the great majority of users will operate in a VBR mode.

The only requirement in the draft is that the reference encoder offer a
CBR mode, if the client requests it.  This hardly seems controversial, an=
d
I can't imagine how such an optional encoder feature could hurt adoption
of the codec.  This CBR mode need not be particularly efficient, though
efficiency is always nice.  Even a padding scheme would satisfy the
requirements.

--Ben


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To: bens@alum.mit.edu, "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu>
References: <4AA16546.8050700@octasic.com> <4AA31726.3040904@usherbrooke.ca><20090906164218.85586qpxwpiv2guy@mail.skype.net><4AA45CC1.7050500@usherbrooke.ca><20090906214819.191431jmd444jtlf@mail.skype.net><4AA495DD.3060606@usherbrooke.ca><20090907080140.GR4137@audi.shelbyville.oz><B043FD61A001424599674F50FC89C2D754F65B32AC@ININMAIL.i3domain.inin.com><20090907194908.161459bhbyrnn0o4@mail.skype.net><B043FD61A001424599674F50FC89C2D754F65B32C3@ININMAIL.i3domain.inin.com> <6e9223710909090757k2e3afdf1r7bb1baa33aa42a86@mail.gmail.com> <AA847E176042A54CBB8BA283835E7BCE0166AAFE@xmb-rtp-219.amer.cisco.com> <1252511229.4aa7cdfda646d@www.usherbrooke.ca> <20090909151546.17786udb93pizklu@mail.skype.net> <4AA87F60.8030602@fas.harvard.edu>
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Cc: codec@ietf.org, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca>
Subject: Re: [codec] Codec guidelines and requirements drafts: eavesdropping
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Quoting Benjamin M. Schwartz:

> No one is suggesting that CBR be the default.  In fact, the requirements
> list VBR specifically as a required codec feature, with the implication
> that the great majority of users will operate in a VBR mode.
>
> The only requirement in the draft is that the reference encoder offer a
> CBR mode, if the client requests it.  This hardly seems controversial, and
> I can't imagine how such an optional encoder feature could hurt adoption
> of the codec.  This CBR mode need not be particularly efficient, though
> efficiency is always nice.  Even a padding scheme would satisfy the
> requirements.

You're right, there is no controversy: we'll have both CBR and VBR.

(My unhappiness is really with the SRTP draft suggesting that SRTP  
should not be used with VBR - I'll take it to the AVT list.)

koen.

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Subject: [codec] BOF at next IETF meeting approved
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Just wanted to drop folks a quick note to let them know that after the  
IAB/IESG BOF call last friday, the CODEC BOF at the next IETF meeting  
has been approved. I'll work with folks to help sort out what are the  
key issues we need to resolve at the next BOF.

Thanks, Cullen <RAI AD>


