
From jean-marc.valin@octasic.com  Wed Oct  7 10:14:15 2009
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Subject: [codec] Updated codec guidelines and requirements drafts
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Hi everyone,

We just updated the guidelines and requirements drafts to take into account 
the feedback received so far. The updated drafts are at:

http://tools.ietf.org/html/draft-valin-codec-guidelines-01
http://tools.ietf.org/html/draft-valin-codec-requirements-01

Please sent comments on these drafts to the codec at ietf.org list and let 
us know if we missed anything. Let's try to get these drafts as complete as 
possible before the BoF.

Cheers,

	Jean-Marc

From alexander.chemeris@gmail.com  Wed Oct  7 11:29:34 2009
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Hi Jean-Marc,

Is there any IETF web-based tool for seeing differences between draft
versions?

On Wed, Oct 7, 2009 at 21:15, Jean-Marc Valin
<jean-marc.valin@octasic.com> wrote:
> Hi everyone,
>
> We just updated the guidelines and requirements drafts to take into accou=
nt
> the feedback received so far. The updated drafts are at:
>
> http://tools.ietf.org/html/draft-valin-codec-guidelines-01
> http://tools.ietf.org/html/draft-valin-codec-requirements-01
>
> Please sent comments on these drafts to the codec at ietf.org list and le=
t
> us know if we missed anything. Let's try to get these drafts as complete =
as
> possible before the BoF.
>
> 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 stpeter@stpeter.im  Wed Oct  7 11:36:11 2009
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On 10/7/09 12:30 PM, Alexander Chemeris wrote:

> Is there any IETF web-based tool for seeing differences between draft
> versions?

You can click the "Diff" links at the top of those pages. So for instance:

http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url2=draft-valin-codec-guidelines-01.txt
http://tools.ietf.org/rfcdiff?url2=draft-valin-codec-guidelines-01.txt

http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url2=draft-valin-codec-requirements-01.txt
http://tools.ietf.org/rfcdiff?url2=draft-valin-codec-requirements-01.txt

/psa


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Thanks!

On Wed, Oct 7, 2009 at 22:37, Peter Saint-Andre <stpeter@stpeter.im> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 10/7/09 12:30 PM, Alexander Chemeris wrote:
>
>> Is there any IETF web-based tool for seeing differences between draft
>> versions?
>
> You can click the "Diff" links at the top of those pages. So for instance:
>
> http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url2=draft-valin-codec-guidelines-01.txt
> http://tools.ietf.org/rfcdiff?url2=draft-valin-codec-guidelines-01.txt
>
> http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url2=draft-valin-codec-requirements-01.txt
> http://tools.ietf.org/rfcdiff?url2=draft-valin-codec-requirements-01.txt
>
> /psa
>
>
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>



-- 
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
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My apologies for the delayed reply.

On 9/4/09 7:41 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.  

Thanks. We have tried to incorporate most of your feedback into the
document.

> 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.

For consistency, we have changed all instances of "free" or "freely
available" (etc.) to "easily distributable" throughout the text.

> 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).

We clarified the wording about this a bit.

> 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.

I disagree. Certainly we received quite a bit of feedback before IETF 75
and at the Stockholm meeting about this topic, so I don't think it's
appropriate to remove it.

> 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".

In fact, here we really are talking about free distribution of software,
so we've left it the way it was, with a slight wording clarification.

> 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.

We've clarified this. There is input that is subject to BCP79 other than
I-Ds, such as mailing list posts and presentations at WG sessions.

> 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.

That is most definitely not the intent of Section 2.3. We've clarified this.

> 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.

Great.

> 8. Section 2.7: developments of RTP payload formats happen in AVT.  No need
> to word around that.

Clarified to specify explicitly that this work will happen in the AVT WG.

> 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.

I don't think we need a statement about this -- it is covered quite well
by existing IETF practices such as RFC 2418.

> 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.

We modified the text slightly here to clarify things, but retained the
preference against bit-exact encoders and decoders.

> 11: Section 5, first bullet, "open", see remark #2

Changed to "easily distributable".

> 12: Section 5, first bullet, "conspicuous gap", see remark #3

Unchanged, see above.

> 13: Section 5, second bullet "freely", see remark #1

Unchanged, because this really does refer to giving away software
without charging for it.

> 14: Section 5, fourth bullet, "free" , see remark #1

Changed to "easily distributable".

> 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.

Changed to:

   the BSD license or whatever
   license is defined as acceptable by the IETF Trust when the relevant
   Internet-Drafts are published.

> 16: Section 5, "royalty-free".  Finally, this reasonably precise term is
> used.  Why not earlier?  

Because royalty-free is one method for achieving the goal of easily
distributable. It is not, however, the only method.

> 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".

This might need to be tightened up further.

> 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.

This text was tightened up a bit.

> 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.

Done.

> 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.

The authors think this section is very important, because it clarifies
the intent of the WG with respect to efforts at other SDOs and many
questions have been raised about this topic on the list and at the
Stockholm BoF. Those questions need to be addressed somewhere, and this
I-D seems to be the best place for them, especially to clarify the use
of existing liaison relationships.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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Again, my apologies for the delayed reply.

On 9/5/09 5:05 AM, Marshall Eubanks wrote:
> 
> On Sep 4, 2009, at 9:41 PM, Stephan Wenger wrote:

<snip/>

>> 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.

See the updated text. This was meant to include mailing list posts,
presentations at WG sessions, and the like.

<snip/>

>> 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 :

The topic of the referenced text pertains to things that can be
copyrighted, not things that can be patented.

> 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.

Fixed.

Peter

- --
Peter Saint-Andre
https://stpeter.im/

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Thanks, Peter.

I find most of my comments adequately addressed, and in some other cases we
can agree to disagree without another discussion at this point in time.  The
one thing I want to comment on, hopefully constructively, is this:

>> 16: Section 5, "royalty-free".  Finally, this reasonably precise term is
>> used.  Why not earlier?
> 
> Because royalty-free is one method for achieving the goal of easily
> distributable. It is not, however, the only method.
> 
>> 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".
> 
> This might need to be tightened up further.
> 


Let me give it a try.  Replace the original paragraph

>    In accordance with BCP 78 [TRUST], the source code for the reference
>    implementation should be available under the BSD license or whatever
>    license is defined as acceptable by the IETF Trust when the relevant
>    Internet-Drafts are published.  Also, in accordance with BCP 79
>    [IPR], the codecs should preferably use technologies with no known
>    IPR claims or technologies with an offer of royalty-free (RF)
>    licensing.  Compatibility with the licensing of typical open source
>    applications requires the avoidance of restrictive IPR claims.

With

"
In accordance with BCP 78 [TRUST], the source code for the reference
implementation should be available under the BSD-style license or whatever
license is defined as acceptable by the IETF Trust when the relevant
Internet-Drafts are published.

In accordance with the spirit of BCP 79 [IPR], the codecs should preferably
use technologies with no known IPR claims, technologies available under
patent non-assert covenants, or technologies with an offer of royalty-free
(RF) licensing (in this order of preference).  Among other goals, adhering
to these preferences should help achieving compatibility with the patent
licensing requirements of most open source licenses, thereby allowing for
open source applications of the specified codec(s).
"

Motivation:
1. BCP78 compliance of the reference software has nothing to do with
avoidance of patents of the algorithm, so the two should not be placed in
the same paragraph.
2. I added non-assert covenants.  As I mentioned before, non-assert
covenants are typically better for open source license compatibility than
mere RF licensing covenants, unless those RF licensing covenants are
intentionally worded to be friendly/compatible to open source.  Few of the
RF licensing covenants posted in the IETF's IPR database are open source
compatible at the time of writing (yes, I have reviewed all of them :-)
3. Expressing a preference of unencumbered-by-known-patents -> non-assert ->
RF-licensing-commitment makes sense as it increases the chances of a "legal"
open source implementation.

Regards,
Stephan


P.s.: Please note that I'm sending this in order to help you guys to express
what I believe are your intentions.  I haven't changed my mind on viability
of the exercise proposed, nor on my preference of the choice of venue for
codec development.

On 10/7/09 1:59 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> My apologies for the delayed reply.
> 
> On 9/4/09 7:41 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.
> 
> Thanks. We have tried to incorporate most of your feedback into the
> document.
> 
>> 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.
> 
> For consistency, we have changed all instances of "free" or "freely
> available" (etc.) to "easily distributable" throughout the text.
> 
>> 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).
> 
> We clarified the wording about this a bit.
> 
>> 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.
> 
> I disagree. Certainly we received quite a bit of feedback before IETF 75
> and at the Stockholm meeting about this topic, so I don't think it's
> appropriate to remove it.
> 
>> 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".
> 
> In fact, here we really are talking about free distribution of software,
> so we've left it the way it was, with a slight wording clarification.
> 
>> 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.
> 
> We've clarified this. There is input that is subject to BCP79 other than
> I-Ds, such as mailing list posts and presentations at WG sessions.
> 
>> 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.
> 
> That is most definitely not the intent of Section 2.3. We've clarified this.
> 
>> 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.
> 
> Great.
> 
>> 8. Section 2.7: developments of RTP payload formats happen in AVT.  No need
>> to word around that.
> 
> Clarified to specify explicitly that this work will happen in the AVT WG.
> 
>> 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.
> 
> I don't think we need a statement about this -- it is covered quite well
> by existing IETF practices such as RFC 2418.
> 
>> 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.
> 
> We modified the text slightly here to clarify things, but retained the
> preference against bit-exact encoders and decoders.
> 
>> 11: Section 5, first bullet, "open", see remark #2
> 
> Changed to "easily distributable".
> 
>> 12: Section 5, first bullet, "conspicuous gap", see remark #3
> 
> Unchanged, see above.
> 
>> 13: Section 5, second bullet "freely", see remark #1
> 
> Unchanged, because this really does refer to giving away software
> without charging for it.
> 
>> 14: Section 5, fourth bullet, "free" , see remark #1
> 
> Changed to "easily distributable".
> 
>> 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.
> 
> Changed to:
> 
>    the BSD license or whatever
>    license is defined as acceptable by the IETF Trust when the relevant
>    Internet-Drafts are published.
> 
>> 16: Section 5, "royalty-free".  Finally, this reasonably precise term is
>> used.  Why not earlier?
> 
> Because royalty-free is one method for achieving the goal of easily
> distributable. It is not, however, the only method.
> 
>> 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".
> 
> This might need to be tightened up further.
> 
>> 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.
> 
> This text was tightened up a bit.
> 
>> 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.
> 
> Done.
> 
>> 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.
> 
> The authors think this section is very important, because it clarifies
> the intent of the WG with respect to efforts at other SDOs and many
> questions have been raised about this topic on the list and at the
> Stockholm BoF. Those questions need to be addressed somewhere, and this
> I-D seems to be the best place for them, especially to clarify the use
> of existing liaison relationships.
> 
> Peter
> 
> - --
> Peter Saint-Andre
> https://stpeter.im/
> 
> 
> -----BEGIN PGP SIGNATURE-----
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> ZgcAoPwT22sH7KGcPsBmAhF+tzQks6No
> =E6pm
> -----END PGP SIGNATURE-----



From jean-marc.valin@usherbrooke.ca  Wed Oct  7 20:32:55 2009
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Hi David,

> Having implemented many codecs, I really like the idea of 16 bit, fixed
> point, bit exact reference implementation.  Bit exact implementations
> just make life so much easier for implementation.  

I agree that being bit-exact with the reference implementation is
probably the easiest way to port the codec and the intent wasn't to
prevent anyone from doing that.

> I can't think of any
> architectures where 16 bit/fixed point won't implement cleanly.  In many
> cases you would just need to cross-compile and you are done with a
> port. 

Probably the best example of architectures where you wouldn't want to be
bit-exact is when you want floating point code (which is faster than
fixed-point on many machines). Another example is for fractional 32x16
multiplications, for which the ITU-T has funny operators (based on older
DSPs that didn't have a signed-unsigned multiplier). I'm sure there are
other examples where you could gain performance by not being "bit-exact"
with the reference implementation.

> In other words I can't see any risk in a "bias" towards a 16 bit fixed
> point architecture.  I can however see a lot of risk allowing a "close
> enough" implementation.  Making the codec easy to implement will mean
> faster acceptance.  Get two vendors arguing over whose implementation is
> correct and the opposite will be true.
> 
> One option would be to specify the algorithm is a bit exact form, then
> allow implementers to do a "close enough" implementation if they wish.

This is exactly the intent of the drafts. In saying that close enough is
OK, it's implied that "exact" is also OK. We can probably make that
clearer. Please feel free to suggest some changes here.

> 2/ In Section 6 what does this mean
> 
>      *  Purely wireless applications will not be considered
> 
> Does it exclude Wifi?  Maybe an example would clarify.....

Wifi is definitely within the scope here. I agree that "purely wireless"
is not a great description. My personal view (we haven't discussed this
in great details) is that we would rule out applications such as ham
radio and cell phones (unless you're doing VoIP on that mobile), but
include "IP over wireless links". Again, a better wording would be
welcome if anyone can suggest one.

> 3/ Section 2.1:
> 
>         Also, the codec should be robust to noise (produce intelligible
>         speech) even at lower bit-rates.
>        
> I suggest tightening this a bit to something similar to the requirement
> for music "the codec should be robust to noise (produce intelligible
> speech and no annoying artifacts) even at lower bit-rates."  

You mean you would like to add "and no annoying artifacts" to the text?

> We will
> also need a way to measure this, e.g. samples corrupted by various noise
> sources.  Samples with periodic background noise (like a vehicle engine)
> can be challenging.

Agreed.

> 4/ Section 2.2:
> 
> Some great ideas in this section, but I suggest changing:
>         To make it easier to determine which streams have to be mixed
>         (and which are noise/silence), it must be possible to measure
>         (or estimate) the level of the audio in a packet
> to:
>         To make it easier to determine which streams have to be mixed (and
>         which are noise/silence), it must be possible to measure (or estimate)
>         voice activity
> 
> By voice activity I mean a bit or test that says "this packet is not just
> background noise".

Agreed that we can be more precise here. VAD without decode was really
the intent, though I'm thinking that the information could be useful for
other purposes as well.

> 5/ Section 2.2:
> 
> I challenge the need for special low-delay requirements, e.g. 10ms.  My
> ping time to the US west coast is 300ms, so any Internet codec delays
> will be trivial compared to the propagation time of the packet.  I guess
> it could make a difference if you are having a teleconference between
> two east-coast US cities, but the Internet is a big place......

The Internet is indeed a big place and the dalays can vary a lot. That
being said, there are many cases where you are communicating either in
the same city, or to a nearby city. I have standard cable connection and
my ping time (that I just measured non-scientifically) is around 10 ms
within the city and 12 ms to Sherbrooke (200 km away). In thoses cases,
a low-delay codec *can* make a difference on the perceptibility of
acoustic echo. That being said, the intent is to make sure not to
penalise the people who do not need this (e.g. making calls from AU to US).

> 6/ Section 2.6, these are interesting apps but how much special effort
> (e.g. a very tight delay requirement) do we want to put in for such a
> rare use of the codec?  I guess if we get the low delay "for free" it's
> OK.  Or am I missing something and live music etc are likely to be a
> major future use of the codec?

Well, the intend is to get this mostly for free. This could either be
through defining a low-delay mode for the codec (if possible), or by
creating a separate codec. All these requirements do not have to be
fulfilled by a single codec (though of course we don't want one codec
per application).

> Most (>99%?) of today's music apps are delay insensitive, e.g.
> streaming.

Keep in mind that a codec that we are developing now would (if
successful) likely still be in use in 10 years, so we need to look into
the future.

> Perhaps I am suggesting that requirements be weighted in proportion to
> projected use, for example applications 2.1, 2,2, and 2.5 would probably
> cover 99% of all instances of the codec.

See above. Let me restate that the intent is not to penalise the "high"
delay case at all.

> 7/ Section 3 (e.g. discussion of prediction etc) reads like suggestions
> for implementation rather than a requirements document.  They are good
> suggestions BTW, but a requirement is more like (x % degradation under a
> y % packet error rate, or x ms to resynchronise).  If some one achieves
> that in an unexpected way we don't really care.

Section 3 is mostly meant to explain how the Internet impacts the
requirements of the next section.

> 8/ Section 3.1.  Paragraph 1: The worst case CPU should be specified as
> a number.

This part of the complexity discussion is only about security (i.e.
making sure we don't get exposed to DoS). I'm not sure what's the best
way to quantify this. Obviously, if the worst case is 100x more complex
than the average, it's not good, but where exactly would you put the limit?

> 8/ Section 4.2.  g729 should be one of the reference codecs as it is
> ubiquitous on today's Internet.  This new codec will be it's likely
> successor. We should aim for a codec that is >= g729 and IPR free.

I don't actually think g729 is really what we want to target. We are
mostly looking at a wideband codec, with narrowband being mostly useful
for "legacy compatibility". Even the 8 kbps bit-rate isn't that useful
considering that we're usually sending 16 kbps worth of headers (RTP
with 20 ms packetization time). Also, G.729 is not exactly what I would
describe as an easily distributable audio codec.

Cheers,

	Jean-Marc

From stpeter@stpeter.im  Thu Oct  8 11:31:20 2009
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Subject: [codec] proposed charter
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

In the Codec BoF request that I submitted to agenda@ietf.org, I provided
an updated charter proposal. It seems that I neglected to send that to
the list. I do so now, with a request for feedback so that we can work
toward consensus regarding the problem statement, objectives, scope,
deliverables, and milestones of the proposed Codec WG.

/psa

***

PROPOSED CHARTER

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among  application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.
Because the Internet Standards Process defines an open, transparent
process for technology standardization, a wide range of codec experts
have committed to actively working on such codecs within the IETF.  The
Codec WG provides a structured forum for these efforts.

Objectives

The goal of this group is to develop one or more high-quality,
widely-distributable audio codecs that is optimized for use over the
Internet. Core considerations include the following:

1. Use in interactive applications (examples include but are not limited
to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music performance)

2. Addressing the real transport conditions of the Internet, especially
packet loss

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP)

Codecs optimized for very low bit rates and codecs for non-interactive
audio are out of scope.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor not to function as a "codec factory" that produces a large
number of codecs.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these will be further
refined by the group in accordance with the usual IETF procedures for
arriving at rough consensus.

Intellectual property rights (IPR) issues will be handled in accordance
with BCP 78 and BCP 79. Many existing codecs with the technical
attributes listed above are encumbered with licensing fees and other IPR
claims that make royalty-free and wide distribution and use across the
Internet community difficult.  The group will try to standardize codecs
that can be relatively freely and easily distributed, and employed as
much as possible. In doing so, it will adhere closely to these BCPs.
More specifically, "In general, IETF working groups prefer technologies
with no known IPR claims or, for technologies with claims against them,
an offer of royalty-free licensing."  Note that in accordance with BCP
79, the group will not explicitly rule out the possibility of adopting
technology that does not meet these IPR requirements; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the basis for achieving consensus. This
document shall be Informational.

2. Detailed technical requirements, using draft-valin-codec-requirements
as the basis for achieving consensus.  This document shall be Informational.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation. The text
description of each codec shall indicate which components of the encoder
and decoder are mandatory, recommended, and optional.  It is envisioned
that each such specification shall be a Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
Mar-2010: WGLC on Requirements
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: Requirements to IESG (Infomational)
Oct-2010: WGLC on codec specification(s)
Jan-2011: Submit codec specification(s) to IESG (Standards Track)

***


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From tme@americafree.tv  Thu Oct  8 12:10:18 2009
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From: Marshall Eubanks <tme@americafree.tv>
To: Peter Saint-Andre <stpeter@stpeter.im>
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This charter is not on the general BOF page for IETF 76

http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76

Can it be added there ? Or is it in transit ? (I would like to send  
the link to some people.)

Regards
Marshall


On Oct 8, 2009, at 2:33 PM, Peter Saint-Andre wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> In the Codec BoF request that I submitted to agenda@ietf.org, I  
> provided
> an updated charter proposal. It seems that I neglected to send that to
> the list. I do so now, with a request for feedback so that we can work
> toward consensus regarding the problem statement, objectives, scope,
> deliverables, and milestones of the proposed Codec WG.
>
> /psa
>
> ***
>
> PROPOSED CHARTER
>
> Problem Statement
>
> There are no standardized, high-quality audio codecs that are  
> optimized
> for use in interactive Internet applications and that can be widely
> implemented and easily distributed among  application developers,
> service operators, and end users.  This gap in the Internet protocol
> stack has hindered protocol designers from being able to specify
> mandatory-to-implement codecs in their protocols for the sake of
> interoperability, developers from building innovative, interactive
> applications, service operators from deploying affordable, high- 
> quality
> services, and end users from enjoying an optimal user experience.
> Because the Internet Standards Process defines an open, transparent
> process for technology standardization, a wide range of codec experts
> have committed to actively working on such codecs within the IETF.   
> The
> Codec WG provides a structured forum for these efforts.
>
> Objectives
>
> The goal of this group is to develop one or more high-quality,
> widely-distributable audio codecs that is optimized for use over the
> Internet. Core considerations include the following:
>
> 1. Use in interactive applications (examples include but are not  
> limited
> to point-to-point voice calls, multi-party voice conferencing,
> telepresence, teleoperation, in-game voice chat, and live music  
> performance)
>
> 2. Addressing the real transport conditions of the Internet,  
> especially
> packet loss
>
> 3. Ensuring interoperability with the Real-time Transport Protocol
> (RTP), including secure transport via SRTP
>
> 4. Ensuring interoperability with Internet signalling technologies  
> such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP)
>
> Codecs optimized for very low bit rates and codecs for non-interactive
> audio are out of scope.
>
> Although the codec(s) produced by the group might be used as
> mandatory-to-implement technologies by designers of particular  
> Internet
> protocols, it is explicitly not a goal of the group to produce a codec
> that will be mandated for use across the entire IETF or Internet  
> community.
>
> The group might produce only one codec or it might produce multiple
> codecs; however, to reduce confusion in the marketplace it shall
> endeavor not to function as a "codec factory" that produces a large
> number of codecs.
>
> Work processes and technical requirements for achieving the foregoing
> objectives are outlined in draft-valin-codec-guidelines and
> draft-valin-codec-requirements respectively; these will be further
> refined by the group in accordance with the usual IETF procedures for
> arriving at rough consensus.
>
> Intellectual property rights (IPR) issues will be handled in  
> accordance
> with BCP 78 and BCP 79. Many existing codecs with the technical
> attributes listed above are encumbered with licensing fees and other  
> IPR
> claims that make royalty-free and wide distribution and use across the
> Internet community difficult.  The group will try to standardize  
> codecs
> that can be relatively freely and easily distributed, and employed as
> much as possible. In doing so, it will adhere closely to these BCPs.
> More specifically, "In general, IETF working groups prefer  
> technologies
> with no known IPR claims or, for technologies with claims against  
> them,
> an offer of royalty-free licensing."  Note that in accordance with BCP
> 79, the group will not explicitly rule out the possibility of adopting
> technology that does not meet these IPR requirements; such decisions
> will be made in accordance with the rough consensus of the group.
>
> Deliverables
>
> 1. Guidelines that define the work process of the group, using
> draft-valin-codec-guidelines as the basis for achieving consensus.  
> This
> document shall be Informational.
>
> 2. Detailed technical requirements, using draft-valin-codec- 
> requirements
> as the basis for achieving consensus.  This document shall be  
> Informational.
>
> 3. Specification of one or more codecs that meet the agreed-upon
> requirements, in the form of Internet-Drafts that define the codec
> algorithm and source code for a reference implementation. The text
> description of each codec shall indicate which components of the  
> encoder
> and decoder are mandatory, recommended, and optional.  It is  
> envisioned
> that each such specification shall be a Proposed Standard document.
>
> Milestones
>
> Mar-2010: WGLC on Codec Standardization Guidelines
> Mar-2010: WGLC on Requirements
> May-2010: Codec Standardization Guidelines to IESG (Informational)
> May-2010: Requirements to IESG (Infomational)
> Oct-2010: WGLC on codec specification(s)
> Jan-2011: Submit codec specification(s) to IESG (Standards Track)
>
> ***
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.8 (Darwin)
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Subject: Re: [codec] proposed charter
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On 10/8/09 1:11 PM, Marshall Eubanks wrote:
> This charter is not on the general BOF page for IETF 76
> 
> http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76

Updated, with link to:

http://trac.tools.ietf.org/bof/trac/attachment/wiki/BifIETF76/codec-charter.txt

/psa
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
Date: Thu, 8 Oct 2009 15:03:09 -0700
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The problem statement and objective number 1 are admirably stated.

I would suggest however the codec has no dependency on the signaling protoc=
ol, whatever that may be.
It is up to the signaling protocol to inform about the codec and not vice v=
ersa.
Last but not least, there are other, non-IETF signaling protocols in wide u=
se on the Internet.

Also the reference to packet loss in the Internet opens a can of worms, sin=
ce broadband networks don't really suffer from sizable random packet loss, =
but may suffer from packet burst loss (such as producing voice clipping) du=
e to congestion or route flapping for which no codec can expected to be tol=
erant or immune.

The can of worms is even bigger for wireless access networks and for mobile=
 devices, but again, for whatever cause, voice clipping is beyond the codec=
s to fix.

If people still want to mention packet loss, what are the correct metrics (=
random and burst statistics) and the values for Internet voice codecs to de=
al with? What is the role of the transport protocols and the role of the co=
decs in dealing with packet loss? For example, how is packet loss dealt wit=
h in streaming HTTP
(Example: <draft-pantos-http-live-streaming>), just to illustrate the point=
 about transport, since streaming HTTP may not be suitable for interactive =
voice .

In summary, IMO there should be no codec dependency on specific signaling p=
rotocols and no mention of packet loss.

Henry


On 10/8/09 1:33 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

In the Codec BoF request that I submitted to agenda@ietf.org, I provided
an updated charter proposal. It seems that I neglected to send that to
the list. I do so now, with a request for feedback so that we can work
toward consensus regarding the problem statement, objectives, scope,
deliverables, and milestones of the proposed Codec WG.

/psa

***

PROPOSED CHARTER

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among  application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.
Because the Internet Standards Process defines an open, transparent
process for technology standardization, a wide range of codec experts
have committed to actively working on such codecs within the IETF.  The
Codec WG provides a structured forum for these efforts.

Objectives

The goal of this group is to develop one or more high-quality,
widely-distributable audio codecs that is optimized for use over the
Internet. Core considerations include the following:

1. Use in interactive applications (examples include but are not limited
to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music performance=
)

2. Addressing the real transport conditions of the Internet, especially
packet loss

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP)

Codecs optimized for very low bit rates and codecs for non-interactive
audio are out of scope.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor not to function as a "codec factory" that produces a large
number of codecs.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these will be further
refined by the group in accordance with the usual IETF procedures for
arriving at rough consensus.

Intellectual property rights (IPR) issues will be handled in accordance
with BCP 78 and BCP 79. Many existing codecs with the technical
attributes listed above are encumbered with licensing fees and other IPR
claims that make royalty-free and wide distribution and use across the
Internet community difficult.  The group will try to standardize codecs
that can be relatively freely and easily distributed, and employed as
much as possible. In doing so, it will adhere closely to these BCPs.
More specifically, "In general, IETF working groups prefer technologies
with no known IPR claims or, for technologies with claims against them,
an offer of royalty-free licensing."  Note that in accordance with BCP
79, the group will not explicitly rule out the possibility of adopting
technology that does not meet these IPR requirements; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the basis for achieving consensus. This
document shall be Informational.

2. Detailed technical requirements, using draft-valin-codec-requirements
as the basis for achieving consensus.  This document shall be Informational=
.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation. The text
description of each codec shall indicate which components of the encoder
and decoder are mandatory, recommended, and optional.  It is envisioned
that each such specification shall be a Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
Mar-2010: WGLC on Requirements
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: Requirements to IESG (Infomational)
Oct-2010: WGLC on codec specification(s)
Jan-2011: Submit codec specification(s) to IESG (Standards Track)

***


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


--_000_C6F3CBCD62DBhsinnreiadobecom_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [codec] proposed charter</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>The problem statement and objective number 1 are admirably stated.<BR=
>
<BR>
I would suggest however the codec has no dependency on the signaling protoc=
ol, whatever that may be. <BR>
It is up to the signaling protocol to inform about the codec and not vice v=
ersa.<BR>
Last but not least, there are other, non-IETF signaling protocols in wide u=
se on the Internet.<BR>
<BR>
Also the reference to packet loss in the Internet opens a can of worms, sin=
ce broadband networks don&#8217;t really suffer from sizable random packet =
loss, but may suffer from packet burst loss (such as producing voice clippi=
ng) due to congestion or route flapping for which no codec can expected to =
be tolerant or immune.<BR>
<BR>
The can of worms is even bigger for wireless access networks and for mobile=
 devices, but again, for whatever cause, voice clipping is beyond the codec=
s to fix.<BR>
<BR>
If people still want to mention packet loss, what are the correct metrics (=
random and burst statistics) and the values for Internet voice codecs to de=
al with? What is the role of the transport protocols and the role of the co=
decs in dealing with packet loss? For example, how is packet loss dealt wit=
h in streaming HTTP <BR>
(Example: &lt;draft-pantos-http-live-streaming&gt;), just to illustrate the=
 point about transport, since streaming HTTP may not be suitable for intera=
ctive voice .<BR>
<BR>
In summary, IMO there should be no codec dependency on specific signaling p=
rotocols and no mention of packet loss.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/8/09 1:33 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@st=
peter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>-----BEGIN PGP SIGNED MESSAGE-----<BR>
Hash: SHA1<BR>
<BR>
In the Codec BoF request that I submitted to <a href=3D"agenda@ietf.org">ag=
enda@ietf.org</a>, I provided<BR>
an updated charter proposal. It seems that I neglected to send that to<BR>
the list. I do so now, with a request for feedback so that we can work<BR>
toward consensus regarding the problem statement, objectives, scope,<BR>
deliverables, and milestones of the proposed Codec WG.<BR>
<BR>
/psa<BR>
<BR>
***<BR>
<BR>
PROPOSED CHARTER<BR>
<BR>
Problem Statement<BR>
<BR>
There are no standardized, high-quality audio codecs that are optimized<BR>
for use in interactive Internet applications and that can be widely<BR>
implemented and easily distributed among &nbsp;application developers,<BR>
service operators, and end users. &nbsp;This gap in the Internet protocol<B=
R>
stack has hindered protocol designers from being able to specify<BR>
mandatory-to-implement codecs in their protocols for the sake of<BR>
interoperability, developers from building innovative, interactive<BR>
applications, service operators from deploying affordable, high-quality<BR>
services, and end users from enjoying an optimal user experience.<BR>
Because the Internet Standards Process defines an open, transparent<BR>
process for technology standardization, a wide range of codec experts<BR>
have committed to actively working on such codecs within the IETF. &nbsp;Th=
e<BR>
Codec WG provides a structured forum for these efforts.<BR>
<BR>
Objectives<BR>
<BR>
The goal of this group is to develop one or more high-quality,<BR>
widely-distributable audio codecs that is optimized for use over the<BR>
Internet. Core considerations include the following:<BR>
<BR>
1. Use in interactive applications (examples include but are not limited<BR=
>
to point-to-point voice calls, multi-party voice conferencing,<BR>
telepresence, teleoperation, in-game voice chat, and live music performance=
)<BR>
<BR>
2. Addressing the real transport conditions of the Internet, especially<BR>
packet loss<BR>
<BR>
3. Ensuring interoperability with the Real-time Transport Protocol<BR>
(RTP), including secure transport via SRTP<BR>
<BR>
4. Ensuring interoperability with Internet signalling technologies such<BR>
as Session Initiation Protocol (SIP), Session Description Protocol<BR>
(SDP), and Extensible Messaging and Presence Protocol (XMPP)<BR>
<BR>
Codecs optimized for very low bit rates and codecs for non-interactive<BR>
audio are out of scope.<BR>
<BR>
Although the codec(s) produced by the group might be used as<BR>
mandatory-to-implement technologies by designers of particular Internet<BR>
protocols, it is explicitly not a goal of the group to produce a codec<BR>
that will be mandated for use across the entire IETF or Internet community.=
<BR>
<BR>
The group might produce only one codec or it might produce multiple<BR>
codecs; however, to reduce confusion in the marketplace it shall<BR>
endeavor not to function as a &quot;codec factory&quot; that produces a lar=
ge<BR>
number of codecs.<BR>
<BR>
Work processes and technical requirements for achieving the foregoing<BR>
objectives are outlined in draft-valin-codec-guidelines and<BR>
draft-valin-codec-requirements respectively; these will be further<BR>
refined by the group in accordance with the usual IETF procedures for<BR>
arriving at rough consensus.<BR>
<BR>
Intellectual property rights (IPR) issues will be handled in accordance<BR>
with BCP 78 and BCP 79. Many existing codecs with the technical<BR>
attributes listed above are encumbered with licensing fees and other IPR<BR=
>
claims that make royalty-free and wide distribution and use across the<BR>
Internet community difficult. &nbsp;The group will try to standardize codec=
s<BR>
that can be relatively freely and easily distributed, and employed as<BR>
much as possible. In doing so, it will adhere closely to these BCPs.<BR>
More specifically, &quot;In general, IETF working groups prefer technologie=
s<BR>
with no known IPR claims or, for technologies with claims against them,<BR>
an offer of royalty-free licensing.&quot; &nbsp;Note that in accordance wit=
h BCP<BR>
79, the group will not explicitly rule out the possibility of adopting<BR>
technology that does not meet these IPR requirements; such decisions<BR>
will be made in accordance with the rough consensus of the group.<BR>
<BR>
Deliverables<BR>
<BR>
1. Guidelines that define the work process of the group, using<BR>
draft-valin-codec-guidelines as the basis for achieving consensus. This<BR>
document shall be Informational.<BR>
<BR>
2. Detailed technical requirements, using draft-valin-codec-requirements<BR=
>
as the basis for achieving consensus. &nbsp;This document shall be Informat=
ional.<BR>
<BR>
3. Specification of one or more codecs that meet the agreed-upon<BR>
requirements, in the form of Internet-Drafts that define the codec<BR>
algorithm and source code for a reference implementation. The text<BR>
description of each codec shall indicate which components of the encoder<BR=
>
and decoder are mandatory, recommended, and optional. &nbsp;It is envisione=
d<BR>
that each such specification shall be a Proposed Standard document.<BR>
<BR>
Milestones<BR>
<BR>
Mar-2010: WGLC on Codec Standardization Guidelines<BR>
Mar-2010: WGLC on Requirements<BR>
May-2010: Codec Standardization Guidelines to IESG (Informational)<BR>
May-2010: Requirements to IESG (Infomational)<BR>
Oct-2010: WGLC on codec specification(s)<BR>
Jan-2011: Submit codec specification(s) to IESG (Standards Track)<BR>
<BR>
***<BR>
<BR>
<BR>
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=3DeQ5m<BR>
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_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] proposed charter
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On Thu, 8 Oct 2009, Henry Sinnreich wrote:

> If people still want to mention packet loss, what are the correct 
> metrics (random and burst statistics) and the values for Internet voice 
> codecs to deal with? What is the role of the transport protocols and the 
> role of the codecs in dealing with packet loss? For example, how is 
> packet loss dealt with in streaming HTTP (Example: 
> <draft-pantos-http-live-streaming>), just to illustrate the point about 
> transport, since streaming HTTP may not be suitable for interactive 
> voice .
>
> In summary, IMO there should be no codec dependency on specific 
> signaling protocols and no mention of packet loss.

I have earlier proposed that the full solution should be able to cope with 
10% packet loss and 1.5-2 seconds of jitter. Of course there would be 
impact on quality, and continous loss (a full second of loss) is of course 
impossible to handle.

But the two major types I see that needs to be handled are:

Wireless networks with ARQ such as 802.11, 3GPP and alike. These typically 
can induce 1-2 seconds of jitter, but with low or no packet loss.

Routers/switches that do buffering and which use either FIFO/taildrop or 
some kind of more intelligent buffer management such as (W)RED or alike. 
These might drop a few percent of packets (usually never more as TCP backs 
off a lot at these packet drop rates and thus the links are usually not 
loaded more than what causes a few percent packet loss) plus induce jitter 
as other traffic affects this. Routers typically never buffer packets more 
than 600ms, often much lower, thus jitter is less unless there are 
multiple congested links in a row.

I'm a firm believer that for the whole solution to work, this can't be 
seen as a layered problem in which the layers can ignore each other, but 
to work well, they need to be tightly integrated.

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

From koen.vos@skype.net  Fri Oct  9 01:59:18 2009
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Date: Fri, 09 Oct 2009 02:00:49 -0700
From: Koen Vos <koen.vos@skype.net>
To: Mikael Abrahamsson <swmike@swm.pp.se>
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] proposed charter
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Mikael,

My view on integrating jitter buffer and codec:

The integration is best done through an interface that lets the  
decoder indicate to the jitter buffer what state the audio signal is  
in, and lets the jitter buffer tell the decoder what action to take.  
For example, if the jitter buffer wants to accelerate the decoding of  
packets, it should (try to) wait until the decoder signals a silent  
period, or at least a stationary part in the signal, and then tell the  
decoder to compress the audio signal. The interface needs to pass just  
a few parameters in both directions for a tight integration between  
decoder and jitter buffer.

The jitter buffer should however not be part of the decoder, because:
1. The jitter buffer, just like other parts of the voice stack, is not  
essential for interoperability.
2. Voice stacks that contain multiple codecs typically share a common  
jitter buffer.
3. A jitter buffer is tuned towards the network and platform. As you  
point out, a wireless 3G network has rather different packet loss and  
jitter behavior than a DSL modem.
4. Even if we would define a jitter buffer that works great for any  
network today, it may not work well with tomorrow's networks.

I also don't see a use in defining how much jitter a codec can handle.  
It's easy to handle 10 seconds of jitter by having a fixed 10 second  
jitter buffer, but depending on the network characteristics that may  
or may not be a good solution.

In short: for any VoIP application to work well, all of its layers and  
components need to be tightly integrated. But the role of a codec is  
merely to provide the interfaces to allow that to happen - not to  
define the complete VoIP application.

best,
koen.



Quoting Mikael Abrahamsson:

> On Thu, 8 Oct 2009, Henry Sinnreich wrote:
>
>> If people still want to mention packet loss, what are the correct  
>> metrics (random and burst statistics) and the values for Internet  
>> voice codecs to deal with? What is the role of the transport  
>> protocols and the role of the codecs in dealing with packet loss?  
>> For example, how is packet loss dealt with in streaming HTTP  
>> (Example: <draft-pantos-http-live-streaming>), just to illustrate  
>> the point about transport, since streaming HTTP may not be suitable  
>> for interactive voice .
>>
>> In summary, IMO there should be no codec dependency on specific  
>> signaling protocols and no mention of packet loss.
>
> I have earlier proposed that the full solution should be able to  
> cope with 10% packet loss and 1.5-2 seconds of jitter. Of course  
> there would be impact on quality, and continous loss (a full second  
> of loss) is of course impossible to handle.
>
> But the two major types I see that needs to be handled are:
>
> Wireless networks with ARQ such as 802.11, 3GPP and alike. These  
> typically can induce 1-2 seconds of jitter, but with low or no  
> packet loss.
>
> Routers/switches that do buffering and which use either  
> FIFO/taildrop or some kind of more intelligent buffer management  
> such as (W)RED or alike. These might drop a few percent of packets  
> (usually never more as TCP backs off a lot at these packet drop  
> rates and thus the links are usually not loaded more than what  
> causes a few percent packet loss) plus induce jitter as other  
> traffic affects this. Routers typically never buffer packets more  
> than 600ms, often much lower, thus jitter is less unless there are  
> multiple congested links in a row.
>
> I'm a firm believer that for the whole solution to work, this can't  
> be seen as a layered problem in which the layers can ignore each  
> other, but to work well, they need to be tightly integrated.
>
> -- 
> Mikael Abrahamsson    email: swmike@swm.pp.se
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>



From ron@debian.org  Fri Oct  9 02:47:04 2009
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Hi,

On Fri, Oct 09, 2009 at 02:00:49AM -0700, Koen Vos wrote:
> Mikael,
> 
> My view on integrating jitter buffer and codec:
> 
> The integration is best done through an interface that lets the
> decoder indicate to the jitter buffer what state the audio signal is
> in, and lets the jitter buffer tell the decoder what action to take.
> For example, if the jitter buffer wants to accelerate the decoding
> of packets, it should (try to) wait until the decoder signals a
> silent period, or at least a stationary part in the signal, and then
> tell the decoder to compress the audio signal. The interface needs
> to pass just a few parameters in both directions for a tight
> integration between decoder and jitter buffer.

Why should the jitter buffer presuppose to know when it would be a
good time for the codec to compress?  Shouldn't the jitter buffer
just be giving the codec stats on dataflow that it can parameterise
how it sees best?

I agree both parties need to be talking about what those stats and
parameters need to be though.

Cheers,
Ron



From lars.eggert@nokia.com  Fri Oct  9 04:34:17 2009
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Hi,

On 2009-10-9, at 9:52, Mikael Abrahamsson wrote:
> I have earlier proposed that the full solution should be able to  
> cope with
> 10% packet loss and 1.5-2 seconds of jitter.

I'm not aware of any Internet paths where such conditions would be  
considered normal. As such I question why such cases should be  
supported as a requirement. (Obviously, if a solution happened to work  
under such conditions *while also* having negligible overhead under  
normal conditions, that'd be fine - but I'm simply not sure this can  
be done.)

Lars

From swmike@swm.pp.se  Fri Oct  9 05:21:06 2009
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On Fri, 9 Oct 2009, Lars Eggert wrote:

> I'm not aware of any Internet paths where such conditions would be considered 
> normal.

During weeks of operation after the tsunami, I saw 7% listed as average 
packet loss for certain networks in the far east. Couple this with some 
wireless network and you have prolonged situations of these conditions. Of 
course this is not good or normal, but it's not very uncommon to see these 
adverse conditions.

> As such I question why such cases should be supported as a requirement. 
> (Obviously, if a solution happened to work under such conditions *while 
> also* having negligible overhead under normal conditions, that'd be fine 
> - but I'm simply not sure this can be done.)

It's my strong belief that if this is not something that is aimed for, 
then we'll end up with network requirements shown to be unrealistic in 
real life. 20ms jitter buffer and 0.1% packet loss as some implementations 
require, ie make the application dictate what the network should do 
instead of trying to adapt the application to real world network 
characteristics.

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

From hsinnrei@adobe.com  Fri Oct  9 06:38:04 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Mikael Abrahamsson <swmike@swm.pp.se>, "codec@ietf.org" <codec@ietf.org>
Date: Fri, 9 Oct 2009 06:39:36 -0700
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It is fortunate this point was discussed.

Designing for 7% packet loss seems to be clearly a non-objective for a high=
 performance voice codec.
Over such networks, IM is much more adequate.

My personal two cents,

Henry


On 10/9/09 7:22 AM, "Mikael Abrahamsson" <swmike@swm.pp.se> wrote:

On Fri, 9 Oct 2009, Lars Eggert wrote:

> I'm not aware of any Internet paths where such conditions would be consid=
ered
> normal.

During weeks of operation after the tsunami, I saw 7% listed as average
packet loss for certain networks in the far east. Couple this with some
wireless network and you have prolonged situations of these conditions. Of
course this is not good or normal, but it's not very uncommon to see these
adverse conditions.

> As such I question why such cases should be supported as a requirement.
> (Obviously, if a solution happened to work under such conditions *while
> also* having negligible overhead under normal conditions, that'd be fine
> - but I'm simply not sure this can be done.)

It's my strong belief that if this is not something that is aimed for,
then we'll end up with network requirements shown to be unrealistic in
real life. 20ms jitter buffer and 0.1% packet loss as some implementations
require, ie make the application dictate what the network should do
instead of trying to adapt the application to real world network
characteristics.

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


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<HTML>
<HEAD>
<TITLE>Re: [codec] proposed charter</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>It is fortunate this point was discussed.<BR>
<BR>
Designing for 7% packet loss seems to be clearly a non-objective for a high=
 performance voice codec.<BR>
Over such networks, IM is much more adequate.<BR>
<BR>
My personal two cents,<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/9/09 7:22 AM, &quot;Mikael Abrahamsson&quot; &lt;<a href=3D"swmike@sw=
m.pp.se">swmike@swm.pp.se</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>On Fri, 9 Oct 2009, Lars Eggert wrote:<BR>
<BR>
&gt; I'm not aware of any Internet paths where such conditions would be con=
sidered<BR>
&gt; normal.<BR>
<BR>
During weeks of operation after the tsunami, I saw 7% listed as average<BR>
packet loss for certain networks in the far east. Couple this with some<BR>
wireless network and you have prolonged situations of these conditions. Of<=
BR>
course this is not good or normal, but it's not very uncommon to see these<=
BR>
adverse conditions.<BR>
<BR>
&gt; As such I question why such cases should be supported as a requirement=
.<BR>
&gt; (Obviously, if a solution happened to work under such conditions *whil=
e<BR>
&gt; also* having negligible overhead under normal conditions, that'd be fi=
ne<BR>
&gt; - but I'm simply not sure this can be done.)<BR>
<BR>
It's my strong belief that if this is not something that is aimed for,<BR>
then we'll end up with network requirements shown to be unrealistic in<BR>
real life. 20ms jitter buffer and 0.1% packet loss as some implementations<=
BR>
require, ie make the application dictate what the network should do<BR>
instead of trying to adapt the application to real world network<BR>
characteristics.<BR>
<BR>
--<BR>
Mikael Abrahamsson &nbsp;&nbsp;&nbsp;email: <a href=3D"swmike@swm.pp.se">sw=
mike@swm.pp.se</a><BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
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From swmike@swm.pp.se  Fri Oct  9 07:17:29 2009
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On Fri, 9 Oct 2009, Henry Sinnreich wrote:

> It is fortunate this point was discussed.

Yes.

> Designing for 7% packet loss seems to be clearly a non-objective for a high performance voice codec.

Why? Is there a concrete technical reason why it can't be done? Over these 
kinds of networks you'll see lots of jitter anyhow, is there a concrete 
technical reason why one can't make sound good enough for the other end to 
understand what is being said even at 10% packet loss with 200ms buffert? 
If you send one packet every 20ms, 10% packet loss means you're losing one 
packet every 200ms (on average).

> Over such networks, IM is much more adequate.

I think people want to have the option to speak to each other if they can, 
otherwise they might as well use morse code with long wave radios, it's 
also an "adequate" means of communication, very resiliant.

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

From dbogdanovic@counterpath.com  Fri Oct  9 07:49:15 2009
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From: Dean Bogdanovic <dbogdanovic@counterpath.com>
To: Mikael Abrahamsson <swmike@swm.pp.se>
Date: Fri, 9 Oct 2009 10:50:31 -0400
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I don't think that IM as always much more adequate. In emergency =20
situations, where time is essential, voice is most efficient =20
communication method and unfortunately in such situations networks =20
have high packetloss and jitter.

Dean

On Oct 9, 2009, at 10:19 AM, Mikael Abrahamsson wrote:

> On Fri, 9 Oct 2009, Henry Sinnreich wrote:
>
>> It is fortunate this point was discussed.
>
> Yes.
>
>> Designing for 7% packet loss seems to be clearly a non-objective =20
>> for a high performance voice codec.
>
> Why? Is there a concrete technical reason why it can't be done? Over =20
> these
> kinds of networks you'll see lots of jitter anyhow, is there a =20
> concrete
> technical reason why one can't make sound good enough for the other =20
> end to
> understand what is being said even at 10% packet loss with 200ms =20
> buffert?
> If you send one packet every 20ms, 10% packet loss means you're =20
> losing one
> packet every 200ms (on average).
>
>> Over such networks, IM is much more adequate.
>
> I think people want to have the option to speak to each other if =20
> they can,
> otherwise they might as well use morse code with long wave radios, =20
> it's
> also an "adequate" means of communication, very resiliant.
>
> --=20
> Mikael Abrahamsson    email: swmike@swm.pp.se
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From lars.eggert@nokia.com  Fri Oct  9 09:00:57 2009
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On 2009-10-9, at 17:50, Dean Bogdanovic wrote:
> I don't think that IM as always much more adequate. In emergency
> situations, where time is essential, voice is most efficient
> communication method and unfortunately in such situations networks
> have high packetloss and jitter.

My personal contribution to RFC1925bis, should it ever be published,  
will be Eggert's law: any IETF effort can be derailed by putting it  
into an emergency services context.

:-)

Lars
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From jean-marc.valin@octasic.com  Fri Oct  9 10:42:11 2009
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I think it's important here to distinguish between the charter and the 
precise requirements. The intent of the charter is to define the kind of 
problem we're trying to solve, whereas the requirements I-D should have all 
the details.

Regarding the charter, the intent of objectives 2, 3 and 4 was to specify 
the operating environment we were primarily targeting, which is the 
Internet. Also, the main reason for mentioning RTP/SIP/SDP/SRTP/XMPP was to 
acknowledge that these protocols are important for the IETF and thus that 
the codec designers should ensure that the codec fits well with these 
protocols and not to do anything that would create interoperability issues. 
As Henry pointed out, this can be interpreted as implying that we would not 
consider other protocols on the Internet, but this was not the intent here. 
We should probably make that clearer.

About packet loss, it's obviously not possible to make a codec that 
produces intelligible speech when you have gaps of 1 second in the 
transmission. I personally believe that it's not even necessary to target a 
particular loss rate (as many pointed out, it varies a lot), but that it's 
more about what to prioritize. For a codec that operates on the Internet, I 
believe it is for example more useful to improve the loss robustness than 
to squeeze a few more bits out of the bit-stream, especially considering 
all the header overhead. On other networks (especially the ones where 
bandwidth is very expensive), the trade-offs could be different.

	Jean-Marc

Mikael Abrahamsson wrote:
> On Fri, 9 Oct 2009, Henry Sinnreich wrote:
> 
>> It is fortunate this point was discussed.
> 
> Yes.
> 
>> Designing for 7% packet loss seems to be clearly a non-objective for a 
>> high performance voice codec.
> 
> Why? Is there a concrete technical reason why it can't be done? Over 
> these kinds of networks you'll see lots of jitter anyhow, is there a 
> concrete technical reason why one can't make sound good enough for the 
> other end to understand what is being said even at 10% packet loss with 
> 200ms buffert? If you send one packet every 20ms, 10% packet loss means 
> you're losing one packet every 200ms (on average).
> 
>> Over such networks, IM is much more adequate.
> 
> I think people want to have the option to speak to each other if they 
> can, otherwise they might as well use morse code with long wave radios, 
> it's also an "adequate" means of communication, very resiliant.
> 


From koen.vos@skype.net  Fri Oct  9 15:32:10 2009
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To: Ron <ron@debian.org>
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Quoting Ron <ron@debian.org>:

> Why should the jitter buffer presuppose to know when it would be a
> good time for the codec to compress?

State-of-the-art jitter buffers adaptively vary the speed of decoding  
packets to balance delay and packet loss when jitter changes.  
Accelerating or slowing down the decode speed can lead to audible  
artifacts however, and the jitter buffer should therefore be careful  
when to trigger such operations.

To do this, the decoder should pass a measure of "perceptual  
sensitivity to time-scale modification" to the jitter buffer (a voice  
activity measure could be a first approximation). The jitter buffer  
then returns one of these commands to the decoder:
  1. decode
  2. conceal
  3. decode + stretch
  4. decode + compress

All this is relevant and important. For example, 3G networks exhibit  
delay spikes of several seconds. Every few minutes the network just  
stops delivering packets, and then releases them in short order. Same  
for WiFi with delay spikes of up to a second (caused by the radio  
scanning for available access points). Quickly-adapting jitter  
buffering greatly helps with such non-stationary jitter.


> Shouldn't the jitter buffer just be giving the codec stats on  
> dataflow that it can parameterise how it sees best?

What kind of stats are you thinking of? The jitter buffer only needs  
the sensitivity measure from the decoder, and the codec developers are  
probably best at defining this.

koen.

From hoene@uni-tuebingen.de  Fri Oct  9 16:10:04 2009
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> > Why should the jitter buffer presuppose to know when it would be a
> > good time for the codec to compress?
> 
> State-of-the-art jitter buffers adaptively vary the speed of decoding
> packets to balance delay and packet loss when jitter changes.
> Accelerating or slowing down the decode speed can lead to audible
> artifacts however, and the jitter buffer should therefore be careful
> when to trigger such operations.
> 
> To do this, the decoder should pass a measure of "perceptual
> sensitivity to time-scale modification" to the jitter buffer (a voice
> activity measure could be a first approximation). The jitter buffer
> then returns one of these commands to the decoder:
>   1. decode
>   2. conceal
>   3. decode + stretch
>   4. decode + compress

Allow me to point out our recent report addressing the interface and those
PLC and strech/compress commands.
http://net.cs.uni-tuebingen.de/html/ristore2net/publications/papers/hoene_sb
c2009.pdf
Chapter 3 contains the speech receiver and dejittering part that might be
relevant for this discussion.

Christian



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Dear Lars Eggert, Dean Bogdanovic and Mikael Abrahamsson,

allow me to raise my voice on the issues of supporting a good voice
transmission even under harsh conditions.

> > I have earlier proposed that the full solution should be able to
> > cope with
> > 10% packet loss and 1.5-2 seconds of jitter.
>=20
> I'm not aware of any Internet paths where such conditions would be
> considered normal. As such I question why such cases should be
> supported as a requirement. (Obviously, if a solution happened to work
> under such conditions *while also* having negligible overhead under
> normal conditions, that'd be fine - but I'm simply not sure this can
> be done.)

Probably there are about a billion phone calls each days world-wide. If =
the
phone systems works well only on for normal conditions (e.g. 99,9% of =
them),
this would result in about one million unhappy users each day. In =
addition,
service centers would kept busy. Thus, if possible one should =
concentrate on
providing a well working solution even for most of remaining 0,099...%
percentage of conditions---as long as the technical overhead is not too
high.

Let me point out a recent research project by Prof. Dr. J=F6rg Ott, HKK
Helsinki called DT Talkie. =
http://www.netlab.tkk.fi/tutkimus/dtn/dttalkie/
It is a phone application for delay tolerant networks and J=F6rg told me =
that
it was the job of a student during a seminar.=20

Knowing these results, I described the requirements for a reliable =
system
and sketched a rough technical solution in
http://www.ietf.org/id/draft-hoene-avt-acs-requirements-00.txt=20
Please read Section 2.4 and 3.2.

Having such as Push-To-Talk like solution would also help to address =
other
problems described by Mikael and Koes,

Mikael:
> But the two major types I see that needs to be handled are:
>=20
> Wireless networks with ARQ such as 802.11, 3GPP and alike. These =
typically
> can induce 1-2 seconds of jitter, but with low or no packet loss.
>=20
> Routers/switches that do buffering and which use either FIFO/taildrop =
or
> some kind of more intelligent buffer management such as (W)RED or =
alike.
> These might drop a few percent of packets (usually never more as TCP =
backs
> off a lot at these packet drop rates and thus the links are usually =
not
> loaded more than what causes a few percent packet loss) plus induce =
jitter
> as other traffic affects this. Routers typically never buffer packets =
more
> than 600ms, often much lower, thus jitter is less unless there are
> multiple congested links in a row.
Koen:
> All this is relevant and important. For example, 3G networks exhibit
> delay spikes of several seconds. Every few minutes the network just
> stops delivering packets, and then releases them in short order. Same
> for WiFi with delay spikes of up to a second (caused by the radio
> scanning for available access points). Quickly-adapting jitter
> buffering greatly helps with such non-stationary jitter.

Of course, providing a DT-Talkie or Push-To-Talk application is not the =
task
of a codec. Rather, it should be standardized in the AVT/transport =
group.
However, a codec could support those applications with two features:

1. A highly bandwidth-efficient transmission of speech (while taking
advantage of high algorithmic delays and very large frame sizes)
2. Support of discontinues transmission modes to detect voice activity =
and
compress silence and background noise.

With best regards,

 Christian Hoene



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On Fri, Oct 09, 2009 at 03:33:53PM -0700, Koen Vos wrote:
> Quoting Ron <ron@debian.org>:
> 
> >Why should the jitter buffer presuppose to know when it would be a
> >good time for the codec to compress?
> 
> State-of-the-art jitter buffers adaptively vary the speed of
> decoding packets to balance delay and packet loss when jitter
> changes.

Fundamentally, elastic buffers are pretty simple and it's probably best
to try to keep them that way.  They need enough tension to stay as small
as possible when they aren't under stress, and enough elasticity to
quickly stretch without ever 'snapping'.  Where 'snapping' in our case
is a 'lost' packet that we've given up on ever receiving.  That's pretty
much all they really need to do to fulfil their role in these systems.
At least for the purpose of examining how they interact with codecs.

> Accelerating or slowing down the decode speed can lead to
> audible artifacts however,

That's the bit where this is a problem in many current codecs, indeed.

> and the jitter buffer should therefore be careful when to trigger
> such operations.

and that's I think where you and I first disagree.  And where Christian
and I see a bit more eye to eye.  But it may just be a choice of words
and polarities :)  I don't think the jitter buffer should be 'trigerring'
these things based on input from the codec -- I think the jitter buffer
should just provide rate information to the codec, and leave it to the
codec to decide when and how to act to meet the longer term goals that
we set for it.  As gradually as it needs to, to avoid abrupt artifacts.

> To do this, the decoder should pass a measure of "perceptual
> sensitivity to time-scale modification" to the jitter buffer (a
> voice activity measure could be a first approximation). The jitter
> buffer then returns one of these commands to the decoder:
>  1. decode
>  2. conceal
>  3. decode + stretch
>  4. decode + compress
> 
> All this is relevant and important.

Yes, but I think that saying it is _all_ that is relevant and important,
supposes too much that new codecs will be just like the old ones in all
the ways that this group is forming to solve :)  'conceal' is a rather
rough approximation for many different actions that could be taken in
many different sets of circumstances.

I think I'd rather see the elastic buffer make available to the codec
(and other parts of the system) some simple information along the
lines of:

 1. is it stretching or contracting.  (but codec need not not care)
 2. does it want the codec to stretch or contract (more/faster/slower)?
 3. how far away from 'snapping' are we at present,
    (ie. how long until the first MIA packet, or the end of the buffer)
    and how big will the hole be when we do.

Then the clever codec people can work their best magic armed with some
small foreknowledge of 'the future', that the delay in the buffer can
make available to them.

Trying to tell the codec which frames it should drop samples in, based
on information it gave you as an abstract hint, seems like too many
round trips and too much information loss.  Letting it know when it is
going too fast, or too slow, or when it should just hum to itself for
a while seems like a much more flexible way to achieve the same goals.
It surely knows better than any outside agent where and when it can
fudge a few samples that most people won't notice.  It may be able to
catch up more aggressively on some frames than the elastic buffer
requested, and amortise that on other frames it can't adjust so far,
for the same or better average rate of change to the buffer size.

Regions of "perceptual sensitivity to time-scale modification" don't
seem like something that would always neatly respect packet boundaries,
so that seems like too coarse a sieve to be optimal, yet the packet
buffer can act upon nothing finer.

> For example, 3G networks exhibit
> delay spikes of several seconds. Every few minutes the network just
> stops delivering packets, and then releases them in short order.
> Same for WiFi with delay spikes of up to a second (caused by the
> radio scanning for available access points). Quickly-adapting jitter
> buffering greatly helps with such non-stationary jitter.

Ok.  That's reality, so we can't argue with that, and we all agree
that when delays are that long most interactive applications are
doomed anyhow.  So the interesting question is how do we deal with
that and give the user as much of what they want as we can anyway.

I see two sides to it:

 1. The elastic buffer must stretch to fit the expected maximum latency.
    You know you'll 'never' lose packets if you just wait long enough
    for them, but some of them may be very, very, late.  But you also
    know that once the next load arrives, you probably won't need to
    wait that long again for a while -- so you can apply a lot of
    tension to contracting it toward 'realtime' again as quickly as
    possible.

 2. Sometimes you will really lose packets.  And at some point you just
    have to give up on those.  No amount of further stretching is likely
    to help you.  But if you gave the codec early warning about the
    probability of it not arriving, it will have had time to plan its
    best action for if it really doesn't come, and help you to wait for
    it as long as possible in case it still does, by stretching out the
    data you do have for as long as is reasonably can.

> What kind of stats are you thinking of? The jitter buffer only needs
> the sensitivity measure from the decoder, and the codec developers
> are probably best at defining this.

If the elastic buffer simply tells the codec when it needs to stretch
or shrink, then that covers the first case above.  As soon as the first
late packets are observed you tell the codec to start slowing down,
then more and more urgently as you realise you're really going to
underflow very soon.  Eventually you fade to noise.  Once the burst
comes in, you tell the codec to hurry things up for a while, because
reality is back and it's not going to wait.

In the case of speech, the user experience is that words  simply   started
slowing       down           [then stopped for a while]    then_came_more_
quickly_for_a_bit before it was all normal again.  But the really important
bit is that you never actually mi   d anything which was said.  Which is
mu h m  e annoy  g, and hard to comprehend when it happens.

In the case of a truly lost individual packet, an intelligent elastic
buffer could warn the codec this was to be expected N frames from now,
and give it time to prepare a synthesis if it thought it could mumble
its way through, or fade to silence gracefully if there will be too
many missing to fake it.  Again early stretching of the buffer and the
codec play rate can buy you a little more time for it to arrive than
would be expected from the "normal rate" of what you currently have
backed up in the buffer.

So to bring it back to what is really relevant here, what does an elastic
buffer need from the codec?  Probably not much.  A way to smoothly stretch
or shrink time by some variable proportion, to balance the tension in the
buffer.  A way to gracefully fade to comfort noise and back again when the
signal is interrupted.  That's about it.

What does the codec need from the elastic buffer?  Forewarning of when the
buffer is about to 'snap' seems like the main information that it could
make some use of.  Anything else?

I'm not really sure the relationship between these two components actually
needs to be much more complicated or formal than that.  What can't we do
with just those simple controls?  We get a "double-ended" elastic buffer
that can stretch in both directions, and require nothing more than that the
codec give us finer control over the average output sample rate than the
traditional quantum leaps from one rate to the next.

If the interactive music use case, or large crowd conferences, become
viable, the idea of only being able to make these sort of adjustments
during 'silence periods' must be steered well clear of, we need to be
able to steal or pad every sample that we can when necessary, since
there is no silence on the scale of a "normal" 1-to-1 phone conversation.

Am I missing something here?  Otherwise I think what we need for this will
mostly be covered by the time-stretching considerations of the codec, no?

 Ron



From hoene@uni-tuebingen.de  Tue Oct 13 02:13:30 2009
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To: "'Jean-Marc Valin'" <jean-marc.valin@usherbrooke.ca>
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HI Jean-Marc Valin,

> Had a quick look at the results section of your document (sorry, =
didn't
> read any of the details). From what I read, I have the following =
comments:

> 1) The mono results appear a bit low compared to what I usually obtain
> with my own testset and PQevalaudio (-0.5 at 64 kb/s), but that could =
be
> explained by a different testset and evaluation tool.

I would recommend to use the original PEAQ version from Opticom.=20

> 2) The stereo results are mostly consistent with the mono results
> because of a bug in the 0.6.0 and 0.6.1 versions (which is now fixed =
in
> git). I'm hoping that 0.7.0 will be the first version to have usable
stereo.

Then, I will start the scripts again.=20

> 3) The SBC results in stereo look a bit suspicious to me. It seems =
like
> the bit-rate only increases by ~10 kb/s for the same quality, which is =
a
> bit surprising. Is it possible that your test samples are almost mono? =


Most of the sample item contain single instruments. Only the =
=93refryc=94 (jazz
music by Ry Cooder) and =93refsop01=94 (opera) sample items contain more =
complex
content. I checked for all used items whether left and right channel =
differ
(they did) but still better sample items could be used for stereo (e.g. =
old
Beatles songs).

> I
> also heard that PEAQ didn't evaluate the stereo image at all.

Yesterday, I got the PEMO-Q audio assessment software from =
www.hoertest.de.
I will check whether it outperforms PEAQ.=20

Christian

>=20
> 	Jean-Marc
>=20
> Christian Hoene a =E9crit :
> >>> Why should the jitter buffer presuppose to know when it would be a
> >>> good time for the codec to compress?
> >> State-of-the-art jitter buffers adaptively vary the speed of =
decoding
> >> packets to balance delay and packet loss when jitter changes.
> >> Accelerating or slowing down the decode speed can lead to audible
> >> artifacts however, and the jitter buffer should therefore be =
careful
> >> when to trigger such operations.
> >>
> >> To do this, the decoder should pass a measure of "perceptual
> >> sensitivity to time-scale modification" to the jitter buffer (a =
voice
> >> activity measure could be a first approximation). The jitter buffer
> >> then returns one of these commands to the decoder:
> >>   1. decode
> >>   2. conceal
> >>   3. decode + stretch
> >>   4. decode + compress
> >
> > Allow me to point out our recent report addressing the interface and
those
> > PLC and strech/compress commands.
> >
http://net.cs.uni-tuebingen.de/html/ristore2net/publications/papers/hoene=
_sb
> > c2009.pdf
> > Chapter 3 contains the speech receiver and dejittering part that =
might
be
> > relevant for this discussion.
> >
> > Christian
> >
> >
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
> >
> >


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o Christian Hoene [10/10/09 01:11]:
>>> Why should the jitter buffer presuppose to know when it would be a
>>> good time for the codec to compress?
>> State-of-the-art jitter buffers adaptively vary the speed of decoding
>> packets to balance delay and packet loss when jitter changes.
>> Accelerating or slowing down the decode speed can lead to audible
>> artifacts however, and the jitter buffer should therefore be careful
>> when to trigger such operations.
>>
>> To do this, the decoder should pass a measure of "perceptual
>> sensitivity to time-scale modification" to the jitter buffer (a voice
>> activity measure could be a first approximation). The jitter buffer
>> then returns one of these commands to the decoder:
>>   1. decode
>>   2. conceal
>>   3. decode + stretch
>>   4. decode + compress
> 
> Allow me to point out our recent report addressing the interface and those
> PLC and strech/compress commands.
> http://net.cs.uni-tuebingen.de/html/ristore2net/publications/papers/hoene_sb
> c2009.pdf
> Chapter 3 contains the speech receiver and dejittering part that might be
> relevant for this discussion.

I had a look on that chapter, and I think it gives interesting 
implementation simplification if the whole design is "pull based" (i.e. 
thought completely from the sink side).

Specifically about the method and interface you describe in 3.2 i have a 
few comments:

1. It makes sense that a PLC in the codec would be employed not only for 
extrapolation, but also for interpolation (when packets after a gap are 
already received).

2. In order to not have excessive late loss, you have to start the audio 
sink with a time offset (timestamp offset relative to the receiver). 
Methods like the one you referenced from Liang et al [27] intend to 
minimize exactly this offset (which contributes to total latency, or 
mouth-to-ear delay), while keeping the expected late loss bounded (e.g. 
in [27] using L filter to estimate). Missing in your description is 
exactly this - how do you setup the offset to the sink in the first 
place, and how do you adapt it, e.g. if you notice changes in jitter, or 
notice clock skew ?

3. It appears that, if there is the need to (e.g. next audio not 
available), the time scale modification in your CDCD needs to slow down 
voice immediately, and do all on one audio block. A buffer based 
approach makes it possible, to *slowly* adapt the playout speed (and in 
both directions) this way this operation is less intrusive, because the 
amount of speed modification does not need to be that high; instead, a 
"longer-term goal" of playout buffer length can be set.

Back to the topic on codec PLC and time scale interface, if the codec 
can optimize those two for the case of interpolation (which I don't know 
whether this is true, being not deeply into codec internals), I think it 
makes sense to consider this as requirement for the codec.

About the question where the actual decision happens whether to 
stretch/compress or not, I think what your paper illustrates is that 
there are so many possibilities to design a dejitter/playout/plc buffer, 
and there are many factors that such a buffer might take into account, 
that it will be simpler to signal relevant voice characteristics 
parameters from the codec to the buffer, than the reverse.

Best Regards
Stefan Sayer

[27] Y. J. Liang, N. Foerber, and B. Girod, Adaptive playout 
scheduling and loss concealment
for voice communication over IP networks, IEEE Transactions on 
Multimedia, vol. 5,
no. 4, pp. 532543, Dec. 2003.

a version of this, optimized for high concurrent call count by 
predicting when to employ TD-WSOLA best, is implemented in SEMS 
(http://iptel.org/sems), see AmAdaptivePlayout in
http://svn.berlios.de/wsvn/sems/trunk/core/AmPlayoutBuffer.cpp .

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

-- 
Stefan Sayer
VoIP Services

stefan.sayer@iptego.com
www.iptego.com

IPTEGO GmbH
Wittenbergplatz 1
10789 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann

From Borilin@spiritdsp.com  Wed Oct 14 02:09:05 2009
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Dear all,

=20

This is to inform you that the new I-D is available, so now IP-MR codec
as the IWAC candidate is also described -
http://tools.ietf.org/search/draft-spiritdsp-ipmr-00=20

=20

best regards,

Slava Borilin

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial'>This is to inform you that the new I-D is =
available,
so now IP-MR codec as the IWAC candidate is also described - <a
href=3D"http://tools.ietf.org/search/draft-spiritdsp-ipmr-00">http://tool=
s.ietf.org/search/draft-spiritdsp-ipmr-00</a>
<o:p></o:p></span></font></p>

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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Subject: [codec] Question about scheduling CODEC BOF
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Andrew Malis wants to come to the CODEC BOF and he is an author of a  
PWE3 document. Currently PWE3 is scheduled at the same time the CODEC  
BOF. To resolve this the secretariat is considering moving the CODEC  
BOF to Friday Nov 13 from 13:00 to 15:15.

If this is a problem for anyone, speak up quickly!

Thanks, Cullen


From koen.vos@skype.net  Thu Oct 15 02:43:49 2009
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From: Koen Vos <koen.vos@skype.net>
To: Cullen Jennings <fluffy@cisco.com>
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Five people from Skype, including myself, were planning to attend on  
Thursday and have made detailed arrangements that would be hard (or at  
least costly) to modify..

koen.


Quoting Cullen Jennings <fluffy@cisco.com>:
>
> Andrew Malis wants to come to the CODEC BOF and he is an author of a  
> PWE3 document. Currently PWE3 is scheduled at the same time the  
> CODEC BOF. To resolve this the secretariat is considering moving the  
> CODEC BOF to Friday Nov 13 from 13:00 to 15:15.
>
> If this is a problem for anyone, speak up quickly!
>
> Thanks, Cullen
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>



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From: "rontlv" <ron.even.tlv@gmail.com>
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Hi,
My view that it should whenever there are fewer other sessions. Apparently
there is a big interest in this BOF. Last session on Friday looks good to
me. It will show real interest.
BTW: In Stockholm, at the Friday AVT session we had a good number of people
since there was a real interest in the ECN work.
Roni Even

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Cullen Jennings
> Sent: Thursday, October 15, 2009 8:29 AM
> To: codec@ietf.org
> Subject: [codec] Question about scheduling CODEC BOF
> 
> 
> Andrew Malis wants to come to the CODEC BOF and he is an author of a
> PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
> BOF. To resolve this the secretariat is considering moving the CODEC
> BOF to Friday Nov 13 from 13:00 to 15:15.
> 
> If this is a problem for anyone, speak up quickly!
> 
> Thanks, Cullen
> 
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> 
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From hsinnrei@adobe.com  Thu Oct 15 06:24:45 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Cullen Jennings <fluffy@cisco.com>, "codec@ietf.org" <codec@ietf.org>
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Moving the BOF to Friday is a problem for me.
Such last minute changes are very costly for intercontinental travel.

Thanks, Henry


On 10/15/09 1:28 AM, "Cullen Jennings" <fluffy@cisco.com> wrote:



Andrew Malis wants to come to the CODEC BOF and he is an author of a
PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
BOF. To resolve this the secretariat is considering moving the CODEC
BOF to Friday Nov 13 from 13:00 to 15:15.

If this is a problem for anyone, speak up quickly!

Thanks, Cullen

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


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<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Moving the BOF to Friday is a problem for me.<BR>
Such last minute changes are very costly for intercontinental travel.<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 10/15/09 1:28 AM, &quot;Cullen Jennings&quot; &lt;<a href=3D"fluffy@cisc=
o.com">fluffy@cisco.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'><BR>
<BR>
Andrew Malis wants to come to the CODEC BOF and he is an author of a <BR>
PWE3 document. Currently PWE3 is scheduled at the same time the CODEC <BR>
BOF. To resolve this the secretariat is considering moving the CODEC <BR>
BOF to Friday Nov 13 from 13:00 to 15:15.<BR>
<BR>
If this is a problem for anyone, speak up quickly!<BR>
<BR>
Thanks, Cullen<BR>
<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
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From hoene@uni-tuebingen.de  Thu Oct 15 06:52:30 2009
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Subject: Re: [codec] proposed charter
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Dear Peter Saint-Andre and all,

inline you will find my personal comments to the proposed charter.

> PROPOSED CHARTER
> 
> Problem Statement
> 
> There are no standardized, high-quality audio codecs that are optimized
> for use in interactive Internet applications and that can be widely
> implemented and easily distributed among  application developers,
> service operators, and end users.  This gap in the Internet protocol
> stack has hindered protocol designers from being able to specify
> mandatory-to-implement codecs in their protocols for the sake of
> interoperability, developers from building innovative, interactive
> applications, service operators from deploying affordable, high-quality
> services, and end users from enjoying an optimal user experience.
> Because the Internet Standards Process defines an open, transparent
> process for technology standardization, a wide range of codec experts
> have committed to actively working on such codecs within the IETF.  The
> Codec WG provides a structured forum for these efforts.

Sounds good!

> 
> Objectives
> 
> The goal of this group is to develop one or more high-quality,
> widely-distributable audio codecs that is optimized for use over the
> Internet. Core considerations include the following:
> 
> 1. Use in interactive applications (examples include but are not limited
> to point-to-point voice calls, multi-party voice conferencing,
> telepresence, teleoperation, in-game voice chat, and live music
performance)
> 
> 2. Addressing the real transport conditions of the Internet, especially
> packet loss

I do not understand this last consideration. What do you mean with this
statement? Are packet losses the only attribute of transmissions over the
Internet? At least it has been agreed upon that jitter compensation is also
relevant. I would suggest to rephrase this requirement to "Addressing the
real transport conditions of the Internet. Important attributes of packet
transmissions over the Internet are identified in cooperation with other
IETF working groups such as AVT and DCCP."

> 3. Ensuring interoperability with the Real-time Transport Protocol
> (RTP), including secure transport via SRTP
> 
> 4. Ensuring interoperability with Internet signalling technologies such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP)
> 
> Codecs optimized for very low bit rates 

Traditionally, very low bit codecs are operating below 2.4 kbps. It might
not be known to the Internet guys.
 
> and codecs for non-interactive
> audio are out of scope.

Maybe you can add "because they require other kind of optimizations". BTW:
Do you address multicast, too? If not, mention it here.
 
> Although the codec(s) produced by the group might be used as
> mandatory-to-implement technologies by designers of particular Internet
> protocols, it is explicitly not a goal of the group to produce a codec
> that will be mandated for use across the entire IETF or Internet
community.
>
> The group might produce only one codec or it might produce multiple
> codecs; however, to reduce confusion in the marketplace it shall
> endeavor not to function as a "codec factory" that produces a large
> number of codecs.

I would suggest to remove the part after the semicolon. Instead, maybe add
something like "The codec(s) shall be address a large range of typical
transmission conditions on the Internet."
 
> Work processes and technical requirements for achieving the foregoing
> objectives are outlined in draft-valin-codec-guidelines and
> draft-valin-codec-requirements respectively; these will be further
> refined by the group in accordance with the usual IETF procedures for
> arriving at rough consensus.

I think it is problematic to predefine a draft that must be chosen. A
critical minded person might think that this standardization process is a
put-up affair. Instead, it is in good tradition of the IETF that the best
technical approach should win. Just imagine that you codec experts cannot
agree on a common requirements document or that other codec guys make a
better proposal. We should allow competition because this leads to better
technical solutions. Thus, I suggest to change it in "... outlined in a
codec-guidelines draft such as the draft-valin-codec-guidelines-01 and in a
codec-requirements draft such as draft-valin-codec-requirements-01
respectively; ..."?

> Intellectual property rights (IPR) issues will be handled in accordance
> with BCP 78 and BCP 79. Many existing codecs with the technical
> attributes listed above are encumbered with licensing fees and other IPR
> claims that make royalty-free and wide distribution and use across the
> Internet community difficult.  The group will try to standardize codecs
> that can be relatively freely and easily distributed, and employed as
> much as possible. In doing so, it will adhere closely to these BCPs.
> More specifically, "In general, IETF working groups prefer technologies
> with no known IPR claims or, for technologies with claims against them,
> an offer of royalty-free licensing."  Note that in accordance with BCP
> 79, the group will not explicitly rule out the possibility of adopting
> technology that does not meet these IPR requirements; such decisions
> will be made in accordance with the rough consensus of the group.
> 
> Deliverables
> 
> 1. Guidelines that define the work process of the group, using
> draft-valin-codec-guidelines as the basis for achieving consensus. This
> document shall be Informational.
> 
> 2. Detailed technical requirements, using draft-valin-codec-requirements
> as the basis for achieving consensus.  This document shall be
Informational.

Again, I would suggest to change it to "Guidelines that defined the work
process of the group define in a common codec-guidelines document." and
"Detailed technical requirements defined in a common codec-guidelines
document." due to the same reasons.

> 3. Specification of one or more codecs that meet the agreed-upon
> requirements, in the form of Internet-Drafts that define the codec
> algorithm and source code for a reference implementation. The text
> description of each codec shall indicate which components of the encoder
> and decoder are mandatory, recommended, and optional.  It is envisioned
> that each such specification shall be a Proposed Standard document.
> 
> Milestones
> 
> Mar-2010: WGLC on Codec Standardization Guidelines
> Mar-2010: WGLC on Requirements
> May-2010: Codec Standardization Guidelines to IESG (Informational)
> May-2010: Requirements to IESG (Infomational)
+r
> Oct-2010: WGLC on codec specification(s)
> Jan-2011: Submit codec specification(s) to IESG (Standards Track)

This is an ambitious schedule. In my opinion the danger of not keeping this
schedule is high. But then again, who cares...

All other parts are fine for me.
 
With best regards,

 Christian


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On 10/15/09 12:28 AM, Cullen Jennings wrote:
> 
> Andrew Malis wants to come to the CODEC BOF and he is an author of a
> PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
> BOF. To resolve this the secretariat is considering moving the CODEC BOF
> to Friday Nov 13 from 13:00 to 15:15.
> 
> If this is a problem for anyone, speak up quickly!

My flight from Osaka leaves at 6:35 PM so this won't work for me.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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Cullen,
This question goes back to the point made by David on P2PSIP about
scheduling sessions on Friday.
I feel frustrated that I take seriously the point that initial agenda is not
final and that that the IETF meeting is from Monday morning to Friday
afternoon and plan accordingly.
Friday should be treated like other meeting day!!!! Unless there is a
different view from the IETF

Roni Even

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Cullen Jennings
> Sent: Thursday, October 15, 2009 8:29 AM
> To: codec@ietf.org
> Subject: [codec] Question about scheduling CODEC BOF
> 
> 
> Andrew Malis wants to come to the CODEC BOF and he is an author of a
> PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
> BOF. To resolve this the secretariat is considering moving the CODEC
> BOF to Friday Nov 13 from 13:00 to 15:15.
> 
> If this is a problem for anyone, speak up quickly!
> 
> Thanks, Cullen
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> 
> 
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From: Koen Vos <koen.vos@skype.net>
To: rontlv <ron.even.tlv@gmail.com>
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So to participate in a 2-hour meeting we all should reserve a whole  
week on our agenda?  And that 3x a year...?

Maybe no problem for people attending many meetings, but it sure seems  
like discrimination against outsiders only interested in a narrow area.

koen.



Quoting rontlv <ron.even.tlv@gmail.com>:

> Cullen,
> This question goes back to the point made by David on P2PSIP about
> scheduling sessions on Friday.
> I feel frustrated that I take seriously the point that initial agenda is not
> final and that that the IETF meeting is from Monday morning to Friday
> afternoon and plan accordingly.
> Friday should be treated like other meeting day!!!! Unless there is a
> different view from the IETF
>
> Roni Even
>
>> -----Original Message-----
>> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
>> Of Cullen Jennings
>> Sent: Thursday, October 15, 2009 8:29 AM
>> To: codec@ietf.org
>> Subject: [codec] Question about scheduling CODEC BOF
>>
>>
>> Andrew Malis wants to come to the CODEC BOF and he is an author of a
>> PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
>> BOF. To resolve this the secretariat is considering moving the CODEC
>> BOF to Friday Nov 13 from 13:00 to 15:15.
>>
>> If this is a problem for anyone, speak up quickly!
>>
>> Thanks, Cullen
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
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>> signature database 4509 (20091015) __________
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>
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> https://www.ietf.org/mailman/listinfo/codec
>



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Ok, clearly moving this to Friday is not an optimal trade off. We come  
up with an alternative plan. Thanks for the quick feedback.

On Oct 15, 2009, at 12:28 AM, Cullen Jennings wrote:

>
> Andrew Malis wants to come to the CODEC BOF and he is an author of a  
> PWE3 document. Currently PWE3 is scheduled at the same time the  
> CODEC BOF. To resolve this the secretariat is considering moving the  
> CODEC BOF to Friday Nov 13 from 13:00 to 15:15.
>
> If this is a problem for anyone, speak up quickly!
>
> Thanks, Cullen
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 10/7/09 9:34 PM, Jean-Marc Valin wrote:

>> 2/ In Section 6 what does this mean
>>
>>      *  Purely wireless applications will not be considered
>>
>> Does it exclude Wifi?  Maybe an example would clarify.....
> 
> Wifi is definitely within the scope here. I agree that "purely wireless"
> is not a great description. My personal view (we haven't discussed this
> in great details) is that we would rule out applications such as ham
> radio and cell phones (unless you're doing VoIP on that mobile), but
> include "IP over wireless links". Again, a better wording would be
> welcome if anyone can suggest one.

How about this?

"Wireless applications that do not communicate over the Internet
Protocol (e.g., radio) will not be considered"

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From jean-marc.valin@octasic.com  Fri Oct 16 13:41:46 2009
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Peter Saint-Andre wrote:
> How about this?
>
> "Wireless applications that do not communicate over the Internet
> Protocol (e.g., radio) will not be considered"

I think that's a very good way to put it.

    Jean-Marc

From stpeter@stpeter.im  Fri Oct 16 13:45:59 2009
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On 10/9/09 11:43 AM, Jean-Marc Valin wrote:

> Regarding the charter, the intent of objectives 2, 3 and 4 was to
> specify the operating environment we were primarily targeting, which is
> the Internet. Also, the main reason for mentioning RTP/SIP/SDP/SRTP/XMPP
> was to acknowledge that these protocols are important for the IETF and
> thus that the codec designers should ensure that the codec fits well
> with these protocols and not to do anything that would create
> interoperability issues. As Henry pointed out, this can be interpreted
> as implying that we would not consider other protocols on the Internet,
> but this was not the intent here. We should probably make that clearer.

Correct. Ensuring interoperability with those parts of the IETF protocol
stack doesn't imply that the WG will ignore interoperability with any
other protocols. IMHO the text "Core considerations *include the
following*..." and "Ensuring interoperability with Internet signalling
technologies *such as*..." shows that these considerations are by no
means exclusive.

Peter

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On 10/15/09 7:52 AM, Christian Hoene wrote:
> Dear Peter Saint-Andre and all,
> 
> inline you will find my personal comments to the proposed charter.

Thank you for your feedback.

>> PROPOSED CHARTER
>>
>> Problem Statement
>>
>> There are no standardized, high-quality audio codecs that are optimized
>> for use in interactive Internet applications and that can be widely
>> implemented and easily distributed among  application developers,
>> service operators, and end users.  This gap in the Internet protocol
>> stack has hindered protocol designers from being able to specify
>> mandatory-to-implement codecs in their protocols for the sake of
>> interoperability, developers from building innovative, interactive
>> applications, service operators from deploying affordable, high-quality
>> services, and end users from enjoying an optimal user experience.
>> Because the Internet Standards Process defines an open, transparent
>> process for technology standardization, a wide range of codec experts
>> have committed to actively working on such codecs within the IETF.  The
>> Codec WG provides a structured forum for these efforts.
> 
> Sounds good!
> 
>> Objectives
>>
>> The goal of this group is to develop one or more high-quality,
>> widely-distributable audio codecs that is optimized for use over the
>> Internet. Core considerations include the following:
>>
>> 1. Use in interactive applications (examples include but are not limited
>> to point-to-point voice calls, multi-party voice conferencing,
>> telepresence, teleoperation, in-game voice chat, and live music
> performance)
>> 2. Addressing the real transport conditions of the Internet, especially
>> packet loss
> 
> I do not understand this last consideration. What do you mean with this
> statement? Are packet losses the only attribute of transmissions over the
> Internet? 

By no means. The word "especially" does not exclude other conditions
that might be important to consider.

> At least it has been agreed upon that jitter compensation is also
> relevant. I would suggest to rephrase this requirement to "Addressing the
> real transport conditions of the Internet. Important attributes of packet
> transmissions over the Internet are identified in cooperation with other
> IETF working groups such as AVT and DCCP."

I think it might be more productive to say this:

***

2. Addressing the real transport conditions of the Internet as
identified and prioritized by the group

***

It's not really the task of the charter to define the real transport
conditions of the Internet, but if we include this is an objective of
the group then they should certainly keep those conditions in mind as
they complete their work.

>> 3. Ensuring interoperability with the Real-time Transport Protocol
>> (RTP), including secure transport via SRTP
>>
>> 4. Ensuring interoperability with Internet signalling technologies such
>> as Session Initiation Protocol (SIP), Session Description Protocol
>> (SDP), and Extensible Messaging and Presence Protocol (XMPP)
>>
>> Codecs optimized for very low bit rates 
> 
> Traditionally, very low bit codecs are operating below 2.4 kbps. It might
> not be known to the Internet guys.

Added.

>> and codecs for non-interactive
>> audio are out of scope.
> 
> Maybe you can add "because they require other kind of optimizations". 

In my working copy I have changed this to:

***

Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

***

> BTW:
> Do you address multicast, too? If not, mention it here.

By "multicast" do you mean many-to-many or one-to-many? We talk a bit
about that in the requirements draft.

>> Although the codec(s) produced by the group might be used as
>> mandatory-to-implement technologies by designers of particular Internet
>> protocols, it is explicitly not a goal of the group to produce a codec
>> that will be mandated for use across the entire IETF or Internet
> community.
>> The group might produce only one codec or it might produce multiple
>> codecs; however, to reduce confusion in the marketplace it shall
>> endeavor not to function as a "codec factory" that produces a large
>> number of codecs.
> 
> I would suggest to remove the part after the semicolon. Instead, maybe add
> something like "The codec(s) shall be address a large range of typical
> transmission conditions on the Internet."

Your proposed addition does not talk about the point you propose to
remove (i.e., we are not going to produce dozens of codecs) and the
point you add here is discussed elsewhere in the charter. Therefore I'm
not in favor of this change.

>> Work processes and technical requirements for achieving the foregoing
>> objectives are outlined in draft-valin-codec-guidelines and
>> draft-valin-codec-requirements respectively; these will be further
>> refined by the group in accordance with the usual IETF procedures for
>> arriving at rough consensus.
> 
> I think it is problematic to predefine a draft that must be chosen. 

I think it is problematic to mention no Internet-Drafts in the charter.
The existing I-Ds will form the starting point for work within the
group. Whether those I-Ds will remain as they are today (I doubt that
they will) is up to the group.

> A
> critical minded person might think that this standardization process is a
> put-up affair. Instead, it is in good tradition of the IETF that the best
> technical approach should win. Just imagine that you codec experts cannot
> agree on a common requirements document or that other codec guys make a
> better proposal. We should allow competition because this leads to better
> technical solutions. Thus, I suggest to change it in "... outlined in a
> codec-guidelines draft such as the draft-valin-codec-guidelines-01 and in a
> codec-requirements draft such as draft-valin-codec-requirements-01
> respectively; ..."?

The "such as" here is not especially helpful as guidance to the group or
as part of our contract with the IESG. Given that no one else on this
list has even mentioned the possibility of working on Internet-Drafts
for the procedural guidelines or technical requirements, I think it is
best to show the IESG that the published I-Ds are available for
inspection and will form the starting point of the group's efforts.
Those I-Ds are not final by any means, but at least they are something
that the group can refine over time. Which is exactly what the current
charter text says. However, I'm sure we can improve things slightly.

How about this?

***

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

***

>> Intellectual property rights (IPR) issues will be handled in accordance
>> with BCP 78 and BCP 79. Many existing codecs with the technical
>> attributes listed above are encumbered with licensing fees and other IPR
>> claims that make royalty-free and wide distribution and use across the
>> Internet community difficult.  The group will try to standardize codecs
>> that can be relatively freely and easily distributed, and employed as
>> much as possible. In doing so, it will adhere closely to these BCPs.
>> More specifically, "In general, IETF working groups prefer technologies
>> with no known IPR claims or, for technologies with claims against them,
>> an offer of royalty-free licensing."  Note that in accordance with BCP
>> 79, the group will not explicitly rule out the possibility of adopting
>> technology that does not meet these IPR requirements; such decisions
>> will be made in accordance with the rough consensus of the group.
>>
>> Deliverables
>>
>> 1. Guidelines that define the work process of the group, using
>> draft-valin-codec-guidelines as the basis for achieving consensus. This
>> document shall be Informational.
>>
>> 2. Detailed technical requirements, using draft-valin-codec-requirements
>> as the basis for achieving consensus.  This document shall be
> Informational.
> 
> Again, I would suggest to change it to "Guidelines that defined the work
> process of the group define in a common codec-guidelines document." and
> "Detailed technical requirements defined in a common codec-guidelines
> document." due to the same reasons.

In my working copy I have changed these to:

***

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the starting point for working
toward consensus. This document shall be Informational.

2. Detailed technical requirements, using
draft-valin-codec-requirementsas the starting point for working
toward consensus.  This document shall be Informational.

***

Does that text address your concerns?

>> 3. Specification of one or more codecs that meet the agreed-upon
>> requirements, in the form of Internet-Drafts that define the codec
>> algorithm and source code for a reference implementation. The text
>> description of each codec shall indicate which components of the encoder
>> and decoder are mandatory, recommended, and optional.  It is envisioned
>> that each such specification shall be a Proposed Standard document.
>>
>> Milestones
>>
>> Mar-2010: WGLC on Codec Standardization Guidelines
>> Mar-2010: WGLC on Requirements
>> May-2010: Codec Standardization Guidelines to IESG (Informational)
>> May-2010: Requirements to IESG (Infomational)
> +r
>> Oct-2010: WGLC on codec specification(s)
>> Jan-2011: Submit codec specification(s) to IESG (Standards Track)
> 
> This is an ambitious schedule. In my opinion the danger of not keeping this
> schedule is high. But then again, who cares...

You're right that it is aggressive. But it seems that we have a core
group of energetic contributors and that we might be able to make fast
progress if the WG is approved. But perhaps giving ourselves a few more
months here and there would be appropriate.

So perhaps this is slightly more realistic yet still ambitious:

***

Mar-2010: WGLC on Codec Standardization Guidelines
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: WGLC on Requirements
Jul-2010: Requirements to IESG (Infomational)
Feb-2011: WGLC on codec specification(s)
Apr-2011: Submit codec specification(s) to IESG (Standards Track)

***

> All other parts are fine for me.

Thanks again for your feedback.

Peter

- --
Peter Saint-Andre
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From: "rontlv" <ron.even.tlv@gmail.com>
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Koen,
The first comment is that the first agenda published is not final and may be
subject to changes. So I do not think that someone who plans his travel
based on the first agenda can complain if it changes later. For example in
the last IETD meeting the second ACT session was moved to Friday on the
final agenda, yet people who had interest attended the session.
I still think that reducing conflicts with other group would be the best on
Friday
Roni Even

> -----Original Message-----
> From: Koen Vos [mailto:koen.vos@skype.net]
> Sent: Friday, October 16, 2009 4:06 AM
> To: rontlv
> Cc: 'Cullen Jennings'; codec@ietf.org
> Subject: Re: [codec] Question about scheduling CODEC BOF
> 
> So to participate in a 2-hour meeting we all should reserve a whole
> week on our agenda?  And that 3x a year...?
> 
> Maybe no problem for people attending many meetings, but it sure seems
> like discrimination against outsiders only interested in a narrow area.
> 
> koen.
> 
> 
> 
> Quoting rontlv <ron.even.tlv@gmail.com>:
> 
> > Cullen,
> > This question goes back to the point made by David on P2PSIP about
> > scheduling sessions on Friday.
> > I feel frustrated that I take seriously the point that initial agenda
> is not
> > final and that that the IETF meeting is from Monday morning to Friday
> > afternoon and plan accordingly.
> > Friday should be treated like other meeting day!!!! Unless there is a
> > different view from the IETF
> >
> > Roni Even
> >
> >> -----Original Message-----
> >> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On
> Behalf
> >> Of Cullen Jennings
> >> Sent: Thursday, October 15, 2009 8:29 AM
> >> To: codec@ietf.org
> >> Subject: [codec] Question about scheduling CODEC BOF
> >>
> >>
> >> Andrew Malis wants to come to the CODEC BOF and he is an author of a
> >> PWE3 document. Currently PWE3 is scheduled at the same time the
> CODEC
> >> BOF. To resolve this the secretariat is considering moving the CODEC
> >> BOF to Friday Nov 13 from 13:00 to 15:15.
> >>
> >> If this is a problem for anyone, speak up quickly!
> >>
> >> Thanks, Cullen
> >>
> >> _______________________________________________
> >> codec mailing list
> >> codec@ietf.org
> >> https://www.ietf.org/mailman/listinfo/codec
> >>
> >>
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From: "rontlv" <ron.even.tlv@gmail.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>
References: <2D01AA05-0931-4516-B06B-D4FE97B5F73C@cisco.com> <6161C53A-5D9F-41CE-AAD3-D70AF68E25C5@cisco.com>
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Subject: Re: [codec] Question about scheduling CODEC BOF
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Cullen,
This is a pity since currently there was a conflict also with ppsp and
Friday would work better. Since the first agenda published is not final and
changes may occur I am not sure why we need to ask about moving to Friday.
In the last IETF AVT was moved on the final agenda to Friday without asking
on the list and still people who had interest attended
Roni Even

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Cullen Jennings
> Sent: Friday, October 16, 2009 7:21 PM
> To: Cullen Jennings
> Cc: codec@ietf.org
> Subject: Re: [codec] Question about scheduling CODEC BOF
> 
> 
> Ok, clearly moving this to Friday is not an optimal trade off. We come
> up with an alternative plan. Thanks for the quick feedback.
> 
> On Oct 15, 2009, at 12:28 AM, Cullen Jennings wrote:
> 
> >
> > Andrew Malis wants to come to the CODEC BOF and he is an author of a
> > PWE3 document. Currently PWE3 is scheduled at the same time the
> > CODEC BOF. To resolve this the secretariat is considering moving the
> > CODEC BOF to Friday Nov 13 from 13:00 to 15:15.
> >
> > If this is a problem for anyone, speak up quickly!
> >
> > Thanks, Cullen
> >
> > _______________________________________________
> > 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
> 
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From hoene@uni-tuebingen.de  Sat Oct 17 00:04:25 2009
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'rontlv'" <ron.even.tlv@gmail.com>
References: <2D01AA05-0931-4516-B06B-D4FE97B5F73C@cisco.com>	<4ad7bf76.100bca0a.5cc8.1a03@mx.google.com>	<20091015190550.213735b0oqb292wu@mail.skype.net> <4ad9077a.0d0bca0a.2406.10c6@mx.google.com>
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Subject: Re: [codec] Question about scheduling CODEC BOF
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Hi Roni Even,

I have to admit that I made the same mistake at Stockholm and missed both
AVT meetings. I think it is a typical beginners mistake to plan only for a
few days. Having this said, I have to agree that it make sense to join the
IETF meetings in full in future. Not only because of short time schedule
changes but also because this time can be used for making progress on own
drafts. Also, one might can go to other WG groups to get a broader picture
beyond the own field of work. Later might be beneficial even for the own WG.

Christian


> Koen,
> The first comment is that the first agenda published is not final and may
be
> subject to changes. So I do not think that someone who plans his travel
> based on the first agenda can complain if it changes later. For example in
> the last IETD meeting the second AVT session was moved to Friday on the
> final agenda, yet people who had interest attended the session.
> I still think that reducing conflicts with other group would be the best
on
> Friday
> Roni Even
> 
> > -----Original Message-----
> > From: Koen Vos [mailto:koen.vos@skype.net]
> > Sent: Friday, October 16, 2009 4:06 AM
> > To: rontlv
> > Cc: 'Cullen Jennings'; codec@ietf.org
> > Subject: Re: [codec] Question about scheduling CODEC BOF
> >
> > So to participate in a 2-hour meeting we all should reserve a whole
> > week on our agenda?  And that 3x a year...?
> >
> > Maybe no problem for people attending many meetings, but it sure seems
> > like discrimination against outsiders only interested in a narrow area.
> >
> > koen.
> >
> >
> >
> > Quoting rontlv <ron.even.tlv@gmail.com>:
> >
> > > Cullen,
> > > This question goes back to the point made by David on P2PSIP about
> > > scheduling sessions on Friday.
> > > I feel frustrated that I take seriously the point that initial agenda
> > is not
> > > final and that that the IETF meeting is from Monday morning to Friday
> > > afternoon and plan accordingly.
> > > Friday should be treated like other meeting day!!!! Unless there is a
> > > different view from the IETF
> > >
> > > Roni Even
> > >
> > >> -----Original Message-----
> > >> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On
> > Behalf
> > >> Of Cullen Jennings
> > >> Sent: Thursday, October 15, 2009 8:29 AM
> > >> To: codec@ietf.org
> > >> Subject: [codec] Question about scheduling CODEC BOF
> > >>
> > >>
> > >> Andrew Malis wants to come to the CODEC BOF and he is an author of a
> > >> PWE3 document. Currently PWE3 is scheduled at the same time the
> > CODEC
> > >> BOF. To resolve this the secretariat is considering moving the CODEC
> > >> BOF to Friday Nov 13 from 13:00 to 15:15.
> > >>
> > >> If this is a problem for anyone, speak up quickly!
> > >>
> > >> Thanks, Cullen
> > >>
> > >> _______________________________________________
> > >> codec mailing list
> > >> codec@ietf.org
> > >> https://www.ietf.org/mailman/listinfo/codec
> > >>
> > >>
> > >> __________ Information from ESET NOD32 Antivirus, version of virus
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> > > The message was checked by ESET NOD32 Antivirus.
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> > >
> > > _______________________________________________
> > > codec mailing list
> > > codec@ietf.org
> > > https://www.ietf.org/mailman/listinfo/codec
> > >
> >
> >
> >
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From gregory.ietf@gmail.com  Sat Oct 17 12:21:06 2009
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To: Cullen Jennings <fluffy@cisco.com>,codec@ietf.org
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Subject: Re: [codec] Question about scheduling CODEC BOF
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I am flying out Friday morning, right after the IESG + IAB meeting, 
due to a one-time family obligation on Saturday that I cannot miss. I 
would have to miss the BOF it was re-scheduled to Friday.

Gregory.

At 11:28 PM 10/14/2009, Cullen Jennings wrote:

>Andrew Malis wants to come to the CODEC BOF and he is an author of a
>PWE3 document. Currently PWE3 is scheduled at the same time the CODEC
>BOF. To resolve this the secretariat is considering moving the CODEC
>BOF to Friday Nov 13 from 13:00 to 15:15.
>
>If this is a problem for anyone, speak up quickly!
>
>Thanks, Cullen
>
>_______________________________________________
>codec mailing list
>codec@ietf.org
>https://www.ietf.org/mailman/listinfo/codec


From henry@sinnreich.net  Sun Oct 18 06:51:46 2009
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

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Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

This is an excellent formulation IMO.

Thanks, Henry


On 10/16/09 4:12 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> On 10/15/09 7:52 AM, Christian Hoene wrote:
>> > Dear Peter Saint-Andre and all,
>> >
>> > inline you will find my personal comments to the proposed charter.
> 
> Thank you for your feedback.
> 
>>> >> PROPOSED CHARTER
>>> >>
>>> >> Problem Statement
>>> >>
>>> >> There are no standardized, high-quality audio codecs that are optimized
>>> >> for use in interactive Internet applications and that can be widely
>>> >> implemented and easily distributed among  application developers,
>>> >> service operators, and end users.  This gap in the Internet protocol
>>> >> stack has hindered protocol designers from being able to specify
>>> >> mandatory-to-implement codecs in their protocols for the sake of
>>> >> interoperability, developers from building innovative, interactive
>>> >> applications, service operators from deploying affordable, high-quality
>>> >> services, and end users from enjoying an optimal user experience.
>>> >> Because the Internet Standards Process defines an open, transparent
>>> >> process for technology standardization, a wide range of codec experts
>>> >> have committed to actively working on such codecs within the IETF.  The
>>> >> Codec WG provides a structured forum for these efforts.
>> >
>> > Sounds good!
>> >
>>> >> Objectives
>>> >>
>>> >> The goal of this group is to develop one or more high-quality,
>>> >> widely-distributable audio codecs that is optimized for use over the
>>> >> Internet. Core considerations include the following:
>>> >>
>>> >> 1. Use in interactive applications (examples include but are not limited
>>> >> to point-to-point voice calls, multi-party voice conferencing,
>>> >> telepresence, teleoperation, in-game voice chat, and live music
>> > performance)
>>> >> 2. Addressing the real transport conditions of the Internet, especially
>>> >> packet loss
>> >
>> > I do not understand this last consideration. What do you mean with this
>> > statement? Are packet losses the only attribute of transmissions over the
>> > Internet?
> 
> By no means. The word "especially" does not exclude other conditions
> that might be important to consider.
> 
>> > At least it has been agreed upon that jitter compensation is also
>> > relevant. I would suggest to rephrase this requirement to "Addressing the
>> > real transport conditions of the Internet. Important attributes of packet
>> > transmissions over the Internet are identified in cooperation with other
>> > IETF working groups such as AVT and DCCP."
> 
> I think it might be more productive to say this:
> 
> ***
> 
> 2. Addressing the real transport conditions of the Internet as
> identified and prioritized by the group
> 
> ***
> 
> It's not really the task of the charter to define the real transport
> conditions of the Internet, but if we include this is an objective of
> the group then they should certainly keep those conditions in mind as
> they complete their work.
> 
>>> >> 3. Ensuring interoperability with the Real-time Transport Protocol
>>> >> (RTP), including secure transport via SRTP
>>> >>
>>> >> 4. Ensuring interoperability with Internet signalling technologies such
>>> >> as Session Initiation Protocol (SIP), Session Description Protocol
>>> >> (SDP), and Extensible Messaging and Presence Protocol (XMPP)
>>> >>
>>> >> Codecs optimized for very low bit rates
>> >
>> > Traditionally, very low bit codecs are operating below 2.4 kbps. It might
>> > not be known to the Internet guys.
> 
> Added.
> 
>>> >> and codecs for non-interactive
>>> >> audio are out of scope.
>> >
>> > Maybe you can add "because they require other kind of optimizations".
> 
> In my working copy I have changed this to:
> 
> ***
> 
> Codecs optimized for very low bit rates (typically below 2.4 kbps) and
> codecs for non-interactive audio are out of scope because they might
> necessitate specialized optimizations.
> 
> ***
> 
>> > BTW:
>> > Do you address multicast, too? If not, mention it here.
> 
> By "multicast" do you mean many-to-many or one-to-many? We talk a bit
> about that in the requirements draft.
> 
>>> >> Although the codec(s) produced by the group might be used as
>>> >> mandatory-to-implement technologies by designers of particular Internet
>>> >> protocols, it is explicitly not a goal of the group to produce a codec
>>> >> that will be mandated for use across the entire IETF or Internet
>> > community.
>>> >> The group might produce only one codec or it might produce multiple
>>> >> codecs; however, to reduce confusion in the marketplace it shall
>>> >> endeavor not to function as a "codec factory" that produces a large
>>> >> number of codecs.
>> >
>> > I would suggest to remove the part after the semicolon. Instead, maybe add
>> > something like "The codec(s) shall be address a large range of typical
>> > transmission conditions on the Internet."
> 
> Your proposed addition does not talk about the point you propose to
> remove (i.e., we are not going to produce dozens of codecs) and the
> point you add here is discussed elsewhere in the charter. Therefore I'm
> not in favor of this change.
> 
>>> >> Work processes and technical requirements for achieving the foregoing
>>> >> objectives are outlined in draft-valin-codec-guidelines and
>>> >> draft-valin-codec-requirements respectively; these will be further
>>> >> refined by the group in accordance with the usual IETF procedures for
>>> >> arriving at rough consensus.
>> >
>> > I think it is problematic to predefine a draft that must be chosen.
> 
> I think it is problematic to mention no Internet-Drafts in the charter.
> The existing I-Ds will form the starting point for work within the
> group. Whether those I-Ds will remain as they are today (I doubt that
> they will) is up to the group.
> 
>> > A
>> > critical minded person might think that this standardization process is a
>> > put-up affair. Instead, it is in good tradition of the IETF that the best
>> > technical approach should win. Just imagine that you codec experts cannot
>> > agree on a common requirements document or that other codec guys make a
>> > better proposal. We should allow competition because this leads to better
>> > technical solutions. Thus, I suggest to change it in "... outlined in a
>> > codec-guidelines draft such as the draft-valin-codec-guidelines-01 and in a
>> > codec-requirements draft such as draft-valin-codec-requirements-01
>> > respectively; ..."?
> 
> The "such as" here is not especially helpful as guidance to the group or
> as part of our contract with the IESG. Given that no one else on this
> list has even mentioned the possibility of working on Internet-Drafts
> for the procedural guidelines or technical requirements, I think it is
> best to show the IESG that the published I-Ds are available for
> inspection and will form the starting point of the group's efforts.
> Those I-Ds are not final by any means, but at least they are something
> that the group can refine over time. Which is exactly what the current
> charter text says. However, I'm sure we can improve things slightly.
> 
> How about this?
> 
> ***
> 
> Work processes and technical requirements for achieving the foregoing
> objectives are outlined in draft-valin-codec-guidelines and
> draft-valin-codec-requirements respectively; these documents will form
> the starting point for working toward consensus and will be refined by
> the group in accordance with the usual IETF procedures.
> 
> ***
> 
>>> >> Intellectual property rights (IPR) issues will be handled in accordance
>>> >> with BCP 78 and BCP 79. Many existing codecs with the technical
>>> >> attributes listed above are encumbered with licensing fees and other IPR
>>> >> claims that make royalty-free and wide distribution and use across the
>>> >> Internet community difficult.  The group will try to standardize codecs
>>> >> that can be relatively freely and easily distributed, and employed as
>>> >> much as possible. In doing so, it will adhere closely to these BCPs.
>>> >> More specifically, "In general, IETF working groups prefer technologies
>>> >> with no known IPR claims or, for technologies with claims against them,
>>> >> an offer of royalty-free licensing."  Note that in accordance with BCP
>>> >> 79, the group will not explicitly rule out the possibility of adopting
>>> >> technology that does not meet these IPR requirements; such decisions
>>> >> will be made in accordance with the rough consensus of the group.
>>> >>
>>> >> Deliverables
>>> >>
>>> >> 1. Guidelines that define the work process of the group, using
>>> >> draft-valin-codec-guidelines as the basis for achieving consensus. This
>>> >> document shall be Informational.
>>> >>
>>> >> 2. Detailed technical requirements, using draft-valin-codec-requirements
>>> >> as the basis for achieving consensus.  This document shall be
>> > Informational.
>> >
>> > Again, I would suggest to change it to "Guidelines that defined the work
>> > process of the group define in a common codec-guidelines document." and
>> > "Detailed technical requirements defined in a common codec-guidelines
>> > document." due to the same reasons.
> 
> In my working copy I have changed these to:
> 
> ***
> 
> 1. Guidelines that define the work process of the group, using
> draft-valin-codec-guidelines as the starting point for working
> toward consensus. This document shall be Informational.
> 
> 2. Detailed technical requirements, using
> draft-valin-codec-requirementsas the starting point for working
> toward consensus.  This document shall be Informational.
> 
> ***
> 
> Does that text address your concerns?
> 
>>> >> 3. Specification of one or more codecs that meet the agreed-upon
>>> >> requirements, in the form of Internet-Drafts that define the codec
>>> >> algorithm and source code for a reference implementation. The text
>>> >> description of each codec shall indicate which components of the encoder
>>> >> and decoder are mandatory, recommended, and optional.  It is envisioned
>>> >> that each such specification shall be a Proposed Standard document.
>>> >>
>>> >> Milestones
>>> >>
>>> >> Mar-2010: WGLC on Codec Standardization Guidelines
>>> >> Mar-2010: WGLC on Requirements
>>> >> May-2010: Codec Standardization Guidelines to IESG (Informational)
>>> >> May-2010: Requirements to IESG (Infomational)
>> > +r
>>> >> Oct-2010: WGLC on codec specification(s)
>>> >> Jan-2011: Submit codec specification(s) to IESG (Standards Track)
>> >
>> > This is an ambitious schedule. In my opinion the danger of not keeping this
>> > schedule is high. But then again, who cares...
> 
> You're right that it is aggressive. But it seems that we have a core
> group of energetic contributors and that we might be able to make fast
> progress if the WG is approved. But perhaps giving ourselves a few more
> months here and there would be appropriate.
> 
> So perhaps this is slightly more realistic yet still ambitious:
> 
> ***
> 
> Mar-2010: WGLC on Codec Standardization Guidelines
> May-2010: Codec Standardization Guidelines to IESG (Informational)
> May-2010: WGLC on Requirements
> Jul-2010: Requirements to IESG (Infomational)
> Feb-2011: WGLC on codec specification(s)
> Apr-2011: Submit codec specification(s) to IESG (Standards Track)
> 
> ***
> 
>> > All other parts are fine for me.
> 
> Thanks again for your feedback.
> 
> Peter
> 
> - --
> Peter Saint-Andre
> https://stpeter.im/
> 
> 
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> =n0pQ
> -----END PGP SIGNATURE-----
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> 


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<HEAD>
<TITLE>Re: [codec] proposed charter</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt=
'>Codecs optimized for very low bit rates (typically below 2.4 kbps) and<BR>
codecs for non-interactive audio are out of scope because they might<BR>
necessitate specialized optimizations.<BR>
<BR>
This is an excellent formulation IMO.<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 10/16/09 4:12 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@stp=
eter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:14pt'>-----BEGIN PGP SIGNED MESSAGE-----<BR>
Hash: SHA1<BR>
<BR>
On 10/15/09 7:52 AM, Christian Hoene wrote:<BR>
&gt; Dear Peter Saint-Andre and all,<BR>
&gt;<BR>
&gt; inline you will find my personal comments to the proposed charter.<BR>
<BR>
Thank you for your feedback.<BR>
<BR>
&gt;&gt; PROPOSED CHARTER<BR>
&gt;&gt;<BR>
&gt;&gt; Problem Statement<BR>
&gt;&gt;<BR>
&gt;&gt; There are no standardized, high-quality audio codecs that are opti=
mized<BR>
&gt;&gt; for use in interactive Internet applications and that can be widel=
y<BR>
&gt;&gt; implemented and easily distributed among &nbsp;application develop=
ers,<BR>
&gt;&gt; service operators, and end users. &nbsp;This gap in the Internet p=
rotocol<BR>
&gt;&gt; stack has hindered protocol designers from being able to specify<B=
R>
&gt;&gt; mandatory-to-implement codecs in their protocols for the sake of<B=
R>
&gt;&gt; interoperability, developers from building innovative, interactive=
<BR>
&gt;&gt; applications, service operators from deploying affordable, high-qu=
ality<BR>
&gt;&gt; services, and end users from enjoying an optimal user experience.<=
BR>
&gt;&gt; Because the Internet Standards Process defines an open, transparen=
t<BR>
&gt;&gt; process for technology standardization, a wide range of codec expe=
rts<BR>
&gt;&gt; have committed to actively working on such codecs within the IETF.=
 &nbsp;The<BR>
&gt;&gt; Codec WG provides a structured forum for these efforts.<BR>
&gt;<BR>
&gt; Sounds good!<BR>
&gt;<BR>
&gt;&gt; Objectives<BR>
&gt;&gt;<BR>
&gt;&gt; The goal of this group is to develop one or more high-quality,<BR>
&gt;&gt; widely-distributable audio codecs that is optimized for use over t=
he<BR>
&gt;&gt; Internet. Core considerations include the following:<BR>
&gt;&gt;<BR>
&gt;&gt; 1. Use in interactive applications (examples include but are not l=
imited<BR>
&gt;&gt; to point-to-point voice calls, multi-party voice conferencing,<BR>
&gt;&gt; telepresence, teleoperation, in-game voice chat, and live music<BR=
>
&gt; performance)<BR>
&gt;&gt; 2. Addressing the real transport conditions of the Internet, espec=
ially<BR>
&gt;&gt; packet loss<BR>
&gt;<BR>
&gt; I do not understand this last consideration. What do you mean with thi=
s<BR>
&gt; statement? Are packet losses the only attribute of transmissions over =
the<BR>
&gt; Internet?<BR>
<BR>
By no means. The word &quot;especially&quot; does not exclude other conditi=
ons<BR>
that might be important to consider.<BR>
<BR>
&gt; At least it has been agreed upon that jitter compensation is also<BR>
&gt; relevant. I would suggest to rephrase this requirement to &quot;Addres=
sing the<BR>
&gt; real transport conditions of the Internet. Important attributes of pac=
ket<BR>
&gt; transmissions over the Internet are identified in cooperation with oth=
er<BR>
&gt; IETF working groups such as AVT and DCCP.&quot;<BR>
<BR>
I think it might be more productive to say this:<BR>
<BR>
***<BR>
<BR>
2. Addressing the real transport conditions of the Internet as<BR>
identified and prioritized by the group<BR>
<BR>
***<BR>
<BR>
It's not really the task of the charter to define the real transport<BR>
conditions of the Internet, but if we include this is an objective of<BR>
the group then they should certainly keep those conditions in mind as<BR>
they complete their work.<BR>
<BR>
&gt;&gt; 3. Ensuring interoperability with the Real-time Transport Protocol=
<BR>
&gt;&gt; (RTP), including secure transport via SRTP<BR>
&gt;&gt;<BR>
&gt;&gt; 4. Ensuring interoperability with Internet signalling technologies=
 such<BR>
&gt;&gt; as Session Initiation Protocol (SIP), Session Description Protocol=
<BR>
&gt;&gt; (SDP), and Extensible Messaging and Presence Protocol (XMPP)<BR>
&gt;&gt;<BR>
&gt;&gt; Codecs optimized for very low bit rates<BR>
&gt;<BR>
&gt; Traditionally, very low bit codecs are operating below 2.4 kbps. It mi=
ght<BR>
&gt; not be known to the Internet guys.<BR>
<BR>
Added.<BR>
<BR>
&gt;&gt; and codecs for non-interactive<BR>
&gt;&gt; audio are out of scope.<BR>
&gt;<BR>
&gt; Maybe you can add &quot;because they require other kind of optimizatio=
ns&quot;.<BR>
<BR>
In my working copy I have changed this to:<BR>
<BR>
***<BR>
<BR>
Codecs optimized for very low bit rates (typically below 2.4 kbps) and<BR>
codecs for non-interactive audio are out of scope because they might<BR>
necessitate specialized optimizations.<BR>
<BR>
***<BR>
<BR>
&gt; BTW:<BR>
&gt; Do you address multicast, too? If not, mention it here.<BR>
<BR>
By &quot;multicast&quot; do you mean many-to-many or one-to-many? We talk a=
 bit<BR>
about that in the requirements draft.<BR>
<BR>
&gt;&gt; Although the codec(s) produced by the group might be used as<BR>
&gt;&gt; mandatory-to-implement technologies by designers of particular Int=
ernet<BR>
&gt;&gt; protocols, it is explicitly not a goal of the group to produce a c=
odec<BR>
&gt;&gt; that will be mandated for use across the entire IETF or Internet<B=
R>
&gt; community.<BR>
&gt;&gt; The group might produce only one codec or it might produce multipl=
e<BR>
&gt;&gt; codecs; however, to reduce confusion in the marketplace it shall<B=
R>
&gt;&gt; endeavor not to function as a &quot;codec factory&quot; that produ=
ces a large<BR>
&gt;&gt; number of codecs.<BR>
&gt;<BR>
&gt; I would suggest to remove the part after the semicolon. Instead, maybe=
 add<BR>
&gt; something like &quot;The codec(s) shall be address a large range of ty=
pical<BR>
&gt; transmission conditions on the Internet.&quot;<BR>
<BR>
Your proposed addition does not talk about the point you propose to<BR>
remove (i.e., we are not going to produce dozens of codecs) and the<BR>
point you add here is discussed elsewhere in the charter. Therefore I'm<BR>
not in favor of this change.<BR>
<BR>
&gt;&gt; Work processes and technical requirements for achieving the forego=
ing<BR>
&gt;&gt; objectives are outlined in draft-valin-codec-guidelines and<BR>
&gt;&gt; draft-valin-codec-requirements respectively; these will be further=
<BR>
&gt;&gt; refined by the group in accordance with the usual IETF procedures =
for<BR>
&gt;&gt; arriving at rough consensus.<BR>
&gt;<BR>
&gt; I think it is problematic to predefine a draft that must be chosen.<BR=
>
<BR>
I think it is problematic to mention no Internet-Drafts in the charter.<BR>
The existing I-Ds will form the starting point for work within the<BR>
group. Whether those I-Ds will remain as they are today (I doubt that<BR>
they will) is up to the group.<BR>
<BR>
&gt; A<BR>
&gt; critical minded person might think that this standardization process i=
s a<BR>
&gt; put-up affair. Instead, it is in good tradition of the IETF that the b=
est<BR>
&gt; technical approach should win. Just imagine that you codec experts can=
not<BR>
&gt; agree on a common requirements document or that other codec guys make =
a<BR>
&gt; better proposal. We should allow competition because this leads to bet=
ter<BR>
&gt; technical solutions. Thus, I suggest to change it in &quot;... outline=
d in a<BR>
&gt; codec-guidelines draft such as the draft-valin-codec-guidelines-01 and=
 in a<BR>
&gt; codec-requirements draft such as draft-valin-codec-requirements-01<BR>
&gt; respectively; ...&quot;?<BR>
<BR>
The &quot;such as&quot; here is not especially helpful as guidance to the g=
roup or<BR>
as part of our contract with the IESG. Given that no one else on this<BR>
list has even mentioned the possibility of working on Internet-Drafts<BR>
for the procedural guidelines or technical requirements, I think it is<BR>
best to show the IESG that the published I-Ds are available for<BR>
inspection and will form the starting point of the group's efforts.<BR>
Those I-Ds are not final by any means, but at least they are something<BR>
that the group can refine over time. Which is exactly what the current<BR>
charter text says. However, I'm sure we can improve things slightly.<BR>
<BR>
How about this?<BR>
<BR>
***<BR>
<BR>
Work processes and technical requirements for achieving the foregoing<BR>
objectives are outlined in draft-valin-codec-guidelines and<BR>
draft-valin-codec-requirements respectively; these documents will form<BR>
the starting point for working toward consensus and will be refined by<BR>
the group in accordance with the usual IETF procedures.<BR>
<BR>
***<BR>
<BR>
&gt;&gt; Intellectual property rights (IPR) issues will be handled in accor=
dance<BR>
&gt;&gt; with BCP 78 and BCP 79. Many existing codecs with the technical<BR=
>
&gt;&gt; attributes listed above are encumbered with licensing fees and oth=
er IPR<BR>
&gt;&gt; claims that make royalty-free and wide distribution and use across=
 the<BR>
&gt;&gt; Internet community difficult. &nbsp;The group will try to standard=
ize codecs<BR>
&gt;&gt; that can be relatively freely and easily distributed, and employed=
 as<BR>
&gt;&gt; much as possible. In doing so, it will adhere closely to these BCP=
s.<BR>
&gt;&gt; More specifically, &quot;In general, IETF working groups prefer te=
chnologies<BR>
&gt;&gt; with no known IPR claims or, for technologies with claims against =
them,<BR>
&gt;&gt; an offer of royalty-free licensing.&quot; &nbsp;Note that in accor=
dance with BCP<BR>
&gt;&gt; 79, the group will not explicitly rule out the possibility of adop=
ting<BR>
&gt;&gt; technology that does not meet these IPR requirements; such decisio=
ns<BR>
&gt;&gt; will be made in accordance with the rough consensus of the group.<=
BR>
&gt;&gt;<BR>
&gt;&gt; Deliverables<BR>
&gt;&gt;<BR>
&gt;&gt; 1. Guidelines that define the work process of the group, using<BR>
&gt;&gt; draft-valin-codec-guidelines as the basis for achieving consensus.=
 This<BR>
&gt;&gt; document shall be Informational.<BR>
&gt;&gt;<BR>
&gt;&gt; 2. Detailed technical requirements, using draft-valin-codec-requir=
ements<BR>
&gt;&gt; as the basis for achieving consensus. &nbsp;This document shall be=
<BR>
&gt; Informational.<BR>
&gt;<BR>
&gt; Again, I would suggest to change it to &quot;Guidelines that defined t=
he work<BR>
&gt; process of the group define in a common codec-guidelines document.&quo=
t; and<BR>
&gt; &quot;Detailed technical requirements defined in a common codec-guidel=
ines<BR>
&gt; document.&quot; due to the same reasons.<BR>
<BR>
In my working copy I have changed these to:<BR>
<BR>
***<BR>
<BR>
1. Guidelines that define the work process of the group, using<BR>
draft-valin-codec-guidelines as the starting point for working<BR>
toward consensus. This document shall be Informational.<BR>
<BR>
2. Detailed technical requirements, using<BR>
draft-valin-codec-requirementsas the starting point for working<BR>
toward consensus. &nbsp;This document shall be Informational.<BR>
<BR>
***<BR>
<BR>
Does that text address your concerns?<BR>
<BR>
&gt;&gt; 3. Specification of one or more codecs that meet the agreed-upon<B=
R>
&gt;&gt; requirements, in the form of Internet-Drafts that define the codec=
<BR>
&gt;&gt; algorithm and source code for a reference implementation. The text=
<BR>
&gt;&gt; description of each codec shall indicate which components of the e=
ncoder<BR>
&gt;&gt; and decoder are mandatory, recommended, and optional. &nbsp;It is =
envisioned<BR>
&gt;&gt; that each such specification shall be a Proposed Standard document=
.<BR>
&gt;&gt;<BR>
&gt;&gt; Milestones<BR>
&gt;&gt;<BR>
&gt;&gt; Mar-2010: WGLC on Codec Standardization Guidelines<BR>
&gt;&gt; Mar-2010: WGLC on Requirements<BR>
&gt;&gt; May-2010: Codec Standardization Guidelines to IESG (Informational)=
<BR>
&gt;&gt; May-2010: Requirements to IESG (Infomational)<BR>
&gt; +r<BR>
&gt;&gt; Oct-2010: WGLC on codec specification(s)<BR>
&gt;&gt; Jan-2011: Submit codec specification(s) to IESG (Standards Track)<=
BR>
&gt;<BR>
&gt; This is an ambitious schedule. In my opinion the danger of not keeping=
 this<BR>
&gt; schedule is high. But then again, who cares...<BR>
<BR>
You're right that it is aggressive. But it seems that we have a core<BR>
group of energetic contributors and that we might be able to make fast<BR>
progress if the WG is approved. But perhaps giving ourselves a few more<BR>
months here and there would be appropriate.<BR>
<BR>
So perhaps this is slightly more realistic yet still ambitious:<BR>
<BR>
***<BR>
<BR>
Mar-2010: WGLC on Codec Standardization Guidelines<BR>
May-2010: Codec Standardization Guidelines to IESG (Informational)<BR>
May-2010: WGLC on Requirements<BR>
Jul-2010: Requirements to IESG (Infomational)<BR>
Feb-2011: WGLC on codec specification(s)<BR>
Apr-2011: Submit codec specification(s) to IESG (Standards Track)<BR>
<BR>
***<BR>
<BR>
&gt; All other parts are fine for me.<BR>
<BR>
Thanks again for your feedback.<BR>
<BR>
Peter<BR>
<BR>
- --<BR>
Peter Saint-Andre<BR>
<a href=3D"https://stpeter.im/">https://stpeter.im/</a><BR>
<BR>
<BR>
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>IMHO the text "Core considerations *include the following*...
>" and "Ensuring interoperability with Internet signalling
>technologies *such as*..." shows that these considerations are by no
>means exclusive.

This seems to me a good formulation.
Maybe it is useful to clarify the generic idea that signaling dependent
codecs are out of scope =AD naturally.

Thanks, Henry


On 10/16/09 3:46 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 10/9/09 11:43 AM, Jean-Marc Valin wrote:
>=20
>> > Regarding the charter, the intent of objectives 2, 3 and 4 was to
>> > specify the operating environment we were primarily targeting, which i=
s
>> > the Internet. Also, the main reason for mentioning RTP/SIP/SDP/SRTP/XM=
PP
>> > was to acknowledge that these protocols are important for the IETF and
>> > thus that the codec designers should ensure that the codec fits well
>> > with these protocols and not to do anything that would create
>> > interoperability issues. As Henry pointed out, this can be interpreted
>> > as implying that we would not consider other protocols on the Internet=
,
>> > but this was not the intent here. We should probably make that clearer=
.
>=20
> Correct. Ensuring interoperability with those parts of the IETF protocol
> stack doesn't imply that the WG will ignore interoperability with any
> other protocols. IMHO the text "Core considerations *include the
> following*..." and "Ensuring interoperability with Internet signalling
> technologies *such as*..." shows that these considerations are by no
> means exclusive.
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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<HTML>
<HEAD>
<TITLE>Re: [codec] proposed charter</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt=
'>&gt;IMHO the text &quot;Core considerations *include the following*...<BR>
&gt;&quot; and &quot;Ensuring interoperability with Internet signalling<BR>
&gt;technologies *such as*...&quot; shows that these considerations are by =
no<BR>
&gt;means exclusive.<BR>
<BR>
This seems to me a good formulation.<BR>
Maybe it is useful to clarify the generic idea that signaling dependent cod=
ecs are out of scope &#8211; naturally.<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 10/16/09 3:46 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@stp=
eter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:14pt'>-----BEGIN PGP SIGNED MESSAGE-----<BR>
Hash: SHA1<BR>
<BR>
On 10/9/09 11:43 AM, Jean-Marc Valin wrote:<BR>
<BR>
&gt; Regarding the charter, the intent of objectives 2, 3 and 4 was to<BR>
&gt; specify the operating environment we were primarily targeting, which i=
s<BR>
&gt; the Internet. Also, the main reason for mentioning RTP/SIP/SDP/SRTP/XM=
PP<BR>
&gt; was to acknowledge that these protocols are important for the IETF and=
<BR>
&gt; thus that the codec designers should ensure that the codec fits well<B=
R>
&gt; with these protocols and not to do anything that would create<BR>
&gt; interoperability issues. As Henry pointed out, this can be interpreted=
<BR>
&gt; as implying that we would not consider other protocols on the Internet=
,<BR>
&gt; but this was not the intent here. We should probably make that clearer=
.<BR>
<BR>
Correct. Ensuring interoperability with those parts of the IETF protocol<BR=
>
stack doesn't imply that the WG will ignore interoperability with any<BR>
other protocols. IMHO the text &quot;Core considerations *include the<BR>
following*...&quot; and &quot;Ensuring interoperability with Internet signa=
lling<BR>
technologies *such as*...&quot; shows that these considerations are by no<B=
R>
means exclusive.<BR>
<BR>
Peter<BR>
<BR>
- --<BR>
Peter Saint-Andre<BR>
<a href=3D"https://stpeter.im/">https://stpeter.im/</a><BR>
<BR>
<BR>
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_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org/=
mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
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On 10/18/09 7:53 AM, henry@sinnreich.net wrote:
>>IMHO the text "Core considerations *include the following*...
>>" and "Ensuring interoperability with Internet signalling
>>technologies *such as*..." shows that these considerations are by no
>>means exclusive.
> 
> This seems to me a good formulation.
> Maybe it is useful to clarify the generic idea that signaling dependent
> codecs are out of scope – naturally.

Yes, that seems reasonable. Thanks for the suggestion. I shall post a
revised version of the proposed charter soon.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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On 10/7/09 4:27 PM, Stephan Wenger wrote:
> Thanks, Peter.
> 
> I find most of my comments adequately addressed, and in some other cases we
> can agree to disagree without another discussion at this point in time.  

OK, that seems reasonable.

> The
> one thing I want to comment on, hopefully constructively, is this:
> 
>>> 16: Section 5, "royalty-free".  Finally, this reasonably precise term is
>>> used.  Why not earlier?
>> Because royalty-free is one method for achieving the goal of easily
>> distributable. It is not, however, the only method.
>>
>>> 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".
>> This might need to be tightened up further.
>>
> 
> 
> Let me give it a try.  Replace the original paragraph
> 
>>    In accordance with BCP 78 [TRUST], the source code for the reference
>>    implementation should be available under the BSD license or whatever
>>    license is defined as acceptable by the IETF Trust when the relevant
>>    Internet-Drafts are published.  Also, in accordance with BCP 79
>>    [IPR], the codecs should preferably use technologies with no known
>>    IPR claims or technologies with an offer of royalty-free (RF)
>>    licensing.  Compatibility with the licensing of typical open source
>>    applications requires the avoidance of restrictive IPR claims.
> 
> With
> 
> "
> In accordance with BCP 78 [TRUST], the source code for the reference
> implementation should be available under the BSD-style license or whatever
> license is defined as acceptable by the IETF Trust when the relevant
> Internet-Drafts are published.
> 
> In accordance with the spirit of BCP 79 [IPR], the codecs should preferably
> use technologies with no known IPR claims, technologies available under
> patent non-assert covenants, or technologies with an offer of royalty-free
> (RF) licensing (in this order of preference).  Among other goals, adhering
> to these preferences should help achieving compatibility with the patent
> licensing requirements of most open source licenses, thereby allowing for
> open source applications of the specified codec(s).
> "
> 
> Motivation:
> 1. BCP78 compliance of the reference software has nothing to do with
> avoidance of patents of the algorithm, so the two should not be placed in
> the same paragraph.
> 2. I added non-assert covenants.  As I mentioned before, non-assert
> covenants are typically better for open source license compatibility than
> mere RF licensing covenants, unless those RF licensing covenants are
> intentionally worded to be friendly/compatible to open source.  Few of the
> RF licensing covenants posted in the IETF's IPR database are open source
> compatible at the time of writing (yes, I have reviewed all of them :-)
> 3. Expressing a preference of unencumbered-by-known-patents -> non-assert ->
> RF-licensing-commitment makes sense as it increases the chances of a "legal"
> open source implementation.

Thank you for your suggested text.

I am not a lawyer and even when I think about "IPR" issues it is in
relation to copyrights, not patents, so I will need to do a bit of
research and thinking before I am comfortable adding the text you have
provided. On the face of it, your text does seem to capture the intent
behind the current text, but I just want to give it a bit more thought
before I make any changes to the I-D.

> P.s.: Please note that I'm sending this in order to help you guys to express
> what I believe are your intentions.  I haven't changed my mind on viability
> of the exercise proposed, nor on my preference of the choice of venue for
> codec development.

Fair enough. I appreciate your generosity.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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Subject: [codec] Interesting IEEE Comm Mag Section on ITU-T Standards
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The most recent issue (October 2009, V. 47 n. 10)  of IEEE  
Communications Magazine has a special section of interest to us here  
on Speech Coding. Ask a friend if you do not have access to the IEEE  
Xplore library or a hard copy.

From stpeter@stpeter.im  Tue Oct 20 14:42:46 2009
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Subject: [codec] updated charter proposal
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Per recent list discussion...

######

Proposed Charter for Codec WG
Last Updated: 2009-10-20

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among  application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.
Because the Internet Standards Process defines an open, transparent
process for technology standardization, a wide range of codec experts
have committed to actively working on such codecs within the IETF.
The Codec WG provides a structured forum for these efforts.

Objectives

The goal of this group is to develop one or more high-quality,
widely-distributable audio codecs that is optimized for use over the
Internet. Core considerations include but are not necessarily limited
to the following:

1. Use in interactive applications (examples include but are not
limited to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music
performance)

2. Addressing the real transport conditions of the Internet as
identified and prioritized by the group

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
signalling-dependent codecs are out of scope

Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor not to function as a "codec factory" that produces a large
number of codecs.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

Intellectual property rights (IPR) issues will be handled in accordance
with BCP 78 and BCP 79. Many existing codecs with the technical
attributes listed above are encumbered with licensing fees and other IPR
claims that make royalty-free and wide distribution and use across the
Internet community difficult.  The group will try to standardize codecs
that can be relatively freely and easily distributed, and employed as
much as possible. In doing so, it will adhere closely to these BCPs.
More specifically, "In general, IETF working groups prefer technologies
with no known IPR claims or, for technologies with claims against them,
an offer of royalty-free licensing."  Note that in accordance with BCP
79, the group will not explicitly rule out the possibility of adopting
technology that does not meet these IPR requirements; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the starting point for working toward
consensus. This document shall be Informational.

2. Detailed technical requirements, using draft-valin-codec-requirements
as the starting point for working toward consensus.  This document
shall be Informational.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation. The text
description of each codec shall indicate which components of the encoder
and decoder are mandatory, recommended, and optional.  It is envisioned
that each such specification shall be a Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: WGLC on Requirements
Jul-2010: Requirements to IESG (Infomational)
Mar-2011: WGLC on codec specification(s)
Jun-2011: Submit codec specification(s) to IESG (Standards Track)

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From stpeter@stpeter.im  Tue Oct 20 14:52:11 2009
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Subject: [codec] BoF chairs
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I'm pleased to announce that Eric Burger has volunteered to co-chair the
Codec BoF with me in Hiroshima. It's doubtful that either of us would
co-chair the WG (if approved), but the two of us are working hard to
ensure that we'll have a productive session at IETF 76.

Thanks, Eric!

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From ron.even.tlv@gmail.com  Tue Oct 20 16:03:13 2009
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From: "rontlv" <ron.even.tlv@gmail.com>
To: "'Peter Saint-Andre'" <stpeter@stpeter.im>, <codec@ietf.org>
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Subject: Re: [codec] updated charter proposal
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Hi,
I did not have time to review the charter proposal being on vacation.

I would like to bring my major concern about 
" The goal of this group is to develop one or more codecs" . My view is that
the one of the major reason for lack of deployment of wideband is that there
are too many wideband codecs.
For example in high quality video conferencing you may need to support G.719
and flavors of AAC full band codecs to interoperate at full band with all
the current products.
I also find it strange that the timeline has one date for all codecs that
will be developed since I do not see any synchronization process between all
the codecs that will be approved if a WG is formed.
I think that having the right to do multiple codecs as part of the charter
will cause the group to approve all proposed codecs probably rubberstamping
them instead of going to collaboration on one good codec. I think that the
initial charter should have one codec and if later there will be a need
there should be a re-charter.

A minor comment is on 
"Because the Internet Standards Process defines an open, transparent process
for technology standardization, a wide range of codec experts have committed
to actively working on such codecs within the IETF.
The Codec WG provides a structured forum for these efforts."
I am not sure why this text is part of the charter. 

Being a little sarcastic about the commitment, I did not see the flexibility
of the experts when they needed to devote time for the IETF meeting.
Roni Even

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Peter Saint-Andre
> Sent: Tuesday, October 20, 2009 11:43 PM
> To: codec@ietf.org
> Subject: [codec] updated charter proposal
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Per recent list discussion...
> 
> ######
> 
> Proposed Charter for Codec WG
> Last Updated: 2009-10-20
> 
> Problem Statement
> 
> There are no standardized, high-quality audio codecs that are optimized
> for use in interactive Internet applications and that can be widely
> implemented and easily distributed among  application developers,
> service operators, and end users.  This gap in the Internet protocol
> stack has hindered protocol designers from being able to specify
> mandatory-to-implement codecs in their protocols for the sake of
> interoperability, developers from building innovative, interactive
> applications, service operators from deploying affordable, high-quality
> services, and end users from enjoying an optimal user experience.
> Because the Internet Standards Process defines an open, transparent
> process for technology standardization, a wide range of codec experts
> have committed to actively working on such codecs within the IETF.
> The Codec WG provides a structured forum for these efforts.
> 
> Objectives
> 
> The goal of this group is to develop one or more high-quality,
> widely-distributable audio codecs that is optimized for use over the
> Internet. Core considerations include but are not necessarily limited
> to the following:
> 
> 1. Use in interactive applications (examples include but are not
> limited to point-to-point voice calls, multi-party voice conferencing,
> telepresence, teleoperation, in-game voice chat, and live music
> performance)
> 
> 2. Addressing the real transport conditions of the Internet as
> identified and prioritized by the group
> 
> 3. Ensuring interoperability with the Real-time Transport Protocol
> (RTP), including secure transport via SRTP
> 
> 4. Ensuring interoperability with Internet signalling technologies such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
> signalling-dependent codecs are out of scope
> 
> Codecs optimized for very low bit rates (typically below 2.4 kbps) and
> codecs for non-interactive audio are out of scope because they might
> necessitate specialized optimizations.
> 
> Although the codec(s) produced by the group might be used as
> mandatory-to-implement technologies by designers of particular Internet
> protocols, it is explicitly not a goal of the group to produce a codec
> that will be mandated for use across the entire IETF or Internet
> community.
> 
> The group might produce only one codec or it might produce multiple
> codecs; however, to reduce confusion in the marketplace it shall
> endeavor not to function as a "codec factory" that produces a large
> number of codecs.
> 
> Work processes and technical requirements for achieving the foregoing
> objectives are outlined in draft-valin-codec-guidelines and
> draft-valin-codec-requirements respectively; these documents will form
> the starting point for working toward consensus and will be refined by
> the group in accordance with the usual IETF procedures.
> 
> Intellectual property rights (IPR) issues will be handled in accordance
> with BCP 78 and BCP 79. Many existing codecs with the technical
> attributes listed above are encumbered with licensing fees and other
> IPR
> claims that make royalty-free and wide distribution and use across the
> Internet community difficult.  The group will try to standardize codecs
> that can be relatively freely and easily distributed, and employed as
> much as possible. In doing so, it will adhere closely to these BCPs.
> More specifically, "In general, IETF working groups prefer technologies
> with no known IPR claims or, for technologies with claims against them,
> an offer of royalty-free licensing."  Note that in accordance with BCP
> 79, the group will not explicitly rule out the possibility of adopting
> technology that does not meet these IPR requirements; such decisions
> will be made in accordance with the rough consensus of the group.
> 
> Deliverables
> 
> 1. Guidelines that define the work process of the group, using
> draft-valin-codec-guidelines as the starting point for working toward
> consensus. This document shall be Informational.
> 
> 2. Detailed technical requirements, using draft-valin-codec-
> requirements
> as the starting point for working toward consensus.  This document
> shall be Informational.
> 
> 3. Specification of one or more codecs that meet the agreed-upon
> requirements, in the form of Internet-Drafts that define the codec
> algorithm and source code for a reference implementation. The text
> description of each codec shall indicate which components of the
> encoder
> and decoder are mandatory, recommended, and optional.  It is envisioned
> that each such specification shall be a Proposed Standard document.
> 
> Milestones
> 
> Mar-2010: WGLC on Codec Standardization Guidelines
> May-2010: Codec Standardization Guidelines to IESG (Informational)
> May-2010: WGLC on Requirements
> Jul-2010: Requirements to IESG (Infomational)
> Mar-2011: WGLC on codec specification(s)
> Jun-2011: Submit codec specification(s) to IESG (Standards Track)
> 
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> =MT0M
> -----END PGP SIGNATURE-----
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> 
> 
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> signature database 4527 (20091020) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> 
 

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From brian@freeswitch.org  Tue Oct 20 16:34:07 2009
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G.719 will never take off unless Ericsson gets off their part of the  
IP and actually will disclose the licensing details without having to  
sign an NDA.

/b

On Oct 20, 2009, at 6:03 PM, rontlv wrote:

> For example in high quality video conferencing you may need to  
> support G.719
> and flavors of AAC full band codecs to interoperate at full band  
> with all
> the current products.


From hoene@uni-tuebingen.de  Wed Oct 21 01:22:44 2009
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'rontlv'" <ron.even.tlv@gmail.com>, "'Peter Saint-Andre'" <stpeter@stpeter.im>, <codec@ietf.org>
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Subject: Re: [codec] updated charter proposal
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Hello,

> I would like to bring my major concern about
> " The goal of this group is to develop one or more codecs" . My view is
that
> the one of the major reason for lack of deployment of wideband is that
there
> are too many wideband codecs.
> For example in high quality video conferencing you may need to support
G.719
> and flavors of AAC full band codecs to interoperate at full band with all
> the current products.
...
> I think that having the right to do multiple codecs as part of the charter
> will cause the group to approve all proposed codecs probably
rubberstamping
> them instead of going to collaboration on one good codec. I think that the
> initial charter should have one codec and if later there will be a need
> there should be a re-charter.

I have to agree to Roni Even. This sentence and also the entire paragraph
remain unclear. I have understood that the questions on how many codecs and
on which to select are going to be solved by rough consensus. However, till
now I do not see in the charter any consensus nor statements on high level
selection or merging criteria. How could you decide on a codec in short
time, when you cannot agree on general selection statements?

In my previous email on the charter I try to address this point with the
statement that "The codec(s) shall address a large range of typical
transmission conditions on the Internet." However, maybe there is a
consensus on the statement: "Every offered codec is selected" following the
old Roman saying "Don't look a gift horse in the mouth". Or on some beauty
contest... Whatsoever, the paragraph remains too vague for me.

Christian


> Roni Even
> 
> > -----Original Message-----
> > From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> > Of Peter Saint-Andre
> > Sent: Tuesday, October 20, 2009 11:43 PM
> > To: codec@ietf.org
> > Subject: [codec] updated charter proposal
> >
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Per recent list discussion...
> >
> > ######
> >
> > Proposed Charter for Codec WG
> > Last Updated: 2009-10-20
> >
> > Problem Statement
> >
> > There are no standardized, high-quality audio codecs that are optimized
> > for use in interactive Internet applications and that can be widely
> > implemented and easily distributed among  application developers,
> > service operators, and end users.  This gap in the Internet protocol
> > stack has hindered protocol designers from being able to specify
> > mandatory-to-implement codecs in their protocols for the sake of
> > interoperability, developers from building innovative, interactive
> > applications, service operators from deploying affordable, high-quality
> > services, and end users from enjoying an optimal user experience.
> > Because the Internet Standards Process defines an open, transparent
> > process for technology standardization, a wide range of codec experts
> > have committed to actively working on such codecs within the IETF.
> > The Codec WG provides a structured forum for these efforts.
> >
> > Objectives
> >
> > The goal of this group is to develop one or more high-quality,
> > widely-distributable audio codecs that is optimized for use over the
> > Internet. Core considerations include but are not necessarily limited
> > to the following:
> >
> > 1. Use in interactive applications (examples include but are not
> > limited to point-to-point voice calls, multi-party voice conferencing,
> > telepresence, teleoperation, in-game voice chat, and live music
> > performance)
> >
> > 2. Addressing the real transport conditions of the Internet as
> > identified and prioritized by the group
> >
> > 3. Ensuring interoperability with the Real-time Transport Protocol
> > (RTP), including secure transport via SRTP
> >
> > 4. Ensuring interoperability with Internet signalling technologies such
> > as Session Initiation Protocol (SIP), Session Description Protocol
> > (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
> > signalling-dependent codecs are out of scope
> >
> > Codecs optimized for very low bit rates (typically below 2.4 kbps) and
> > codecs for non-interactive audio are out of scope because they might
> > necessitate specialized optimizations.
> >
> > Although the codec(s) produced by the group might be used as
> > mandatory-to-implement technologies by designers of particular Internet
> > protocols, it is explicitly not a goal of the group to produce a codec
> > that will be mandated for use across the entire IETF or Internet
> > community.
> >
> > The group might produce only one codec or it might produce multiple
> > codecs; however, to reduce confusion in the marketplace it shall
> > endeavor not to function as a "codec factory" that produces a large
> > number of codecs.
> >
> > Work processes and technical requirements for achieving the foregoing
> > objectives are outlined in draft-valin-codec-guidelines and
> > draft-valin-codec-requirements respectively; these documents will form
> > the starting point for working toward consensus and will be refined by
> > the group in accordance with the usual IETF procedures.
> >
> > Intellectual property rights (IPR) issues will be handled in accordance
> > with BCP 78 and BCP 79. Many existing codecs with the technical
> > attributes listed above are encumbered with licensing fees and other
> > IPR
> > claims that make royalty-free and wide distribution and use across the
> > Internet community difficult.  The group will try to standardize codecs
> > that can be relatively freely and easily distributed, and employed as
> > much as possible. In doing so, it will adhere closely to these BCPs.
> > More specifically, "In general, IETF working groups prefer technologies
> > with no known IPR claims or, for technologies with claims against them,
> > an offer of royalty-free licensing."  Note that in accordance with BCP
> > 79, the group will not explicitly rule out the possibility of adopting
> > technology that does not meet these IPR requirements; such decisions
> > will be made in accordance with the rough consensus of the group.
> >
> > Deliverables
> >
> > 1. Guidelines that define the work process of the group, using
> > draft-valin-codec-guidelines as the starting point for working toward
> > consensus. This document shall be Informational.
> >
> > 2. Detailed technical requirements, using draft-valin-codec-
> > requirements
> > as the starting point for working toward consensus.  This document
> > shall be Informational.
> >
> > 3. Specification of one or more codecs that meet the agreed-upon
> > requirements, in the form of Internet-Drafts that define the codec
> > algorithm and source code for a reference implementation. The text
> > description of each codec shall indicate which components of the
> > encoder
> > and decoder are mandatory, recommended, and optional.  It is envisioned
> > that each such specification shall be a Proposed Standard document.
> >
> > Milestones
> >
> > Mar-2010: WGLC on Codec Standardization Guidelines
> > May-2010: Codec Standardization Guidelines to IESG (Informational)
> > May-2010: WGLC on Requirements
> > Jul-2010: Requirements to IESG (Infomational)
> > Mar-2011: WGLC on codec specification(s)
> > Jun-2011: Submit codec specification(s) to IESG (Standards Track)
> >
> > -----BEGIN PGP SIGNATURE-----
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> > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
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> > /eoAoNomjEgkTahDyFLfIFH+zTxYSWGI
> > =MT0M
> > -----END PGP SIGNATURE-----
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
> >
> >
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From lars.eggert@nokia.com  Wed Oct 21 03:07:10 2009
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Subject: Re: [codec] updated charter proposal
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Hi,

the current charter text is starting to look OK. I'd like to raise one  
point below where I think we're missing a requirement.

To put this in perspective: I'm still not convinced that the IETF  
should actually start this work until we've clarified with the ITU-T/ 
3GPP/MPEG/etc. how this intersects with their work. But that's an  
orthogonal issue. If we were to start work, the current charter is  
looking reasonable. Maybe modulo what I'm going to say next...

On 2009-10-21, at 0:42, Peter Saint-Andre wrote:
> The goal of this group is to develop one or more high-quality,
> widely-distributable audio codecs that is optimized for use over the
> Internet. Core considerations include but are not necessarily limited
> to the following:
>
> 1. Use in interactive applications (examples include but are not
> limited to point-to-point voice calls, multi-party voice conferencing,
> telepresence, teleoperation, in-game voice chat, and live music
> performance)
>
> 2. Addressing the real transport conditions of the Internet as
> identified and prioritized by the group
>
> 3. Ensuring interoperability with the Real-time Transport Protocol
> (RTP), including secure transport via SRTP
>
> 4. Ensuring interoperability with Internet signalling technologies  
> such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
> signalling-dependent codecs are out of scope

The Internet uses a distributed congestion control mechanism built  
around the notion that when a flow observes packet loss, it is suppose  
to reduce its sending rate in some form over some time, in order to  
take load off a congested path. This behavior is something that I (as  
a transport guy) would see as a key differentiator between an  
"Internet codec" and a "telco codec", and I'd like to suggest that we  
reflect this in the charter in some form.

We can certainly argue about how useful this behavior would be in a  
low-bitrate voice codec. But it's my understanding that this effort  
isn't really targeting those, but is instead looking at wideband  
codecs - maybe ones using substantial levels of redundancy in the  
encoding - that can consume some noticeable amount of bandwidth. For  
those sorts of voice codecs, and even more so for video codecs, which  
have been brought up as the next logical thing for a potential CODEC  
WG to target, some sort of congestion-responsiveness is important when  
they are sharing path capacity over the Internet.

See for example http://tools.ietf.org/html/draft-westerlund-avt-ecn-for-rtp 
  for ongoing work related to exposing congestion information to media  
streams earlier than through packet loss (i.e., using ECN).

To be perfectly clear: This is NOT about enforcing TCP-friendliness or  
forcing the resulting codecs to conform to some other strict behavior  
under congestion. Due to the complexities of each coding scheme, how  
soon and by how much it can react to path congestion will be extremely  
codec-specific. But the idea is to recommend (or maybe even require)  
for "Internet codecs" to react appropriately to path congestion when  
they can.

Thoughts?

Lars
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From stewe@stewe.org  Wed Oct 21 04:40:21 2009
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Subject: Re: [codec] updated charter proposal
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Hi Lars,


On 10/21/09 3:07 AM, "Lars Eggert" <lars.eggert@nokia.com> wrote:

> [...]
> To be perfectly clear: This is NOT about enforcing TCP-friendliness or
> forcing the resulting codecs to conform to some other strict behavior
> under congestion. Due to the complexities of each coding scheme, how
> soon and by how much it can react to path congestion will be extremely
> codec-specific. But the idea is to recommend (or maybe even require)
> for "Internet codecs" to react appropriately to path congestion when
> they can.
> 
> Thoughts?

The traditional solution to achieve network-dictated (or suggested) bitrate
adaptivity is a scalable bitstream.  G.729.1 would be one example in the
audio space, and the SVC profiles of H.264 one in the video space.

The result of dropping layers in a scalable codec is reduced fidelity.
Whether users accept such a reduced fidelity depends IMHO mostly on the
application.  

To make one audio-centric example, in cheap/free desktop (video)
conferencing, dropping from wideband audio to telephony band speech in case
of congestion seems entirely reasonably, and is implemented in several
products I'm aware of.  For a music streaming/distribution service, the
story would be very different.  That is, in part, why those services put
heavy FEC on their unelastic media streams (quite to the contrary of the
intention of congestion control), or fall back to TCP transport with
multisecond buffers.

Adding scalability support to the requirements (perhaps at "SHOULD" level)
may make sense.  Or, at least, other ways of supporting network-dictated
variable bitrates, for example in the way the quantizer in MPEG'solder audio
codecs work (not that a lot of people use those mechanisms for bandwidth
adaptivity...).  Both ways lead into patent minefields, though.

In video, in contrast to audio, a real-time encoder has an ample number of
parameters to vary the outgoing bandwidth.  Most encoders operating under
real-time constraints play with those parameters constantly in a mechanism
that is commonly known as rate control.  They have to, because without rate
control the outgoing bitrate is variable over several orders of magnitude,
depending on content complexity.  Adding a network-dictated, pseudo-variable
target bitrate to the list of input parameters a rate control has to obey is
almost trivial.  Implementing something along these lines is common for
point-to-point video with real-time encoders.  Still, if that bitrate is too
low, your quality goes down into an unacceptable range.  But then, if you
get excessive losses due to congestion, your quality sucks, too.

When operating with non real-time encoders, or in multipoint scenarios
without transcoding, scalability is the technology of choice for bandwidth
adaptation.  All modern standardized video codecs offer scalability in one
form or another.  The SVC payload format draft contains quite a few words on
this subject.

Regards,
Stephan




From hoene@uni-tuebingen.de  Wed Oct 21 04:45:49 2009
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Dear Lars Eggert,

> The Internet uses a distributed congestion control mechanism built
> around the notion that when a flow observes packet loss, it is suppose
> to reduce its sending rate in some form over some time, in order to
> take load off a congested path. This behavior is something that I (as
> a transport guy) would see as a key differentiator between an
> "Internet codec" and a "telco codec", and I'd like to suggest that we
> reflect this in the charter in some form.
> 
> We can certainly argue about how useful this behavior would be in a
> low-bitrate voice codec. But it's my understanding that this effort
> isn't really targeting those, but is instead looking at wideband
> codecs - maybe ones using substantial levels of redundancy in the
> encoding - that can consume some noticeable amount of bandwidth. For
> those sorts of voice codecs, and even more so for video codecs, which
> have been brought up as the next logical thing for a potential CODEC
> WG to target, some sort of congestion-responsiveness is important when
> they are sharing path capacity over the Internet.
> 
> See for example
http://tools.ietf.org/html/draft-westerlund-avt-ecn-for-rtp
>   for ongoing work related to exposing congestion information to media
> streams earlier than through packet loss (i.e., using ECN).
> 
> To be perfectly clear: This is NOT about enforcing TCP-friendliness or
> forcing the resulting codecs to conform to some other strict behavior
> under congestion. Due to the complexities of each coding scheme, how
> soon and by how much it can react to path congestion will be extremely
> codec-specific. But the idea is to recommend (or maybe even require)
> for "Internet codecs" to react appropriately to path congestion when
> they can.

Since months I am raising this point on this mailing list. I suggested to
establish a procedure on how transport and codec expert should work together
to address these issues. 

However, based on the comments on this mailing list, I have the impression
that this is not wanted. I just remember some statements such as that we are
"only interested in a narrow area." 

I do have the impression that it is not intended to address the fundamental
problems of frequently bad VoIP quality or to enhance the Quality of
Experience - which would require to consider transport issues - but only in
making codecs. Hopefully, my observation is wrong.

With best regards,

 Christian Hoene





From lars.eggert@nokia.com  Wed Oct 21 04:56:59 2009
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Hi,

On 2009-10-21, at 14:45, Christian Hoene wrote:
> Since months I am raising this point on this mailing list. I  
> suggested to
> establish a procedure on how transport and codec expert should work  
> together
> to address these issues.

I'm sorry if I missed this - I cannot follow this email list very  
regularly.

Lars
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Stephan Wenger" <stewe@stewe.org>, "Lars Eggert" <lars.eggert@nokia.com>,  "Peter Saint-Andre" <stpeter@stpeter.im>
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Stephan,

You are absolutely right on scalability and its benefits - that's what
we are doing in SPIRIT IP-MR codec
(http://tools.ietf.org/search/draft-spiritdsp-ipmr-00) for many years
already.

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Stephan Wenger
Sent: Wednesday, October 21, 2009 3:40 PM
To: Lars Eggert; Peter Saint-Andre
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Hi Lars,


On 10/21/09 3:07 AM, "Lars Eggert" <lars.eggert@nokia.com> wrote:

> [...]
> To be perfectly clear: This is NOT about enforcing TCP-friendliness or
> forcing the resulting codecs to conform to some other strict behavior
> under congestion. Due to the complexities of each coding scheme, how
> soon and by how much it can react to path congestion will be extremely
> codec-specific. But the idea is to recommend (or maybe even require)
> for "Internet codecs" to react appropriately to path congestion when
> they can.
>=20
> Thoughts?

The traditional solution to achieve network-dictated (or suggested)
bitrate
adaptivity is a scalable bitstream.  G.729.1 would be one example in the
audio space, and the SVC profiles of H.264 one in the video space.

The result of dropping layers in a scalable codec is reduced fidelity.
Whether users accept such a reduced fidelity depends IMHO mostly on the
application. =20

To make one audio-centric example, in cheap/free desktop (video)
conferencing, dropping from wideband audio to telephony band speech in
case
of congestion seems entirely reasonably, and is implemented in several
products I'm aware of.  For a music streaming/distribution service, the
story would be very different.  That is, in part, why those services put
heavy FEC on their unelastic media streams (quite to the contrary of the
intention of congestion control), or fall back to TCP transport with
multisecond buffers.

Adding scalability support to the requirements (perhaps at "SHOULD"
level)
may make sense.  Or, at least, other ways of supporting network-dictated
variable bitrates, for example in the way the quantizer in MPEG'solder
audio
codecs work (not that a lot of people use those mechanisms for bandwidth
adaptivity...).  Both ways lead into patent minefields, though.

In video, in contrast to audio, a real-time encoder has an ample number
of
parameters to vary the outgoing bandwidth.  Most encoders operating
under
real-time constraints play with those parameters constantly in a
mechanism
that is commonly known as rate control.  They have to, because without
rate
control the outgoing bitrate is variable over several orders of
magnitude,
depending on content complexity.  Adding a network-dictated,
pseudo-variable
target bitrate to the list of input parameters a rate control has to
obey is
almost trivial.  Implementing something along these lines is common for
point-to-point video with real-time encoders.  Still, if that bitrate is
too
low, your quality goes down into an unacceptable range.  But then, if
you
get excessive losses due to congestion, your quality sucks, too.

When operating with non real-time encoders, or in multipoint scenarios
without transcoding, scalability is the technology of choice for
bandwidth
adaptation.  All modern standardized video codecs offer scalability in
one
form or another.  The SVC payload format draft contains quite a few
words on
this subject.

Regards,
Stephan



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

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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
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This proposed charter for the Codec WG is indeed fine tuned to the needs of=
 the Internet and its users.
IMO it should be left as it is.

No doubt, the opposition to the Internet codecs concept will continue, so l=
et's insure not to allow it being a distraction from moving on with the act=
ual work on the codec architecture, hopefully one or more reference impleme=
ntations and reports on test results - far more productive than endless mai=
l discussions.

So can we now please move on to the actual design documents?
Design proposals and reference code enhancements instead of objections?

Are any of the codec designers willing to set up a web site as is usual for=
 OS projects?
Should an Internet codec development web site (or more than one) be sponsor=
ed by the proposed WG?
Conceptual precedent: SIP Forum and its SIPit interoperability testing even=
ts - the codec web sites will obviously be different, but the intent can be=
 similar.

My strict personal 2 cents, including the na=EFve questions,

Henry


On 10/20/09 4:42 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Per recent list discussion...

######

Proposed Charter for Codec WG
Last Updated: 2009-10-20

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among  application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.
Because the Internet Standards Process defines an open, transparent
process for technology standardization, a wide range of codec experts
have committed to actively working on such codecs within the IETF.
The Codec WG provides a structured forum for these efforts.

Objectives

The goal of this group is to develop one or more high-quality,
widely-distributable audio codecs that is optimized for use over the
Internet. Core considerations include but are not necessarily limited
to the following:

1. Use in interactive applications (examples include but are not
limited to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music
performance)

2. Addressing the real transport conditions of the Internet as
identified and prioritized by the group

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
signalling-dependent codecs are out of scope

Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor not to function as a "codec factory" that produces a large
number of codecs.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

Intellectual property rights (IPR) issues will be handled in accordance
with BCP 78 and BCP 79. Many existing codecs with the technical
attributes listed above are encumbered with licensing fees and other IPR
claims that make royalty-free and wide distribution and use across the
Internet community difficult.  The group will try to standardize codecs
that can be relatively freely and easily distributed, and employed as
much as possible. In doing so, it will adhere closely to these BCPs.
More specifically, "In general, IETF working groups prefer technologies
with no known IPR claims or, for technologies with claims against them,
an offer of royalty-free licensing."  Note that in accordance with BCP
79, the group will not explicitly rule out the possibility of adopting
technology that does not meet these IPR requirements; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the starting point for working toward
consensus. This document shall be Informational.

2. Detailed technical requirements, using draft-valin-codec-requirements
as the starting point for working toward consensus.  This document
shall be Informational.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation. The text
description of each codec shall indicate which components of the encoder
and decoder are mandatory, recommended, and optional.  It is envisioned
that each such specification shall be a Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: WGLC on Requirements
Jul-2010: Requirements to IESG (Infomational)
Mar-2011: WGLC on codec specification(s)
Jun-2011: Submit codec specification(s) to IESG (Standards Track)

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<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>This proposed charter for the Codec WG is indeed fine tuned to the ne=
eds of the Internet and its users.<BR>
IMO it should be left as it is. <BR>
<BR>
No doubt, the opposition to the Internet codecs concept will continue, so l=
et&#8217;s insure not to allow it being a distraction from moving on with t=
he actual work on the codec architecture, hopefully one or more reference i=
mplementations and reports on test results &#8211; far more productive than=
 endless mail discussions.<BR>
<BR>
So can we now please move on to the actual design documents?<BR>
Design proposals and reference code enhancements instead of objections?<BR>
<BR>
Are any of the codec designers willing to set up a web site as is usual for=
 OS projects?<BR>
Should an Internet codec development web site (or more than one) be sponsor=
ed by the proposed WG?<BR>
Conceptual precedent: SIP Forum and its SIPit interoperability testing even=
ts &#8211; the codec web sites will obviously be different, but the intent =
can be similar.<BR>
<BR>
My strict personal 2 cents, including the na&iuml;ve questions,<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/20/09 4:42 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@s=
tpeter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>-----BEGIN PGP SIGNED MESSAGE-----<BR>
Hash: SHA1<BR>
<BR>
Per recent list discussion...<BR>
<BR>
######<BR>
<BR>
Proposed Charter for Codec WG<BR>
Last Updated: 2009-10-20<BR>
<BR>
Problem Statement<BR>
<BR>
There are no standardized, high-quality audio codecs that are optimized<BR>
for use in interactive Internet applications and that can be widely<BR>
implemented and easily distributed among &nbsp;application developers,<BR>
service operators, and end users. &nbsp;This gap in the Internet protocol<B=
R>
stack has hindered protocol designers from being able to specify<BR>
mandatory-to-implement codecs in their protocols for the sake of<BR>
interoperability, developers from building innovative, interactive<BR>
applications, service operators from deploying affordable, high-quality<BR>
services, and end users from enjoying an optimal user experience.<BR>
Because the Internet Standards Process defines an open, transparent<BR>
process for technology standardization, a wide range of codec experts<BR>
have committed to actively working on such codecs within the IETF.<BR>
The Codec WG provides a structured forum for these efforts.<BR>
<BR>
Objectives<BR>
<BR>
The goal of this group is to develop one or more high-quality,<BR>
widely-distributable audio codecs that is optimized for use over the<BR>
Internet. Core considerations include but are not necessarily limited<BR>
to the following:<BR>
<BR>
1. Use in interactive applications (examples include but are not<BR>
limited to point-to-point voice calls, multi-party voice conferencing,<BR>
telepresence, teleoperation, in-game voice chat, and live music<BR>
performance)<BR>
<BR>
2. Addressing the real transport conditions of the Internet as<BR>
identified and prioritized by the group<BR>
<BR>
3. Ensuring interoperability with the Real-time Transport Protocol<BR>
(RTP), including secure transport via SRTP<BR>
<BR>
4. Ensuring interoperability with Internet signalling technologies such<BR>
as Session Initiation Protocol (SIP), Session Description Protocol<BR>
(SDP), and Extensible Messaging and Presence Protocol (XMPP); however,<BR>
signalling-dependent codecs are out of scope<BR>
<BR>
Codecs optimized for very low bit rates (typically below 2.4 kbps) and<BR>
codecs for non-interactive audio are out of scope because they might<BR>
necessitate specialized optimizations.<BR>
<BR>
Although the codec(s) produced by the group might be used as<BR>
mandatory-to-implement technologies by designers of particular Internet<BR>
protocols, it is explicitly not a goal of the group to produce a codec<BR>
that will be mandated for use across the entire IETF or Internet community.=
<BR>
<BR>
The group might produce only one codec or it might produce multiple<BR>
codecs; however, to reduce confusion in the marketplace it shall<BR>
endeavor not to function as a &quot;codec factory&quot; that produces a lar=
ge<BR>
number of codecs.<BR>
<BR>
Work processes and technical requirements for achieving the foregoing<BR>
objectives are outlined in draft-valin-codec-guidelines and<BR>
draft-valin-codec-requirements respectively; these documents will form<BR>
the starting point for working toward consensus and will be refined by<BR>
the group in accordance with the usual IETF procedures.<BR>
<BR>
Intellectual property rights (IPR) issues will be handled in accordance<BR>
with BCP 78 and BCP 79. Many existing codecs with the technical<BR>
attributes listed above are encumbered with licensing fees and other IPR<BR=
>
claims that make royalty-free and wide distribution and use across the<BR>
Internet community difficult. &nbsp;The group will try to standardize codec=
s<BR>
that can be relatively freely and easily distributed, and employed as<BR>
much as possible. In doing so, it will adhere closely to these BCPs.<BR>
More specifically, &quot;In general, IETF working groups prefer technologie=
s<BR>
with no known IPR claims or, for technologies with claims against them,<BR>
an offer of royalty-free licensing.&quot; &nbsp;Note that in accordance wit=
h BCP<BR>
79, the group will not explicitly rule out the possibility of adopting<BR>
technology that does not meet these IPR requirements; such decisions<BR>
will be made in accordance with the rough consensus of the group.<BR>
<BR>
Deliverables<BR>
<BR>
1. Guidelines that define the work process of the group, using<BR>
draft-valin-codec-guidelines as the starting point for working toward<BR>
consensus. This document shall be Informational.<BR>
<BR>
2. Detailed technical requirements, using draft-valin-codec-requirements<BR=
>
as the starting point for working toward consensus. &nbsp;This document<BR>
shall be Informational.<BR>
<BR>
3. Specification of one or more codecs that meet the agreed-upon<BR>
requirements, in the form of Internet-Drafts that define the codec<BR>
algorithm and source code for a reference implementation. The text<BR>
description of each codec shall indicate which components of the encoder<BR=
>
and decoder are mandatory, recommended, and optional. &nbsp;It is envisione=
d<BR>
that each such specification shall be a Proposed Standard document.<BR>
<BR>
Milestones<BR>
<BR>
Mar-2010: WGLC on Codec Standardization Guidelines<BR>
May-2010: Codec Standardization Guidelines to IESG (Informational)<BR>
May-2010: WGLC on Requirements<BR>
Jul-2010: Requirements to IESG (Infomational)<BR>
Mar-2011: WGLC on codec specification(s)<BR>
Jun-2011: Submit codec specification(s) to IESG (Standards Track)<BR>
<BR>
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_______________________________________________<BR>
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<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
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Subject: Re: [codec] updated charter proposal
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Hi,

Lars Eggert wrote:
> The Internet uses a distributed congestion control mechanism built 
> around the notion that when a flow observes packet loss, it is suppose 
> to reduce its sending rate in some form over some time, in order to take 
> load off a congested path. This behavior is something that I (as a 
> transport guy) would see as a key differentiator between an "Internet 
> codec" and a "telco codec", and I'd like to suggest that we reflect this 
> in the charter in some form.

I totally agree that the bit-rate should adapt to network congestion. This 
is why the requirements draft includes "Having the possibility to use 
smooth bit-rate changes with one byte/frame resolution". This means that 
the codec can produce any rate that is needed to avoid congestion (not just 
a few pre-selected ones). Now, we must be careful to assign issues to the 
correct "layer" in the VoIP stack. I do not believe it is the codec's job 
to *decide* which bit-rate to select. This should be left to another 
component of the application (the RTP stack?), which can then tell the 
codec what bit-rate to use.

If there are more features that the codec can add to help with congestion, 
this is certainly something we can add to the drafts. Scalable codecs have 
also been discussed. I don't like the word because people seem to use it 
for different meanings. In one meaning it is simply the ability to use any 
rate, which is addressed above. In the other meaning, it means having 
layers that can be stripped off. I believe this can be a useful feature 
(which *is* listed in the current drafts), but mostly for conferencing, 
rather than congestion. The point of layered codec for congestion in 
particular is that a router in the middle can reduce the bit-rate to adjust 
to congestion. However, I see two problems here: 1) most routers does not 
(and should not have) knowledge of the codec and thus cannot change the 
rate; and 2) layers are usually larger than a single byte per frame so 
relying only on a layered approach might compromise the idea of having 
small bit-rate increments. But as I said earlier, I'm not opposed to 
layered codecs because they have other uses. This is something that would 
have to be carefully analysed during development.

> To be perfectly clear: This is NOT about enforcing TCP-friendliness or 
> forcing the resulting codecs to conform to some other strict behavior 
> under congestion. Due to the complexities of each coding scheme, how 
> soon and by how much it can react to path congestion will be extremely 
> codec-specific. But the idea is to recommend (or maybe even require) for 
> "Internet codecs" to react appropriately to path congestion when they can.

OK, I may be wrong here, but my understanding is that how much the rate 
should change depending on congestion is actually not that codec-specific. 
I get the impression that the tough part is figuring out what rate you can 
use without causing packets to be dropped. Once you figure this out, you 
can just tell the codec to use that rate. On top of that, the codec may be 
able to do a bit better by providing a "service" to the other layers. On 
example here (also included in the draft) is VBR, which means that the 
application specifies an average rate and the codec decides how to vary the 
rate to achieve that (subject to some rate control constraints). Would 
other such mechanisms be useful to deal with congestion?

Cheers,

	Jean-Marc

From mramalho@cisco.com  Wed Oct 21 06:57:09 2009
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "Jean-Marc Valin" <jean-marc.valin@octasic.com>, "Lars Eggert" <lars.eggert@nokia.com>
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All,

Just a thought for everyone. Prior to the last BoF I posted email that
suggested that the codec should have an ability to change encoding rate
as a function of some form of congestion intelligence (perhaps through
an API, perhaps a hint at the RTP layer, perhaps on a congestion
indication, etc.). That "should" was not necessarily a "must" because a
given application may tell the codec to go full steam ahead anyway
because what it knows about the transport environment. And I think that
wording has been accommodated.

Consider Jean-Marc's words: "I get the impression that the tough part is
figuring out what rate you can use without causing packets to be
dropped."

Indeed.

Consider the case where the bottleneck link for your end-to-end codec
application (where most of your packets are being dropped) is such that
all the UDP traffic (e.g., your VoIP and other UDP applications) is a
minority traffic. That is, the majority of traffic is TCP-friendly (TCP,
SCTP, etc.). And that the bandwidth of this link is large relative to
your VoIP stream. Now suppose that this link is someplace in one of the
middle hops of your end-to-end flow.

That link is being "filled up" (and causing drops for your flow)
according to the TCP dynamics of the other TCP traffic. Virtually
NOTHING YOU DO ON YOUR FLOW will have a significant impact on your
packet drop performance! Any bandwidth you "vacate" by reducing your
rate becomes "absorbed" (in a sense) by the TCP flows sharing the
remainder of the bandwidth.

If your application somehow (magically) knew this was the case, perhaps
it would NOT reduce rate. Now that isn't TCP friendly. And some codec
designs actually increase rate in those cases (by reducing rate all they
can - but then adding FEC bandwidth to the lowest rate).

I think the best we can do here is to design in the knobs to have the
codec change rate (or add FEC if that is a codec capability) to
accommodate scenarios such as the above.

Michael Ramalho

PS - As a side note, if the bottleneck link had majority traffic
consisting of similar VoIP flows - the correct behavior would be to
reduce rate as TCP does. That is why the answer for a link dominated by
ITSP traffic would be to reduce rate. And this is why the VoIP
application ideally would benefit from "other intelligence" regarding
congestion properties.

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Wednesday, October 21, 2009 9:27 AM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Hi,

Lars Eggert wrote:
> The Internet uses a distributed congestion control mechanism built=20
> around the notion that when a flow observes packet loss, it is suppose

> to reduce its sending rate in some form over some time, in order to
take=20
> load off a congested path. This behavior is something that I (as a=20
> transport guy) would see as a key differentiator between an "Internet=20
> codec" and a "telco codec", and I'd like to suggest that we reflect
this=20
> in the charter in some form.

I totally agree that the bit-rate should adapt to network congestion.
This=20
is why the requirements draft includes "Having the possibility to use=20
smooth bit-rate changes with one byte/frame resolution". This means that

the codec can produce any rate that is needed to avoid congestion (not
just=20
a few pre-selected ones). Now, we must be careful to assign issues to
the=20
correct "layer" in the VoIP stack. I do not believe it is the codec's
job=20
to *decide* which bit-rate to select. This should be left to another=20
component of the application (the RTP stack?), which can then tell the=20
codec what bit-rate to use.

If there are more features that the codec can add to help with
congestion,=20
this is certainly something we can add to the drafts. Scalable codecs
have=20
also been discussed. I don't like the word because people seem to use it

for different meanings. In one meaning it is simply the ability to use
any=20
rate, which is addressed above. In the other meaning, it means having=20
layers that can be stripped off. I believe this can be a useful feature=20
(which *is* listed in the current drafts), but mostly for conferencing,=20
rather than congestion. The point of layered codec for congestion in=20
particular is that a router in the middle can reduce the bit-rate to
adjust=20
to congestion. However, I see two problems here: 1) most routers does
not=20
(and should not have) knowledge of the codec and thus cannot change the=20
rate; and 2) layers are usually larger than a single byte per frame so=20
relying only on a layered approach might compromise the idea of having=20
small bit-rate increments. But as I said earlier, I'm not opposed to=20
layered codecs because they have other uses. This is something that
would=20
have to be carefully analysed during development.

> To be perfectly clear: This is NOT about enforcing TCP-friendliness or

> forcing the resulting codecs to conform to some other strict behavior=20
> under congestion. Due to the complexities of each coding scheme, how=20
> soon and by how much it can react to path congestion will be extremely

> codec-specific. But the idea is to recommend (or maybe even require)
for=20
> "Internet codecs" to react appropriately to path congestion when they
can.

OK, I may be wrong here, but my understanding is that how much the rate=20
should change depending on congestion is actually not that
codec-specific.=20
I get the impression that the tough part is figuring out what rate you
can=20
use without causing packets to be dropped. Once you figure this out, you

can just tell the codec to use that rate. On top of that, the codec may
be=20
able to do a bit better by providing a "service" to the other layers. On

example here (also included in the draft) is VBR, which means that the=20
application specifies an average rate and the codec decides how to vary
the=20
rate to achieve that (subject to some rate control constraints). Would=20
other such mechanisms be useful to deal with congestion?

Cheers,

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

From br@brianrosen.net  Wed Oct 21 07:04:56 2009
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From: Brian Rosen <br@brianrosen.net>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
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Please keep in mind that depending on the characteristics of the codec, you
don't want a lot of rate changes because it annoys the user.  I've had one
codec designer claim users weren't annoyed with relatively frequent changes
in rate (and thus audio quality), but I've never tested one that didn't.

Usually, if you encounter congestion, you can ratchet down, and then you
roughly never go back up.

When in doubt, run usability tests.  Easy to run, and gets you a solid
answer of what the codec ought to do.

Brian

On 10/21/09 9:57 AM, "Michael Ramalho (mramalho)" <mramalho@cisco.com>
wrote:

> All,
> 
> Just a thought for everyone. Prior to the last BoF I posted email that
> suggested that the codec should have an ability to change encoding rate
> as a function of some form of congestion intelligence (perhaps through
> an API, perhaps a hint at the RTP layer, perhaps on a congestion
> indication, etc.). That "should" was not necessarily a "must" because a
> given application may tell the codec to go full steam ahead anyway
> because what it knows about the transport environment. And I think that
> wording has been accommodated.
> 
> Consider Jean-Marc's words: "I get the impression that the tough part is
> figuring out what rate you can use without causing packets to be
> dropped."
> 
> Indeed.
> 
> Consider the case where the bottleneck link for your end-to-end codec
> application (where most of your packets are being dropped) is such that
> all the UDP traffic (e.g., your VoIP and other UDP applications) is a
> minority traffic. That is, the majority of traffic is TCP-friendly (TCP,
> SCTP, etc.). And that the bandwidth of this link is large relative to
> your VoIP stream. Now suppose that this link is someplace in one of the
> middle hops of your end-to-end flow.
> 
> That link is being "filled up" (and causing drops for your flow)
> according to the TCP dynamics of the other TCP traffic. Virtually
> NOTHING YOU DO ON YOUR FLOW will have a significant impact on your
> packet drop performance! Any bandwidth you "vacate" by reducing your
> rate becomes "absorbed" (in a sense) by the TCP flows sharing the
> remainder of the bandwidth.
> 
> If your application somehow (magically) knew this was the case, perhaps
> it would NOT reduce rate. Now that isn't TCP friendly. And some codec
> designs actually increase rate in those cases (by reducing rate all they
> can - but then adding FEC bandwidth to the lowest rate).
> 
> I think the best we can do here is to design in the knobs to have the
> codec change rate (or add FEC if that is a codec capability) to
> accommodate scenarios such as the above.
> 
> Michael Ramalho
> 
> PS - As a side note, if the bottleneck link had majority traffic
> consisting of similar VoIP flows - the correct behavior would be to
> reduce rate as TCP does. That is why the answer for a link dominated by
> ITSP traffic would be to reduce rate. And this is why the VoIP
> application ideally would benefit from "other intelligence" regarding
> congestion properties.
> 
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Wednesday, October 21, 2009 9:27 AM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> 
> Hi,
> 
> Lars Eggert wrote:
>> The Internet uses a distributed congestion control mechanism built
>> around the notion that when a flow observes packet loss, it is suppose
> 
>> to reduce its sending rate in some form over some time, in order to
> take 
>> load off a congested path. This behavior is something that I (as a
>> transport guy) would see as a key differentiator between an "Internet
>> codec" and a "telco codec", and I'd like to suggest that we reflect
> this 
>> in the charter in some form.
> 
> I totally agree that the bit-rate should adapt to network congestion.
> This 
> is why the requirements draft includes "Having the possibility to use
> smooth bit-rate changes with one byte/frame resolution". This means that
> 
> the codec can produce any rate that is needed to avoid congestion (not
> just 
> a few pre-selected ones). Now, we must be careful to assign issues to
> the 
> correct "layer" in the VoIP stack. I do not believe it is the codec's
> job 
> to *decide* which bit-rate to select. This should be left to another
> component of the application (the RTP stack?), which can then tell the
> codec what bit-rate to use.
> 
> If there are more features that the codec can add to help with
> congestion, 
> this is certainly something we can add to the drafts. Scalable codecs
> have 
> also been discussed. I don't like the word because people seem to use it
> 
> for different meanings. In one meaning it is simply the ability to use
> any 
> rate, which is addressed above. In the other meaning, it means having
> layers that can be stripped off. I believe this can be a useful feature
> (which *is* listed in the current drafts), but mostly for conferencing,
> rather than congestion. The point of layered codec for congestion in
> particular is that a router in the middle can reduce the bit-rate to
> adjust 
> to congestion. However, I see two problems here: 1) most routers does
> not 
> (and should not have) knowledge of the codec and thus cannot change the
> rate; and 2) layers are usually larger than a single byte per frame so
> relying only on a layered approach might compromise the idea of having
> small bit-rate increments. But as I said earlier, I'm not opposed to
> layered codecs because they have other uses. This is something that
> would 
> have to be carefully analysed during development.
> 
>> To be perfectly clear: This is NOT about enforcing TCP-friendliness or
> 
>> forcing the resulting codecs to conform to some other strict behavior
>> under congestion. Due to the complexities of each coding scheme, how
>> soon and by how much it can react to path congestion will be extremely
> 
>> codec-specific. But the idea is to recommend (or maybe even require)
> for 
>> "Internet codecs" to react appropriately to path congestion when they
> can.
> 
> OK, I may be wrong here, but my understanding is that how much the rate
> should change depending on congestion is actually not that
> codec-specific. 
> I get the impression that the tough part is figuring out what rate you
> can 
> use without causing packets to be dropped. Once you figure this out, you
> 
> can just tell the codec to use that rate. On top of that, the codec may
> be 
> able to do a bit better by providing a "service" to the other layers. On
> 
> example here (also included in the draft) is VBR, which means that the
> application specifies an average rate and the codec decides how to vary
> the 
> rate to achieve that (subject to some rate control constraints). Would
> other such mechanisms be useful to deal with congestion?
> 
> Cheers,
> 
> Jean-Marc
> _______________________________________________
> 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



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> I totally agree that the bit-rate should adapt to network congestion. This 
> is why the requirements draft includes "Having the possibility to use 
> smooth bit-rate changes with one byte/frame resolution". This means that 
> the codec can produce any rate that is needed to avoid congestion (not just 
> a few pre-selected ones). Now, we must be careful to assign issues to the 
> correct "layer" in the VoIP stack. I do not believe it is the codec's job 
> to *decide* which bit-rate to select. This should be left to another 
> component of the application (the RTP stack?), which can then tell the 
> codec what bit-rate to use.

Reading these statements, I would recommend to read the DCCP charter
http://www.ietf.org/dyn/wg/charter/dccp-charter.html and especially the
document http://www.ietf.org/rfc/rfc4828.txt

 Christian



From hoene@uni-tuebingen.de  Wed Oct 21 10:13:08 2009
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Hello Brian,

> Please keep in mind that depending on the characteristics of the codec,
you
> don't want a lot of rate changes because it annoys the user.  I've had one
> codec designer claim users weren't annoyed with relatively frequent
changes
> in rate (and thus audio quality), but I've never tested one that didn't.

Some years ago, we have done some tests with AMR. The distortion caused by a
change of the coding rate is roughly as large as the distortion caused by a
frame loss
(http://www.net.in.tum.de/fileadmin/bibtex/publications/papers/hoene_paper21
b.pdf). Consequently, frequent (multiple per second) mode switches are not
possible with AMR. However, frequent mode changes may be triggered by
transport protocols, which change their frame rates typically after a round
trip time.

The ITU-T G.718 codec was designed to allow frequent mode switched without
causing annoying artifacts. A similar requirement is missing in the current
draft-valin-codec-requirements document. I did not read any argument on why
it was not included.

> Usually, if you encounter congestion, you can ratchet down, and then you
> roughly never go back up.

TCP does increase linearly after an ECN notification or a frame loss event.
Depending on the selected algorithm, DCCP does behave similar.

 Christian



From jean-marc.valin@octasic.com  Wed Oct 21 11:04:07 2009
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Christian Hoene wrote:
> Some years ago, we have done some tests with AMR. The distortion caused by a
> change of the coding rate is roughly as large as the distortion caused by a
> frame loss
> (http://www.net.in.tum.de/fileadmin/bibtex/publications/papers/hoene_paper21
> b.pdf). Consequently, frequent (multiple per second) mode switches are not
> possible with AMR. However, frequent mode changes may be triggered by
> transport protocols, which change their frame rates typically after a round
> trip time.

 From what I've been told (haven't verified myself), the AMR-NB modes are 
more like a set of independent codecs put together with duct tape, rather 
than one codec with 8 rates. This is probably what explains the switching 
artefacts.

> The ITU-T G.718 codec was designed to allow frequent mode switched without
> causing annoying artifacts. A similar requirement is missing in the current
> draft-valin-codec-requirements document. I did not read any argument on why
> it was not included.

The requirements draft (both versions) currently includes the following text:

    "This means that the codecs have to be able to vary their output
    bit-rate dynamically (in real-time), without requiring an
    out-of-band signaling mechanism, and without causing audible
    artifacts at the bit-rate change boundaries."

I think "without causing audible artifacts" is a good description of what 
we want here. Or do you have something else to suggest?

	Jean-Marc


From gmaxwell@juniper.net  Wed Oct 21 11:38:28 2009
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So=97 I think this is really just an argument for codecs designed for the i=
nternet=85

Consider these examples:

CBR CELT (git) running at 48000, 256 sample frames, 21 bytes/frame (31500 b=
its per second which is a bit too low for good music quality with this smal=
l framesize)
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/21bytes.wav

CBR CELT (git) running at 48000, 256 sample frames, 43 bytes/frame (64500 b=
its per second)
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/43bytes.wav=20

CELT (git) 48000,256 ramping with from 43-21-43 in one byte per frame incre=
ments:
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/21-ramp1-43bytes=
.wav
(i.e. the frame sizes sizes are 43,42,41,40...22,21,22,23,24...)

Same ramping pattern, but sending 8 frames at each size:  i.e. 43,43,43,43,=
43,43,43,43,42,42,42....  or roughly 42ms per rate change and  one second p=
er complete cycle:=20
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/21-ramp8-43bytes=
.wav

Also,
Hard switching every other frame: 43,21,43,21,43,21....
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/21-mod2-43bytes.=
wav

And every 4th frame: (43,43,43,43,21,21,21,21,43...)
http://myrandomnode.dyndns.org:8080/~gmaxwell/celt_congest/21-mod4-43bytes.=
wav

While it would be very interesting to use a blind listening test to objecti=
vely characterize the impact of changing rates,  I believe that a casual li=
sten shows conclusively that for CELT a change in rates is nothing at all l=
ike a packet loss (or, perhaps you have very high expectations of what 50% =
loss sounds like!).

I would I expect reasonable rate control implementations to have some degre=
e of filtering to limit rate oscillations and using the codec in VBR mode a=
nd simply slewing the target is likely to be far less harmful, but I think =
that these simple examples show that aggressive adaptation is possible with=
out a ruinous effect to quality.

If anyone has profiles of what kind of rate adaptation patterns I could exp=
ect (i.e. from DCCP) I'll gladly generate test cases and can probably arran=
ge for a small/casual blind listening test.



________________________________________
From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Christia=
n Hoene [hoene@uni-tuebingen.de]
Sent: Wednesday, October 21, 2009 6:12 PM
To: 'Brian Rosen'
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Hello Brian,

> Please keep in mind that depending on the characteristics of the codec,
you
> don't want a lot of rate changes because it annoys the user.  I've had on=
e
> codec designer claim users weren't annoyed with relatively frequent
changes
> in rate (and thus audio quality), but I've never tested one that didn't.

Some years ago, we have done some tests with AMR. The distortion caused by =
a
change of the coding rate is roughly as large as the distortion caused by a
frame loss
(http://www.net.in.tum.de/fileadmin/bibtex/publications/papers/hoene_paper2=
1
b.pdf). Consequently, frequent (multiple per second) mode switches are not
possible with AMR. However, frequent mode changes may be triggered by
transport protocols, which change their frame rates typically after a round
trip time.

The ITU-T G.718 codec was designed to allow frequent mode switched without
causing annoying artifacts. A similar requirement is missing in the current
draft-valin-codec-requirements document. I did not read any argument on why
it was not included.

> Usually, if you encounter congestion, you can ratchet down, and then you
> roughly never go back up.

TCP does increase linearly after an ECN notification or a frame loss event.
Depending on the selected algorithm, DCCP does behave similar.

 Christian


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

From hoene@uni-tuebingen.de  Wed Oct 21 12:04:18 2009
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Hello Jean-Marc,

> > Some years ago, we have done some tests with AMR. The distortion caused
by a
> > change of the coding rate is roughly as large as the distortion caused
by a
> > frame loss
> >
(http://www.net.in.tum.de/fileadmin/bibtex/publications/papers/hoene_paper21
> > b.pdf). Consequently, frequent (multiple per second) mode switches are
not
> > possible with AMR. However, frequent mode changes may be triggered by
> > transport protocols, which change their frame rates typically after a
round
> > trip time.
> 
>  From what I've been told (haven't verified myself), the AMR-NB modes are
> more like a set of independent codecs put together with duct tape, rather
> than one codec with 8 rates. This is probably what explains the switching
> artefacts.

Yes. This might be the reason for this. The AMR standards were stating that
state switching is seamlessly. I never tested CELT.

> 
> > The ITU-T G.718 codec was designed to allow frequent mode switched
without
> > causing annoying artifacts. A similar requirement is missing in the
current
> > draft-valin-codec-requirements document. I did not read any argument on
why
> > it was not included.
> 
> The requirements draft (both versions) currently includes the following
text:
> 
>     "This means that the codecs have to be able to vary their output
>     bit-rate dynamically (in real-time), without requiring an
>     out-of-band signaling mechanism, and without causing audible
>     artifacts at the bit-rate change boundaries."
> 
> I think "without causing audible artifacts" is a good description of what
> we want here. Or do you have something else to suggest?

No, you are right. I missed it. Sorry.

 Christian



From hoene@uni-tuebingen.de  Wed Oct 21 12:06:39 2009
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Hello Gregory,

> While it would be very interesting to use a blind listening test to
> objectively characterize the impact of changing rates,  I believe that a
> casual listen shows conclusively that for CELT a change in rates is
nothing at
> all like a packet loss (or, perhaps you have very high expectations of
what
> 50% loss sounds like!).

Fine! Well done!

May I ask another question? What does happen if you switched between CELT
and SILK or any of the other codecs?

> I would I expect reasonable rate control implementations to have some
degree
> of filtering to limit rate oscillations 

Yes, they have. The DCCP algorithms change less frequent and less severe
than TCP.

> and using the codec in VBR mode and
> simply slewing the target is likely to be far less harmful, but I think
that
> these simple examples show that aggressive adaptation is possible without
a
> ruinous effect to quality.
> 
> If anyone has profiles of what kind of rate adaptation patterns I could
expect
> (i.e. from DCCP) I'll gladly generate test cases and can probably arrange
for
> a small/casual blind listening test.

Allow me address another issue. The RFC4828 document discusses the tradeoff
between bit rate and frame rate. To what extend is this tradeoff currently
considered in the requirements draft?

With best regards,

 Christian

 
> 
> 
> 
> ________________________________________
> From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of
Christian
> Hoene [hoene@uni-tuebingen.de]
> Sent: Wednesday, October 21, 2009 6:12 PM
> To: 'Brian Rosen'
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> 
> Hello Brian,
> 
> > Please keep in mind that depending on the characteristics of the codec,
> you
> > don't want a lot of rate changes because it annoys the user.  I've had
one
> > codec designer claim users weren't annoyed with relatively frequent
> changes
> > in rate (and thus audio quality), but I've never tested one that didn't.
> 
> Some years ago, we have done some tests with AMR. The distortion caused by
a
> change of the coding rate is roughly as large as the distortion caused by
a
> frame loss
>
(http://www.net.in.tum.de/fileadmin/bibtex/publications/papers/hoene_paper21
> b.pdf). Consequently, frequent (multiple per second) mode switches are not
> possible with AMR. However, frequent mode changes may be triggered by
> transport protocols, which change their frame rates typically after a
round
> trip time.
> 
> The ITU-T G.718 codec was designed to allow frequent mode switched without
> causing annoying artifacts. A similar requirement is missing in the
current
> draft-valin-codec-requirements document. I did not read any argument on
why
> it was not included.
> 
> > Usually, if you encounter congestion, you can ratchet down, and then you
> > roughly never go back up.
> 
> TCP does increase linearly after an ECN notification or a frame loss
event.
> Depending on the selected algorithm, DCCP does behave similar.
> 
>  Christian
> 
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


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On 10/20/09 5:03 PM, rontlv wrote:
> Hi,
> I did not have time to review the charter proposal being on vacation.
> 
> I would like to bring my major concern about 
> " The goal of this group is to develop one or more codecs" . My view is that
> the one of the major reason for lack of deployment of wideband is that there
> are too many wideband codecs.
> For example in high quality video conferencing you may need to support G.719
> and flavors of AAC full band codecs to interoperate at full band with all
> the current products.

At the Stockholm BoF, people raised concerns about saying that the WG
would limit itself beforehand to defining one and only one codec ("too
restrictive!"). They also raised concerns about the WG becoming a codec
factory that would produce lots of codecs ("too permissive!"). The
current text attemps to address both concerns by expressing a preference
for one codec while recognizing that it is possible the WG would produce
more than one codec (albeit a very small number of codecs).

> I also find it strange that the timeline has one date for all codecs that
> will be developed since I do not see any synchronization process between all
> the codecs that will be approved if a WG is formed.
>
> I think that having the right to do multiple codecs as part of the charter
> will cause the group to approve all proposed codecs probably rubberstamping
> them instead of going to collaboration on one good codec. I think that the
> initial charter should have one codec and if later there will be a need
> there should be a re-charter.

That seems acceptable.

> A minor comment is on 
> "Because the Internet Standards Process defines an open, transparent process
> for technology standardization, a wide range of codec experts have committed
> to actively working on such codecs within the IETF.
> The Codec WG provides a structured forum for these efforts."
> I am not sure why this text is part of the charter. 

That text is true, but you are right that it is not properly part of the
problem statement.

> Being a little sarcastic about the commitment, I did not see the flexibility
> of the experts when they needed to devote time for the IETF meeting.

Your sarcasm is appreciated.

Peter

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On 10/21/09 1:34 PM, Peter Saint-Andre wrote:
> On 10/20/09 5:03 PM, rontlv wrote:
>> Hi,
>> I did not have time to review the charter proposal being on vacation.
> 
>> I would like to bring my major concern about 
>> " The goal of this group is to develop one or more codecs" . My view is that
>> the one of the major reason for lack of deployment of wideband is that there
>> are too many wideband codecs.
>> For example in high quality video conferencing you may need to support G.719
>> and flavors of AAC full band codecs to interoperate at full band with all
>> the current products.
> 
> At the Stockholm BoF, people raised concerns about saying that the WG
> would limit itself beforehand to defining one and only one codec ("too
> restrictive!"). They also raised concerns about the WG becoming a codec
> factory that would produce lots of codecs ("too permissive!"). The
> current text attemps to address both concerns by expressing a preference
> for one codec while recognizing that it is possible the WG would produce
> more than one codec (albeit a very small number of codecs).
> 
>> I also find it strange that the timeline has one date for all codecs that
>> will be developed since I do not see any synchronization process between all
>> the codecs that will be approved if a WG is formed.
> 
>> I think that having the right to do multiple codecs as part of the charter
>> will cause the group to approve all proposed codecs probably rubberstamping
>> them instead of going to collaboration on one good codec. I think that the
>> initial charter should have one codec and if later there will be a need
>> there should be a re-charter.
> 
> That seems acceptable.

Well, the charter currently includes the following text:

   The group might produce only one codec or it might produce multiple
   codecs; however, to reduce confusion in the marketplace it shall
   endeavor not to function as a "codec factory" that produces a large
   number of codecs.

Perhaps that could be modified to more clearly express a strong
preference for producing one codec:

   The group might produce only one codec or it might produce multiple
   codecs; however, to reduce confusion in the marketplace it shall
   endeavor to produce only one codec if possible, and in any case shall
   not function as a "codec factory" that produces a large number of
   codecs.

Peter

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On 10/21/09 5:45 AM, Christian Hoene wrote:
> Dear Lars Eggert,
> 
>> The Internet uses a distributed congestion control mechanism built
>> around the notion that when a flow observes packet loss, it is suppose
>> to reduce its sending rate in some form over some time, in order to
>> take load off a congested path. This behavior is something that I (as
>> a transport guy) would see as a key differentiator between an
>> "Internet codec" and a "telco codec", and I'd like to suggest that we
>> reflect this in the charter in some form.
>>
>> We can certainly argue about how useful this behavior would be in a
>> low-bitrate voice codec. But it's my understanding that this effort
>> isn't really targeting those, but is instead looking at wideband
>> codecs - maybe ones using substantial levels of redundancy in the
>> encoding - that can consume some noticeable amount of bandwidth. For
>> those sorts of voice codecs, and even more so for video codecs, which
>> have been brought up as the next logical thing for a potential CODEC
>> WG to target, some sort of congestion-responsiveness is important when
>> they are sharing path capacity over the Internet.
>>
>> See for example
> http://tools.ietf.org/html/draft-westerlund-avt-ecn-for-rtp
>>   for ongoing work related to exposing congestion information to media
>> streams earlier than through packet loss (i.e., using ECN).
>>
>> To be perfectly clear: This is NOT about enforcing TCP-friendliness or
>> forcing the resulting codecs to conform to some other strict behavior
>> under congestion. Due to the complexities of each coding scheme, how
>> soon and by how much it can react to path congestion will be extremely
>> codec-specific. But the idea is to recommend (or maybe even require)
>> for "Internet codecs" to react appropriately to path congestion when
>> they can.
> 
> Since months I am raising this point on this mailing list. I suggested to
> establish a procedure on how transport and codec expert should work together
> to address these issues. 
> 
> However, based on the comments on this mailing list, I have the impression
> that this is not wanted. I just remember some statements such as that we are
> "only interested in a narrow area." 
> 
> I do have the impression that it is not intended to address the fundamental
> problems of frequently bad VoIP quality or to enhance the Quality of
> Experience - which would require to consider transport issues - but only in
> making codecs. Hopefully, my observation is wrong.

It's not clear to me why the Codec WG would focus on transport issues.
Certainly it should coordinate with transport WGs (AVT, DCCP, etc.),
just as it should coordinate with WGs and experts from RAI, Apps,
Security, architecture (IAB), etc.

Do you think that the charter text needs to be updated to express the
fact that the Codec WG would need to coordinate with other groups within
the IETF? Typically that is part of normal cross-area review, but
perhaps the extraordinary impact on the Codec WG of considerations that
are properly the focus of other groups necessitates more intensive
interaction here. I'm not convinced of that yet, but adding some text
about this in the charter might be appropriate if we think it is needed.

Peter

- --
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From jean-marc.valin@octasic.com  Wed Oct 21 13:16:59 2009
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Christian Hoene wrote:
> May I ask another question? What does happen if you switched between CELT
> and SILK or any of the other codecs?

(my answer is based on CELT and SILK but it would be about the same for any 
two codecs that are sufficiently different)
If done in a simple (dumb) way, it would insert a glitch that would be 
comparable to a lost packet. If done a bit smarter, it would merely require 
  a slight increase in bitrate at the boundary frame along with a careful 
overlap of the audio. If the codecs were designed jointly, then it *may* 
also be possible make the bit-rate increase pretty small (or zero).

	Jean-Marc

From Borilin@spiritdsp.com  Wed Oct 21 13:20:43 2009
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sorry i am not getting the idea WHY you should switch codecs during the =
call?
i understand the idea os changing bitrate on-the-fly, but why =
renegotiate and change codec on the fly? only in a rare case of call =
transfeerring to a different device- but in this case you have a delay =
of switching which makes such change unnoticable anyway.
am i wrong?
=20

________________________________

=EF=D4: codec-bounces@ietf.org =CF=D4 =C9=CD=C5=CE=C9 Jean-Marc Valin
=EF=D4=D0=D2=C1=D7=CC=C5=CE=CF: =FE=D4, 22.10.2009 0:17
=EB=CF=CD=D5: Christian Hoene
=EB=CF=D0=C9=D1: codec@ietf.org
=F4=C5=CD=C1: Re: [codec] updated charter proposal



Christian Hoene wrote:
> May I ask another question? What does happen if you switched between =
CELT
> and SILK or any of the other codecs?

(my answer is based on CELT and SILK but it would be about the same for =
any
two codecs that are sufficiently different)
If done in a simple (dumb) way, it would insert a glitch that would be
comparable to a lost packet. If done a bit smarter, it would merely =
require
  a slight increase in bitrate at the boundary frame along with a =
careful
overlap of the audio. If the codecs were designed jointly, then it *may*
also be possible make the bit-rate increase pretty small (or zero).

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



From Jean-Marc.Valin@USherbrooke.ca  Wed Oct 21 13:29:02 2009
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Quoting Slava Borilin <Borilin@spiritdsp.com>:

> sorry i am not getting the idea WHY you should switch codecs during the call?
> i understand the idea os changing bitrate on-the-fly, but why renegotiate and
> change codec on the fly? only in a rare case of call transfeerring to a
> different device- but in this case you have a delay of switching which makes
> such change unnoticable anyway.
> am i wrong?

You are perfectly right here, I was just answering the question from a
theoretical perspective. At this point, I do not see a reason for changing
codec on the fly, or at least not without some kind of exceptional event
happening (e.g. call transfer) -- in which case a single glitch wouldn't be a
problem.

Cheers,

   Jean-Marc

>
> ________________________________
>
> : codec-bounces@ietf.org   Jean-Marc Valin
> : , 22.10.2009 0:17
> : Christian Hoene
> : codec@ietf.org
> : Re: [codec] updated charter proposal
>
>
>
> Christian Hoene wrote:
> > May I ask another question? What does happen if you switched between CELT
> > and SILK or any of the other codecs?
>
> (my answer is based on CELT and SILK but it would be about the same for any
> two codecs that are sufficiently different)
> If done in a simple (dumb) way, it would insert a glitch that would be
> comparable to a lost packet. If done a bit smarter, it would merely require
>   a slight increase in bitrate at the boundary frame along with a careful
> overlap of the audio. If the codecs were designed jointly, then it *may*
> also be possible make the bit-rate increase pretty small (or zero).
>
>         Jean-Marc
> _______________________________________________
> 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
>




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On 20 Oct 2009, at 22:42, Peter Saint-Andre wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Per recent list discussion...
>
> ######
>
> Proposed Charter for Codec WG
> Last Updated: 2009-10-20
...
> 4. Ensuring interoperability with Internet signalling technologies  
> such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
> signalling-dependent codecs are out of scope


Most recent codecs are dependent on some form of out-of-band  
signalling to negotiate various options and parameters. Prohibiting  
any dependence on the signalling seems extreme. You might wish to say  
something like:


4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP) by  
defining the parameters that must be negotiated to support the codec;  
the result should not, however, depend on the details of any  
particular signalling protocol

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




From jean-marc.valin@octasic.com  Wed Oct 21 13:58:11 2009
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>> 4. Ensuring interoperability with Internet signalling technologies such
>> as Session Initiation Protocol (SIP), Session Description Protocol
>> (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
>> signalling-dependent codecs are out of scope
> 
> Most recent codecs are dependent on some form of out-of-band signalling 
> to negotiate various options and parameters. Prohibiting any dependence 
> on the signalling seems extreme. You might wish to say something like:

IIRC, the intent was just to clarify that the codec wouldn't be tied to SIP 
  or XMPP for example. However, your point about out-of-band signalling is 
actually a valid one. We will probably need some amount of out-of-band 
signalling, but I think it should be simple enough and it should fit well 
within SDP (this should probably go in the requirements draft).

> 4. Ensuring interoperability with Internet signalling technologies such
> as Session Initiation Protocol (SIP), Session Description Protocol
> (SDP), and Extensible Messaging and Presence Protocol (XMPP) by defining 
> the parameters that must be negotiated to support the codec; the result 
> should not, however, depend on the details of any particular signalling 
> protocol

Sounds good.

	Jean-Marc

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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: [codec] HA:  updated charter proposal
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You could change codecs "on the fly" using in-band signaling, the same way
as (some have argued) AMR-WB does.  This is mostly a question of
terminology: when you switch between two distinct coding technologies that
are a) specified in the same standard, b) mandatory to implement, and c)
in-band signaled, do you then switch codecs or modes of the same codec?

(Whether a specification like this makes a lot of sense in our environment,
is, IMHO, very questionable.  A lot of codec standards of this type exist,
but they are often at least in part the result of a political process and
not necessarily only motivated by objectively justifiable technical design
choices.)

(OTOH, I personally prefer a clean single-mechanism spec, over a monster
spec with everything mandatory, over a pick-and-choose spec---and the latte=
r
over lots of individual specs.  That is, of course, on the grounds of
interoperability.)
Stephan


On 10/21/09 1:29 PM, "Jean-Marc Valin" <Jean-Marc.Valin@USherbrooke.ca>
wrote:

> Quoting Slava Borilin <Borilin@spiritdsp.com>:
>=20
>> sorry i am not getting the idea WHY you should switch codecs during the =
call?
>> i understand the idea os changing bitrate on-the-fly, but why renegotiat=
e and
>> change codec on the fly? only in a rare case of call transfeerring to a
>> different device- but in this case you have a delay of switching which m=
akes
>> such change unnoticable anyway.
>> am i wrong?
>=20
> You are perfectly right here, I was just answering the question from a
> theoretical perspective. At this point, I do not see a reason for changin=
g
> codec on the fly, or at least not without some kind of exceptional event
> happening (e.g. call transfer) -- in which case a single glitch wouldn't =
be a
> problem.
>=20
> Cheers,
>=20
>    Jean-Marc
>=20
>>=20
>> ________________________________
>>=20
>> =C3=AF=C3=94: codec-bounces@ietf.org =C3=8F=C3=94 =C3=89=C3=8D=C3=85=C3=8E=C3=89 Jean-Marc Valin
>> =C3=AF=C3=94=C3=90=C3=92=C3=81=C3=97=C3=8C=C3=85=C3=8E=C3=8F: =C3=BE=C3=94, 22.10.2009 0:17
>> =C3=AB=C3=8F=C3=8D=C3=95: Christian Hoene
>> =C3=AB=C3=8F=C3=90=C3=89=C3=91: codec@ietf.org
>> =C3=B4=C3=85=C3=8D=C3=81: Re: [codec] updated charter proposal
>>=20
>>=20
>>=20
>> Christian Hoene wrote:
>>> May I ask another question? What does happen if you switched between CE=
LT
>>> and SILK or any of the other codecs?
>>=20
>> (my answer is based on CELT and SILK but it would be about the same for =
any
>> two codecs that are sufficiently different)
>> If done in a simple (dumb) way, it would insert a glitch that would be
>> comparable to a lost packet. If done a bit smarter, it would merely requ=
ire
>>   a slight increase in bitrate at the boundary frame along with a carefu=
l
>> overlap of the audio. If the codecs were designed jointly, then it *may*
>> also be possible make the bit-rate increase pretty small (or zero).
>>=20
>>         Jean-Marc
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>=20
>>=20
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>=20
>=20
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec



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On 21 Oct 2009, at 21:00, Peter Saint-Andre wrote:
...
> It's not clear to me why the Codec WG would focus on transport  
> issues. Certainly it should coordinate with transport WGs (AVT,  
> DCCP, etc.), just as it should coordinate with WGs and experts from  
> RAI, Apps, Security, architecture (IAB), etc.
>
> Do you think that the charter text needs to be updated to express  
> the fact that the Codec WG would need to coordinate with other  
> groups within the IETF? Typically that is part of normal cross-area  
> review, but perhaps the extraordinary impact on the Codec WG of  
> considerations that are properly the focus of other groups  
> necessitates more intensive interaction here. I'm not convinced of  
> that yet, but adding some text about this in the charter might be  
> appropriate if we think it is needed.


That sort of text is relatively common in charters (for example, the  
AVT charter says "in collaboration with the MPLS and ROHC WGs, to  
develop a solution for header compression of RTP across MPLS networks  
that avoid decompression and compression at each MPLS node."), and  
would seem to be appropriate here. The key points would seem to be to  
work with AVT and TSVWG to understand the degree of rate adaptation  
desirable, and to attempt to reflect that in the design of a codec  
that can adjust its transmission in a way that minimises disruption to  
the audio. The technical details of how that might be done - in terms  
of the congestion control or the codec - are likely out of scope for  
the charter.

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




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On 10/21/09 2:58 PM, Jean-Marc Valin wrote:
>>> 4. Ensuring interoperability with Internet signalling technologies such
>>> as Session Initiation Protocol (SIP), Session Description Protocol
>>> (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
>>> signalling-dependent codecs are out of scope
>>
>> Most recent codecs are dependent on some form of out-of-band
>> signalling to negotiate various options and parameters. Prohibiting
>> any dependence on the signalling seems extreme. You might wish to say
>> something like:
> 
> IIRC, the intent was just to clarify that the codec wouldn't be tied to
> SIP  or XMPP for example. However, your point about out-of-band
> signalling is actually a valid one. We will probably need some amount of
> out-of-band signalling, but I think it should be simple enough and it
> should fit well within SDP (this should probably go in the requirements
> draft).
> 
>> 4. Ensuring interoperability with Internet signalling technologies such
>> as Session Initiation Protocol (SIP), Session Description Protocol
>> (SDP), and Extensible Messaging and Presence Protocol (XMPP) by
>> defining the parameters that must be negotiated to support the codec;
>> the result should not, however, depend on the details of any
>> particular signalling protocol
> 
> Sounds good.

Yes, that's better, thanks.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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On 10/21/09 3:04 PM, Colin Perkins wrote:
> On 21 Oct 2009, at 21:00, Peter Saint-Andre wrote:
> ...
>> It's not clear to me why the Codec WG would focus on transport issues.
>> Certainly it should coordinate with transport WGs (AVT, DCCP, etc.),
>> just as it should coordinate with WGs and experts from RAI, Apps,
>> Security, architecture (IAB), etc.
>>
>> Do you think that the charter text needs to be updated to express the
>> fact that the Codec WG would need to coordinate with other groups
>> within the IETF? Typically that is part of normal cross-area review,
>> but perhaps the extraordinary impact on the Codec WG of considerations
>> that are properly the focus of other groups necessitates more
>> intensive interaction here. I'm not convinced of that yet, but adding
>> some text about this in the charter might be appropriate if we think
>> it is needed.
> 
> 
> That sort of text is relatively common in charters (for example, the AVT
> charter says "in collaboration with the MPLS and ROHC WGs, to develop a
> solution for header compression of RTP across MPLS networks that avoid
> decompression and compression at each MPLS node."), and would seem to be
> appropriate here. The key points would seem to be to work with AVT and
> TSVWG to understand the degree of rate adaptation desirable, and to
> attempt to reflect that in the design of a codec that can adjust its
> transmission in a way that minimises disruption to the audio. The
> technical details of how that might be done - in terms of the congestion
> control or the codec - are likely out of scope for the charter.

That seems reasonable.

To capture all the integration points that have been mentioned on the
list, we could include in the charter text such as this:

***

In completing its work, the group should collaborate with other IETF
working groups to complete particular tasks.  These might include but
would not be limited to the following:

- - Within the AVT WG, define the codec's payload format for use with the
  Real-time Transport Protocol (RTP).

- - Collaborate with the AVT WG and TSV WG to understand the degree of
  rate adaptation desirable, and to reflect that understanding in the
  design of a codec that can adjust its transmission in a way that
  minimizes disruption to the audio.

- - Collaborate with the SIPCORE WG and XMPP WG to ensure that information
  about the codec can be easily represented at the signalling layer.

- - Collaborate with the AVT WG and DCCP WG to identify important
  aspects of packet transmission over the Internet.

***

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From eburger@alum.mit.edu  Wed Oct 21 15:58:59 2009
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Here is a start:
http://www.comsoc.org/livepubs/ci1/public/2009/oct/index.html

Still need to log in. I'll see if I can get it freed. In the mean  
time, if anyone wants to join...

On Oct 20, 2009, at 4:15 PM, Eric Burger wrote:

> The most recent issue (October 2009, V. 47 n. 10)  of IEEE  
> Communications Magazine has a special section of interest to us here  
> on Speech Coding. Ask a friend if you do not have access to the IEEE  
> Xplore library or a hard copy.


From ingemar.s.johansson@ericsson.com  Thu Oct 22 00:22:00 2009
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From: "Ingemar Johansson S" <ingemar.s.johansson@ericsson.com>
To: "Christian Hoene" <hoene@uni-tuebingen.de>, "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca>
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Subject: Re: [codec] updated charter proposal
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Hi Cristian, Jean-Marc


Some comments to the comments on AMR

I have not had time to read the report in full but I reacted on he
comment below (figure 5 in the report).=20

AMR rate swithing _is_ seamless. As far as I can recall rate switching
was part of the codec characterization in 3GPP, in addition, rate
control in circuit switched networks (which is faster than the rate
control in packet switched networks will ever be) would be impossible
without seamless switching. So I am a bit surprised that you get these
results.=20
Did you do a listening test as well ?, I wonder if there is a flaw in
the PESQ model.. What kind of input stimuli did you use ?. I actually
had to make a testrun where I switched between AMR102 and AMR122 every
20ms and I did not hear any swithing artifacts. I even asked another
listener at the office (with better ears than mine) to listen and he
wasn't able to hear any switching artifacts either.

And finally=20
AMR is _one_ codec, not a selection of 8 codecs. The modes share the
same framework but of course the quanitizer resolution (LSP, adaptive
CB, fixed CB) depends on the bitrate + that there are a few extra
features added for a few modes. =20
Some of the AMR modes are compatible with fixed rate codecs, AMR122 is
compatible with the EFR codec, AMR67 is used in a Japanese cellular
system, still AMR is one codec.

Regards
/Ingemar

> -----Original Message-----
> From: Christian Hoene [mailto:hoene@uni-tuebingen.de]=20
> Sent: den 21 oktober 2009 19:13
> To: 'Brian Rosen'
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Hello Brian,
>=20
> > Please keep in mind that depending on the characteristics of the=20
> > codec,
> you
> > don't want a lot of rate changes because it annoys the=20
> user.  I've had=20
> > one codec designer claim users weren't annoyed with relatively=20
> > frequent
> changes
> > in rate (and thus audio quality), but I've never tested one=20
> that didn't.
>=20
> Some years ago, we have done some tests with AMR. The=20
> distortion caused by a change of the coding rate is roughly=20
> as large as the distortion caused by a frame loss
> (http://www.net.in.tum.de/fileadmin/bibtex/publications/papers
> /hoene_paper21
> b.pdf). Consequently, frequent (multiple per second) mode=20
> switches are not possible with AMR. However, frequent mode=20
> changes may be triggered by transport protocols, which change=20
> their frame rates typically after a round trip time.
>=20
> The ITU-T G.718 codec was designed to allow frequent mode=20
> switched without causing annoying artifacts. A similar=20
> requirement is missing in the current=20
> draft-valin-codec-requirements document. I did not read any=20
> argument on why it was not included.
>=20
> > Usually, if you encounter congestion, you can ratchet down,=20
> and then=20
> > you roughly never go back up.
>=20
> TCP does increase linearly after an ECN notification or a=20
> frame loss event.
> Depending on the selected algorithm, DCCP does behave similar.
>=20
>  Christian
>=20
>=20
>=20
>=20

From lars.eggert@nokia.com  Thu Oct 22 00:44:18 2009
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Hi,

On 2009-10-21, at 16:27, Jean-Marc Valin wrote:
> I totally agree that the bit-rate should adapt to network  
> congestion. This
> is why the requirements draft includes "Having the possibility to use
> smooth bit-rate changes with one byte/frame resolution". This means  
> that
> the codec can produce any rate that is needed to avoid congestion  
> (not just
> a few pre-selected ones). Now, we must be careful to assign issues  
> to the
> correct "layer" in the VoIP stack. I do not believe it is the  
> codec's job
> to *decide* which bit-rate to select. This should be left to another
> component of the application (the RTP stack?), which can then tell the
> codec what bit-rate to use.

RTP doesn't do congestion control either. DCCP would, but nobody's  
using it currently, mostly due to NAT traversal issues and lack of  
implementations.

Also, what we've learned from media over DCCP is that if you don't  
involve the codec in some form, performance/quality will be bad. Some  
codecs need time to adjust the rate. Some can only adjust it in pretty  
coarse steps. Etc. If you let DCCP adjust the rate blindly, without  
involving the codec in some form, you might get very good TCP  
friendliness, but maybe quality will suffer.

(And there is always the ramping up part - congestion control will  
never attempt to probe for more capacity unless the codec generates  
more data, and why would the codec do that if it's converged to  
transmit at some lower rate.)

> The point of layered codec for congestion in
> particular is that a router in the middle can reduce the bit-rate to  
> adjust
> to congestion. However, I see two problems here: 1) most routers  
> does not
> (and should not have) knowledge of the codec and thus cannot change  
> the
> rate; and 2) layers are usually larger than a single byte per frame so
> relying only on a layered approach might compromise the idea of having
> small bit-rate increments.

By no means should routers have to understand codec formats! Routers  
drop or mark IP packets. It is up to the application sending those IP  
packets to react appropriately to the marks or losses. In other words,  
a layered codec will see losses/marks to packets sent on all layers,  
and will then need to figure out whether to strip off a layer or not  
in response to whatever congestion level those losses/marks indicate.

>> To be perfectly clear: This is NOT about enforcing TCP-friendliness  
>> or
>> forcing the resulting codecs to conform to some other strict behavior
>> under congestion. Due to the complexities of each coding scheme, how
>> soon and by how much it can react to path congestion will be  
>> extremely
>> codec-specific. But the idea is to recommend (or maybe even  
>> require) for
>> "Internet codecs" to react appropriately to path congestion when  
>> they can.
>
> OK, I may be wrong here, but my understanding is that how much the  
> rate
> should change depending on congestion is actually not that codec- 
> specific.

It *can* be non-codec-specific, see DCCP for example. But quality will  
suffer IMO.

> I get the impression that the tough part is figuring out what rate  
> you can
> use without causing packets to be dropped.

Most Internet congestion control is based on *creating* losses/marks  
to determine capacity. People have tried to create congestion control  
schemes that avoid causing losses (delay-based schemed, for example),  
but they are perceived to be more fragile.

> Once you figure this out, you
> can just tell the codec to use that rate.

It's not that easy, because rates in the Internet are never stable -  
the rate computation is usually clocked by the RTT. You can obviously  
dampen this somehow, but you can never say "now I have found a rate  
that I can safely use forever."

> On top of that, the codec may be
> able to do a bit better by providing a "service" to the other  
> layers. On
> example here (also included in the draft) is VBR, which means that the
> application specifies an average rate and the codec decides how to  
> vary the
> rate to achieve that (subject to some rate control constraints). Would
> other such mechanisms be useful to deal with congestion?

I'm not sure this would work, because congestion control will  
basically change the VBR input parameter continuously.

Lars
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Subject: Re: [codec] updated charter proposal
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Hello Ingemar,

I made the tests some years ago. The raw data and the measurement =
scripts
are already on backup CDs. I will revive it. Please allow me some time
before I can send it to you.=20

With best regards,

 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: Ingemar Johansson S [mailto:ingemar.s.johansson@ericsson.com]
> Sent: Thursday, October 22, 2009 9:19 AM
> To: Christian Hoene; Jean-Marc Valin
> Cc: codec@ietf.org; Brian Rosen; Ingemar Johansson S
> Subject: RE: [codec] updated charter proposal
>=20
> Hi Cristian, Jean-Marc
>=20
>=20
> Some comments to the comments on AMR
>=20
> I have not had time to read the report in full but I reacted on he
> comment below (figure 5 in the report).
>=20
> AMR rate swithing _is_ seamless. As far as I can recall rate switching
> was part of the codec characterization in 3GPP, in addition, rate
> control in circuit switched networks (which is faster than the rate
> control in packet switched networks will ever be) would be impossible
> without seamless switching. So I am a bit surprised that you get these
> results.
> Did you do a listening test as well ?, I wonder if there is a flaw in
> the PESQ model.. What kind of input stimuli did you use ?. I actually
> had to make a testrun where I switched between AMR102 and AMR122 every
> 20ms and I did not hear any swithing artifacts. I even asked another
> listener at the office (with better ears than mine) to listen and he
> wasn't able to hear any switching artifacts either.
>=20
> And finally
> AMR is _one_ codec, not a selection of 8 codecs. The modes share the
> same framework but of course the quanitizer resolution (LSP, adaptive
> CB, fixed CB) depends on the bitrate + that there are a few extra
> features added for a few modes.
> Some of the AMR modes are compatible with fixed rate codecs, AMR122 is
> compatible with the EFR codec, AMR67 is used in a Japanese cellular
> system, still AMR is one codec.
>=20
> Regards
> /Ingemar
>=20
> > -----Original Message-----
> > From: Christian Hoene [mailto:hoene@uni-tuebingen.de]
> > Sent: den 21 oktober 2009 19:13
> > To: 'Brian Rosen'
> > Cc: codec@ietf.org
> > Subject: Re: [codec] updated charter proposal
> >
> > Hello Brian,
> >
> > > Please keep in mind that depending on the characteristics of the
> > > codec,
> > you
> > > don't want a lot of rate changes because it annoys the
> > user.  I've had
> > > one codec designer claim users weren't annoyed with relatively
> > > frequent
> > changes
> > > in rate (and thus audio quality), but I've never tested one
> > that didn't.
> >
> > Some years ago, we have done some tests with AMR. The
> > distortion caused by a change of the coding rate is roughly
> > as large as the distortion caused by a frame loss
> > (http://www.net.in.tum.de/fileadmin/bibtex/publications/papers
> > /hoene_paper21
> > b.pdf). Consequently, frequent (multiple per second) mode
> > switches are not possible with AMR. However, frequent mode
> > changes may be triggered by transport protocols, which change
> > their frame rates typically after a round trip time.
> >
> > The ITU-T G.718 codec was designed to allow frequent mode
> > switched without causing annoying artifacts. A similar
> > requirement is missing in the current
> > draft-valin-codec-requirements document. I did not read any
> > argument on why it was not included.
> >
> > > Usually, if you encounter congestion, you can ratchet down,
> > and then
> > > you roughly never go back up.
> >
> > TCP does increase linearly after an ECN notification or a
> > frame loss event.
> > Depending on the selected algorithm, DCCP does behave similar.
> >
> >  Christian
> >
> >
> >
> >


From lars.eggert@nokia.com  Thu Oct 22 00:52:51 2009
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Subject: Re: [codec] updated charter proposal
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Hi,

On 2009-10-21, at 23:00, Peter Saint-Andre wrote:
> It's not clear to me why the Codec WG would focus on transport issues.
> Certainly it should coordinate with transport WGs (AVT, DCCP, etc.),
> just as it should coordinate with WGs and experts from RAI, Apps,
> Security, architecture (IAB), etc.

I'm not saying it should focus on transport issues. I'm saying that  
because this effort is trying to spec "Internet codecs", these codecs  
should be better integrated with Internet transports than those done  
elsewhere, esp. when it comes to congestion response. The idea is that  
through that integration (or at least awareness) it should be possible  
to arrive at better quality under loss while still doing an  
appropriate congestion response.

Lars



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Hi,

On 2009-10-22, at 1:31, Peter Saint-Andre wrote:
> - - Collaborate with the AVT WG and TSV WG to understand the degree of
>  rate adaptation desirable, and to reflect that understanding in the
>  design of a codec that can adjust its transmission in a way that
>  minimizes disruption to the audio.
...
> - - Collaborate with the AVT WG and DCCP WG to identify important
>  aspects of packet transmission over the Internet.

Minor nit. I'd change both TSV WG and DCCP to TSV area, since it's not  
clear which WG this would land in (and there is a chance that DCCP  
will close soon).

Lars
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From swmike@swm.pp.se  Thu Oct 22 01:01:06 2009
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From: Mikael Abrahamsson <swmike@swm.pp.se>
To: "codec@ietf.org" <codec@ietf.org>
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Subject: Re: [codec] updated charter proposal
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On Thu, 22 Oct 2009, Lars Eggert wrote:

> Most Internet congestion control is based on *creating* losses/marks to 
> determine capacity. People have tried to create congestion control 
> schemes that avoid causing losses (delay-based schemed, for example), 
> but they are perceived to be more fragile.

I don't know exactly how the information should be handled at each layer, 
but taking into account jitter, delay, packet loss, ECN-bits set etc, the 
congestion signals seen by TCP should be available via similar mechanisms 
to a sound transport solution. I initially thought that the near constant 
rate over time nature of a sound transport stream (compared to the often 
bursty nature of TCP) would mean that it should "stand its ground", but I 
have now changed my opinion and I do think that the bitrate should be 
lowered when adverse network conditions are seen, even if it's only done 
in a few steps (low/mid/high or something like that).

This would mean that when links are filled with just sound streams, they 
can adapt when this link goes full by adapting the rate, and in case of 
congestion where there is a lot of TCP streams, at least we can play a 
little friendly by lowering the bitrate, even though it wont make much of 
a difference overall.

Also, it might make a big difference for a link that has a lot of voice 
traffic, if the packet per second rate could be decreased, so if the 
default is one packet per 20ms, going to one packet per 40ms halves the 
pps and lowers the bandwidth used because of less packet overhead (on a 
DSL link where it might be VoiceinRTPinIPinEthernetinATM, the overhead 
might well be hundreds of percent). So if the initial 50pps meant 16 bytes 
of voice per packet (which might in worst case conditions mean 3 ATM cells 
per packet), increasing this to 32 bytes at 25pps will decrease the 
bandwidth seen on the ATM layer from 63 kilobits/s to 32 kilobits/s (if 
the new packet still fits in 3 cells, otherwise it'll be 42 kilobit/s (4 
cells per packet)).

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

From hoene@uni-tuebingen.de  Thu Oct 22 04:37:12 2009
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Dear Mr. Colin Perkins,

> > 4. Ensuring interoperability with Internet signalling technologies
> > such
> > as Session Initiation Protocol (SIP), Session Description Protocol
> > (SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
> > signalling-dependent codecs are out of scope
> 
> 
> Most recent codecs are dependent on some form of out-of-band
> signalling to negotiate various options and parameters. Prohibiting
> any dependence on the signalling seems extreme. 

Some time ago, Alexander Chemeris was raising this issue.
http://www.ietf.org/mail-archive/web/codec/current/msg00508.html
He was complaining that there is a violation between transport and signaling
"layer" (or better plane) for some modern codecs - especially if the
signaling protocol has to store states about the negotiated coding mode.
This might causes problems if the signaling engine must renegotiate
transmission connections or if connections are to be forwarded. He was
telling me that the signaling part of a call forwarding engine must include
codec specific extension and that this should be avoided.

I have to admit that I did not fully understood his use-case but having a
clear split between signaling and transport states seems to be a good
approach to reduce system complexity.

With best regards,

 Christian

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


From csp@csperkins.org  Thu Oct 22 05:26:45 2009
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On 22 Oct 2009, at 12:37, Christian Hoene wrote:
>>> 4. Ensuring interoperability with Internet signalling technologies
>>> such as Session Initiation Protocol (SIP), Session Description  
>>> Protocol (SDP), and Extensible Messaging and Presence Protocol  
>>> (XMPP); however, signalling-dependent codecs are out of scope
>>
>> Most recent codecs are dependent on some form of out-of-band  
>> signalling to negotiate various options and parameters. Prohibiting  
>> any dependence on the signalling seems extreme.
>
> Some time ago, Alexander Chemeris was raising this issue.
> http://www.ietf.org/mail-archive/web/codec/current/msg00508.html
> He was complaining that there is a violation between transport and  
> signaling "layer" (or better plane) for some modern codecs -  
> especially if the signaling protocol has to store states about the  
> negotiated coding mode. This might causes problems if the signaling  
> engine must renegotiate transmission connections or if connections  
> are to be forwarded. He was telling me that the signaling part of a  
> call forwarding engine must include codec specific extension and  
> that this should be avoided.

RTP transport fundamentally relies on out-of-band signalling for some  
codec parameters.  That's unavoidable, but doesn't need to be a great  
source of complexity.

In the message you reference, Alexander's main concern seems to be  
that the RTP payload formats for AMR and AMR-WB rely on complex out-of- 
band signalling, while he would prefer in-band signalling with fewer  
options. It's true that those payload formats do make extensive use of  
out-of-band signalling, and suffer some complexity as a result, but  
this was very much an intentional design choice:

- There was a desire to support several different modes of operation,  
optimised for different use cases, which required incompatible packet  
formats, where the overhead of in-band signalling was considered too  
high.

- There was a desire to allow some end-points to not implement certain  
features and options of the payload format, and this required out-of- 
band signalling to determine end-point capabilities.

You may disagree with those design choices, but the working group was  
aware of the issues and the complexity of the out-of-band signalling  
at the time, and didn't believe that in-band signalling was  
appropriate. Some other payload formats have made different choices,  
and chosen a different trade-off; it depends on the codec, and the  
design of the RTP payload format.

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




From jean-marc.valin@octasic.com  Thu Oct 22 06:45:17 2009
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Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't 
> involve the codec in some form, performance/quality will be bad. Some 
> codecs need time to adjust the rate. Some can only adjust it in pretty 
> coarse steps. Etc. If you let DCCP adjust the rate blindly, without 
> involving the codec in some form, you might get very good TCP 
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to be 
careful not to include in the codec things that don't belong there (e.g. a 
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will never 
> attempt to probe for more capacity unless the codec generates more data, 
> and why would the codec do that if it's converged to transmit at some 
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to 
probe for more capacity, then you tell the codec that it can use a higher 
rate. Now, if there's a special way in which you want to ask the codec, 
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers 
> drop or mark IP packets. It is up to the application sending those IP 
> packets to react appropriately to the marks or losses. In other words, a 
> layered codec will see losses/marks to packets sent on all layers, and 
> will then need to figure out whether to strip off a layer or not in 
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this is 
why I was saying that IMO layered codecs aren't too useful for congestion 
control (though they have other uses). If there's nothing in the middle of 
the network to strip off the layers, then you might as well not have layers 
at all. All you need is to ask the sender to reduce the rate. You don't 
need layers to have a multi-rate codec. Layering just means that you can 
strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the rate
>> should change depending on congestion is actually not that 
>> codec-specific.
> 
> It *can* be non-codec-specific, see DCCP for example. But quality will 
> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific and 
implement just those. Does anyone have a list of the issues codecs should 
be aware of?

> Most Internet congestion control is based on *creating* losses/marks to 
> determine capacity. People have tried to create congestion control 
> schemes that avoid causing losses (delay-based schemed, for example), 
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
> 
> It's not that easy, because rates in the Internet are never stable - the 
> rate computation is usually clocked by the RTT. You can obviously dampen 
> this somehow, but you can never say "now I have found a rate that I can 
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that you 
can have one module that determines in real-time the best rate to use at 
that moment and then it tells the codec what to use. If it needs help from 
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers. On
>> example here (also included in the draft) is VBR, which means that the
>> application specifies an average rate and the codec decides how to 
>> vary the
>> rate to achieve that (subject to some rate control constraints). Would
>> other such mechanisms be useful to deal with congestion?
> 
> I'm not sure this would work, because congestion control will basically 
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just means 
that you'd like to have an average of N kb/s for now, but you're OK if 
there are variations from frame to frame.

	Jean-Marc

From stpeter@stpeter.im  Thu Oct 22 08:02:10 2009
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On 10/22/09 1:56 AM, Lars Eggert wrote:
> Hi,
> 
> On 2009-10-22, at 1:31, Peter Saint-Andre wrote:
>> - - Collaborate with the AVT WG and TSV WG to understand the degree of
>>  rate adaptation desirable, and to reflect that understanding in the
>>  design of a codec that can adjust its transmission in a way that
>>  minimizes disruption to the audio.
> ...
>> - - Collaborate with the AVT WG and DCCP WG to identify important
>>  aspects of packet transmission over the Internet.
> 
> Minor nit. I'd change both TSV WG and DCCP to TSV area, since it's not
> clear which WG this would land in (and there is a chance that DCCP will
> close soon).

Done in my working copy. I'll post a revised version later today.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From mknappe@juniper.net  Thu Oct 22 11:03:21 2009
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From: Michael Knappe <mknappe@juniper.net>
To: "jean-marc.valin@octasic.com" <jean-marc.valin@octasic.com>, "koen.vos@skype.net" <koen.vos@skype.net>, "borilin@spiritdsp.net" <borilin@spiritdsp.net>, "xiphmont@xiph.org" <xiphmont@xiph.org>, "rchen@broadcom.com" <rchen@broadcom.com>
Date: Thu, 22 Oct 2009 10:56:59 -0700
Thread-Topic: [codec] codec tandeming/codec-requirements
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One thing we might want to add to 'draft-valin-codec-requirements' is a bri=
ef discussion of quality expectations in a tandemed codec transmission path=
. For example, a potential scenario for the proposed codec, at least early =
on, may have the proposed codec tandemed between potentially two disparate =
cell codecs:

Cell phone 1 codec type 'A' -> transcode to internet codec -> transcode to =
cell phone 2 codec type 'B'

As we develop a quality methodology, it will be worthwhile to specify some =
typical tandem scenarios to evaluate voice quality in mixed codec environme=
nts, including tandeming of the proposed codec with itself (e.g. Two IP PBX=
's that get connected to each other over via TDM). Evaluations of different=
 LBR codecs in the past have shown that under the same tandem scenario, two=
 candidate codecs may get markedly different tandem MOS scores even though =
the two might have very similar single encode/decode scores.

Thoughts on that?

Michael Knappe
Juniper Networks


From Borilin@spiritdsp.com  Thu Oct 22 11:23:19 2009
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I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?=20

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't=20
> involve the codec in some form, performance/quality will be bad. Some=20
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without=20
> involving the codec in some form, you might get very good TCP=20
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be=20
careful not to include in the codec things that don't belong there (e.g.
a=20
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never=20
> attempt to probe for more capacity unless the codec generates more
data,=20
> and why would the codec do that if it's converged to transmit at some=20
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to=20
probe for more capacity, then you tell the codec that it can use a
higher=20
rate. Now, if there's a special way in which you want to ask the codec,=20
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers=20
> drop or mark IP packets. It is up to the application sending those IP=20
> packets to react appropriately to the marks or losses. In other words,
a=20
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in=20
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is=20
why I was saying that IMO layered codecs aren't too useful for
congestion=20
control (though they have other uses). If there's nothing in the middle
of=20
the network to strip off the layers, then you might as well not have
layers=20
at all. All you need is to ask the sender to reduce the rate. You don't=20
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that=20
>> codec-specific.
>=20
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and=20
implement just those. Does anyone have a list of the issues codecs
should=20
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to=20
> determine capacity. People have tried to create congestion control=20
> schemes that avoid causing losses (delay-based schemed, for example),=20
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>=20
> It's not that easy, because rates in the Internet are never stable -
the=20
> rate computation is usually clocked by the RTT. You can obviously
dampen=20
> this somehow, but you can never say "now I have found a rate that I
can=20
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you=20
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from=20
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to=20
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>=20
> I'm not sure this would work, because congestion control will
basically=20
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means=20
that you'd like to have an average of N kb/s for now, but you're OK if=20
there are variations from frame to frame.

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

From gmaxwell@juniper.net  Thu Oct 22 11:54:59 2009
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From: Gregory Maxwell <gmaxwell@juniper.net>
To: Michael Knappe <mknappe@juniper.net>, "jean-marc.valin@octasic.com" <jean-marc.valin@octasic.com>, "koen.vos@skype.net" <koen.vos@skype.net>, "borilin@spiritdsp.net" <borilin@spiritdsp.net>, "xiphmont@xiph.org" <xiphmont@xiph.org>, "rchen@broadcom.com" <rchen@broadcom.com>
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Michael Knappe wrote:
> One thing we might want to add to 'draft-valin-codec-requirements' is a b=
rief discussion of quality expectations in a tandemed codec transmission pa=
th.
[snip]
> two candidate codecs may get markedly different tandem MOS scores even th=
ough the two might have very similar single encode/decode scores.

In the single codec case it can also be important to distinguish 'synchrono=
us' and 'asynchronous' tandeming: I.e. is the sample the same position in a=
 frame in all cases or has there been some uncompensated offset.  This can =
make a fairly large difference (for example, tandeming is currently almost =
nearly in CELT (in terms of PQevalaudio ODG loss[1], at least) if you're ca=
reful to retain alignment=97 but the quality degrades with tandeming cycles=
 if you do not).  I expect that since frame alignment can't always be easil=
y preserved (i.e. an TDM stage in between) that both cases are worth profil=
ing.=20

I'm not sure that this could easily be converted into a clear requirement=
=97 but I think no one would consider it acceptable for a single stage to c=
ause an enormous quality drop.=20


[1] On a 64kbit/sec CBR encode of a random music clip I get a PEAQ ODG of -=
0.346 for one pass through the codec; -0.362 for 1000 passes. This is somew=
hat less than the loss from a 3kbit reduction in bitrate at  -0.369)


From hsinnrei@adobe.com  Thu Oct 22 12:09:56 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Slava Borilin <Borilin@spiritdsp.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
Date: Thu, 22 Oct 2009 12:09:51 -0700
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Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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


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<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Slava,<BR>
<BR>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<BR>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<BR>
&gt;not let router understand the codec?<BR>
<BR>
The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.<BR>
All applications that nobody planned for can run over it.<BR>
The tragedy of the telephone network was is knew everything about its only =
one application.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spiri=
tdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>I am not that sure that routers should not =
be aware of the codecs. Why?<BR>
If we are building MPLS and other systems that is content-dependent, why<BR=
>
not let router understand the codec?<BR>
<BR>
Especially if we have hordes of voice channels going thru router, it can<BR=
>
be too hard and too long to tell each sender to reduce the rate - why<BR>
not to leave this freedom to router as this may be more efficient for<BR>
congestion recovery?<BR>
<BR>
Same applies to H.264SVC for example - at the end of the day, router do<BR>
not need to understand anything except that it is LAYERED streams (even<BR>
don't know whether its voice, video or audio or anything else of layered<BR=
>
nature, and router &quot;has the right&quot; to strip out layers in certain=
 order<BR>
to prevent congestion?<BR>
<BR>
best regards,<BR>
Slava Borilin<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a hre=
f=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] On B=
ehalf<BR>
Of Jean-Marc Valin<BR>
Sent: Thursday, October 22, 2009 5:45 PM<BR>
To: Lars Eggert<BR>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
Subject: Re: [codec] updated charter proposal<BR>
<BR>
Lars Eggert wrote:<BR>
&gt; Also, what we've learned from media over DCCP is that if you don't<BR>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
BR>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<BR>
<BR>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<BR=
>
&gt; involving the codec in some form, you might get very good TCP<BR>
&gt; friendliness, but maybe quality will suffer.<BR>
<BR>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt to<BR>
be<BR>
careful not to include in the codec things that don't belong there (e.g.<BR=
>
a<BR>
DCCP implementation or a whole jitter buffer).<BR>
<BR>
&gt; (And there is always the ramping up part - congestion control will<BR>
never<BR>
&gt; attempt to probe for more capacity unless the codec generates more<BR>
data,<BR>
&gt; and why would the codec do that if it's converged to transmit at some<=
BR>
&gt; lower rate.)<BR>
<BR>
Well, the codec transmits at the rate you ask it to use. If you want to<BR>
probe for more capacity, then you tell the codec that it can use a<BR>
higher<BR>
rate. Now, if there's a special way in which you want to ask the codec,<BR>
then maybe that can become an additional feature for the codec.<BR>
<BR>
&gt; By no means should routers have to understand codec formats! Routers<B=
R>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
BR>
&gt; packets to react appropriately to the marks or losses. In other words,=
<BR>
a<BR>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<BR>
<BR>
&gt; will then need to figure out whether to strip off a layer or not in<BR=
>
&gt; response to whatever congestion level those losses/marks indicate.<BR>
<BR>
I definitely agree that the routers shouldn't know about codecs and this<BR=
>
is<BR>
why I was saying that IMO layered codecs aren't too useful for<BR>
congestion<BR>
control (though they have other uses). If there's nothing in the middle<BR>
of<BR>
the network to strip off the layers, then you might as well not have<BR>
layers<BR>
at all. All you need is to ask the sender to reduce the rate. You don't<BR>
need layers to have a multi-rate codec. Layering just means that you can<BR=
>
<BR>
strip off bits and still have the result sound OK.<BR>
<BR>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<BR>
rate<BR>
&gt;&gt; should change depending on congestion is actually not that<BR>
&gt;&gt; codec-specific.<BR>
&gt;<BR>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<BR>
<BR>
&gt; suffer IMO.<BR>
<BR>
Well, we need to figure out the aspects that should be codec-specific<BR>
and<BR>
implement just those. Does anyone have a list of the issues codecs<BR>
should<BR>
be aware of?<BR>
<BR>
&gt; Most Internet congestion control is based on *creating* losses/marks<B=
R>
to<BR>
&gt; determine capacity. People have tried to create congestion control<BR>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
BR>
&gt; but they are perceived to be more fragile.<BR>
<BR>
Yes, I can imagine the problem not being trivial at all.<BR>
<BR>
&gt;&gt; Once you figure this out, you<BR>
&gt;&gt; can just tell the codec to use that rate.<BR>
&gt;<BR>
&gt; It's not that easy, because rates in the Internet are never stable -<B=
R>
the<BR>
&gt; rate computation is usually clocked by the RTT. You can obviously<BR>
dampen<BR>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<BR>
can<BR>
&gt; safely use forever.&quot;<BR>
<BR>
Oh, I never said we should set a rate forever. I was just saying that<BR>
you<BR>
can have one module that determines in real-time the best rate to use at<BR=
>
<BR>
that moment and then it tells the codec what to use. If it needs help<BR>
from<BR>
the codec, then we can define special functionality.<BR>
<BR>
&gt;&gt; On top of that, the codec may be<BR>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the =
other layers.<BR>
On<BR>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<BR>
the<BR>
&gt;&gt; application specifies an average rate and the codec decides how to=
<BR>
&gt;&gt; vary the<BR>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<B=
R>
Would<BR>
&gt;&gt; other such mechanisms be useful to deal with congestion?<BR>
&gt;<BR>
&gt; I'm not sure this would work, because congestion control will<BR>
basically<BR>
&gt; change the VBR input parameter continuously.<BR>
<BR>
Changing the VBR parameters continuously is not a problem. VBR just<BR>
means<BR>
that you'd like to have an average of N kb/s for now, but you're OK if<BR>
there are variations from frame to frame.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
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From: "Slava Borilin" <Borilin@spiritdsp.com>
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Henry,

With all of my respect (you know) - this is too high-level consideration
and too generic statement which doesn't answer MPLS issue for example.

=20

best regards,

Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent,
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it
knows nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its
only one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Henry,<o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>With all of my =
respect
(you know) &#8211; this is too high-level consideration and too generic
statement which doesn&#8217;t answer MPLS issue for =
example.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,</span></font><font
color=3Dnavy><span lang=3DEN-US =
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Henry Sinnreich [mailto:hsinnrei@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 =
face=3DCalibri><span
style=3D'font-size:18.0pt;font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; =
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 =
face=3DCalibri><span
style=3D'font-size:18.0pt;font-family:Calibri'>I am not that sure that =
routers
should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
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/mailman/listinfo/codec</a><br>
_______________________________________________<br>
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Slava Borilin [Borilin@spiritdsp.com] wrote:
> I am not that sure that routers should not be aware of the codecs. Why?
> If we are building MPLS and other systems that is content-dependent, why
> not let router understand the codec?
> Especially if we have hordes of voice channels going thru router, it can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?

The most obvious internet friendly way of accomplishing this would be to pl=
ace separate layers in separate packets with distinct TOS markings so that =
they may be differentially dropped.  This would work on already deployed ne=
tworks and networking equipment.=20

However, even ignoring the problems created by the effect of independent ji=
tter on these distinct flows,  the typical ratio of bitrates to packet rate=
s for acceptable latency means that even with a single 'layer' the packets =
already have a lot of overhead (25% for 48kbit/sec into 20ms for IPv4). I w=
ould not expect doubling the packet rate (and overhead) just to achieve dif=
ferential dropping to be a much of a win, compared to an alternative like s=
ending every packet twice by including the previous packet in each frame. =
=20

Client protocol RFCs which have expected widespread changes to infrastructu=
re packet processing have not had a recent history of good success  (I.e. m=
y ECN enabled client still hits problems accessing websites).

Expecting the general packet forwarding infrastructure to gain the ability =
to twiddle the codec bits is pretty much the opposite of what I would expec=
t to do in a codec intended for the internet.

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Are there standards showing the linkage between MPLS and specific applicati=
on components, such as codecs?

Henry

On 10/22/09 2:12 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

Henry,
With all of my respect (you know) - this is too high-level consideration an=
d too generic statement which doesn't answer MPLS issue for example.


best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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


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<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Are there standards showing the linkage between MPLS and specific app=
lication components, such as codecs?<BR>
<BR>
Henry<BR>
<BR>
On 10/22/09 2:12 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spiri=
tdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT COLOR=3D"#000080"><FONT SIZE=3D"0"><FONT FA=
CE=3D"Arial"><SPAN STYLE=3D'font-size:10pt'>Henry,<BR>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion and too generic statement which doesn&#8217;t answer MPLS issue for exa=
mple.<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Aria=
l"><SPAN STYLE=3D'font-size:18pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"0"><FONT FACE=3D"Arial"=
><SPAN STYLE=3D'font-size:10pt'>best regards,<BR>
Slava Borilin<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Aria=
l"><SPAN STYLE=3D'font-size:18pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT SIZE=3D"1"><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'><HR ALIGN=3DCENTER SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT></FONT>
<P>
<FONT SIZE=3D"0"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:10pt'><B>From:</B> Henry Sinnreich [<a href=3D"mailto:hsinn=
rei@adobe.com">mailto:hsinnrei@adobe.com</a>] <BR>
<B>Sent:</B> Thursday, October 22, 2009 11:10 PM<BR>
<B>To:</B> Slava Borilin; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT SIZE=3D"1"><FONT FACE=3D"Times New Roman"><SPAN =
STYLE=3D'font-size:12pt'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPA=
N STYLE=3D'font-size:18pt'>Slava,<BR>
<BR>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<BR>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<BR>
&gt;not let router understand the codec?<BR>
<BR>
The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.<BR>
All applications that nobody planned for can run over it.<BR>
The tragedy of the telephone network was is knew everything about its only =
one application.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spiri=
tdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
I am not that sure that routers should not be aware of the codecs. Why?<BR>
If we are building MPLS and other systems that is content-dependent, why<BR=
>
not let router understand the codec?<BR>
<BR>
Especially if we have hordes of voice channels going thru router, it can<BR=
>
be too hard and too long to tell each sender to reduce the rate - why<BR>
not to leave this freedom to router as this may be more efficient for<BR>
congestion recovery?<BR>
<BR>
Same applies to H.264SVC for example - at the end of the day, router do<BR>
not need to understand anything except that it is LAYERED streams (even<BR>
don't know whether its voice, video or audio or anything else of layered<BR=
>
nature, and router &quot;has the right&quot; to strip out layers in certain=
 order<BR>
to prevent congestion?<BR>
<BR>
best regards,<BR>
Slava Borilin<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a hre=
f=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] On B=
ehalf<BR>
Of Jean-Marc Valin<BR>
Sent: Thursday, October 22, 2009 5:45 PM<BR>
To: Lars Eggert<BR>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
Subject: Re: [codec] updated charter proposal<BR>
<BR>
Lars Eggert wrote:<BR>
&gt; Also, what we've learned from media over DCCP is that if you don't<BR>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
BR>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<BR>
<BR>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<BR=
>
&gt; involving the codec in some form, you might get very good TCP<BR>
&gt; friendliness, but maybe quality will suffer.<BR>
<BR>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt to<BR>
be<BR>
careful not to include in the codec things that don't belong there (e.g.<BR=
>
a<BR>
DCCP implementation or a whole jitter buffer).<BR>
<BR>
&gt; (And there is always the ramping up part - congestion control will<BR>
never<BR>
&gt; attempt to probe for more capacity unless the codec generates more<BR>
data,<BR>
&gt; and why would the codec do that if it's converged to transmit at some<=
BR>
&gt; lower rate.)<BR>
<BR>
Well, the codec transmits at the rate you ask it to use. If you want to<BR>
probe for more capacity, then you tell the codec that it can use a<BR>
higher<BR>
rate. Now, if there's a special way in which you want to ask the codec,<BR>
then maybe that can become an additional feature for the codec.<BR>
<BR>
&gt; By no means should routers have to understand codec formats! Routers<B=
R>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
BR>
&gt; packets to react appropriately to the marks or losses. In other words,=
<BR>
a<BR>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<BR>
<BR>
&gt; will then need to figure out whether to strip off a layer or not in<BR=
>
&gt; response to whatever congestion level those losses/marks indicate.<BR>
<BR>
I definitely agree that the routers shouldn't know about codecs and this<BR=
>
is<BR>
why I was saying that IMO layered codecs aren't too useful for<BR>
congestion<BR>
control (though they have other uses). If there's nothing in the middle<BR>
of<BR>
the network to strip off the layers, then you might as well not have<BR>
layers<BR>
at all. All you need is to ask the sender to reduce the rate. You don't<BR>
need layers to have a multi-rate codec. Layering just means that you can<BR=
>
<BR>
strip off bits and still have the result sound OK.<BR>
<BR>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<BR>
rate<BR>
&gt;&gt; should change depending on congestion is actually not that<BR>
&gt;&gt; codec-specific.<BR>
&gt;<BR>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<BR>
<BR>
&gt; suffer IMO.<BR>
<BR>
Well, we need to figure out the aspects that should be codec-specific<BR>
and<BR>
implement just those. Does anyone have a list of the issues codecs<BR>
should<BR>
be aware of?<BR>
<BR>
&gt; Most Internet congestion control is based on *creating* losses/marks<B=
R>
to<BR>
&gt; determine capacity. People have tried to create congestion control<BR>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
BR>
&gt; but they are perceived to be more fragile.<BR>
<BR>
Yes, I can imagine the problem not being trivial at all.<BR>
<BR>
&gt;&gt; Once you figure this out, you<BR>
&gt;&gt; can just tell the codec to use that rate.<BR>
&gt;<BR>
&gt; It's not that easy, because rates in the Internet are never stable -<B=
R>
the<BR>
&gt; rate computation is usually clocked by the RTT. You can obviously<BR>
dampen<BR>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<BR>
can<BR>
&gt; safely use forever.&quot;<BR>
<BR>
Oh, I never said we should set a rate forever. I was just saying that<BR>
you<BR>
can have one module that determines in real-time the best rate to use at<BR=
>
<BR>
that moment and then it tells the codec what to use. If it needs help<BR>
from<BR>
the codec, then we can define special functionality.<BR>
<BR>
&gt;&gt; On top of that, the codec may be<BR>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the =
other layers.<BR>
On<BR>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<BR>
the<BR>
&gt;&gt; application specifies an average rate and the codec decides how to=
<BR>
&gt;&gt; vary the<BR>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<B=
R>
Would<BR>
&gt;&gt; other such mechanisms be useful to deal with congestion?<BR>
&gt;<BR>
&gt; I'm not sure this would work, because congestion control will<BR>
basically<BR>
&gt; change the VBR input parameter continuously.<BR>
<BR>
Changing the VBR parameters continuously is not a problem. VBR just<BR>
means<BR>
that you'd like to have an average of N kb/s for now, but you're OK if<BR>
there are variations from frame to frame.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Slava Borilin" <Borilin@spiritdsp.com>, "Henry Sinnreich" <hsinnrei@adobe.com>, "Jean-Marc Valin" <jean-marc.valin@octasic.com>, "Lars Eggert" <lars.eggert@nokia.com>
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Again - my point is that router does NOT have to know that something =
specifically is the codec, or specific codec- the only goal is to let =
router aware that the stream (and every packet) is LAYERED, and that it =
is legitimate to strip out some layers if necessary due to congestion.

The content of such "layered" can be any - voice codec, video codec, =
audio codec, any type of content.

=20

And I agree with Gregory - that having current infrastructure using =
layers in separate streams will require too much overhead (packets) =
which does not make sense as well - and I am thinking about one stream =
of packets comtaining all the layers, which, however, can be modified on =
the fly "in the middle".

So we may think about more generic concept of "declared layered content" =
which does NOT loose the information (well, most of it) by stripping out =
portions of the packet and reducing the traffic, so router can strip out =
any type of layered content, without going deep into specific codec.

=20

Again- this is "na=EFve" suggestions from me (very limited engineer) - =
and definitely I am looking for objections even more then for agreement =
- for the sake of either finding the idea useful or making judged deny =
of it, and I agree that this concept is only one possible way of working =
with congestion.

=20

best regards,

Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Slava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Henry,

With all of my respect (you know) - this is too high-level consideration =
and too generic statement which doesn't answer MPLS issue for example.

=20

best regards,

Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, =
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it =
knows nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its =
only one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Again - my point =
is that
router does NOT have to know that something specifically is the codec, =
or
specific codec- the only goal is to let router aware that the stream =
(and every
packet) is LAYERED, and that it is legitimate to strip out some layers =
if
necessary due to congestion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>The content of =
such &#8220;layered&#8221;
can be any &#8211; voice codec, video codec, audio codec, any type of =
content.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>And I agree with =
Gregory &#8211;
that having current infrastructure using layers in separate streams will =
require
too much overhead (packets) which does not make sense as well &#8211; =
and I am
thinking about one stream of packets comtaining all the layers, which, =
however,
can be modified on the fly &#8220;in the =
middle&#8221;.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>So we may think =
about more
generic concept of &#8220;declared layered content&#8221; which does NOT =
loose
the information (well, most of it) by stripping out portions of the =
packet and reducing
the traffic, so router can strip out any type of layered content, =
without going
deep into specific codec.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Again- this is =
&#8220;na&iuml;ve&#8221;
suggestions from me (very limited engineer) &#8211; and definitely I am =
looking
for objections even more then for agreement &#8211; for the sake of =
either
finding the idea useful or making judged deny of it, and I agree that =
this
concept is only one possible way of working with =
congestion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,</span></font><font
color=3Dnavy><span lang=3DEN-US =
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Henry,<o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>With all of my =
respect
(you know) &#8211; this is too high-level consideration and too generic
statement which doesn&#8217;t answer MPLS issue for =
example.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,</span></font><font
color=3Dnavy><span lang=3DEN-US =
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Henry Sinnreich [mailto:hsinnrei@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated charter
proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 =
face=3DCalibri><span
style=3D'font-size:18.0pt;font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; =
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 =
face=3DCalibri><span
style=3D'font-size:18.0pt;font-family:Calibri'>I am not that sure that =
routers
should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a></span></font><o:p></o:p></p>

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Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec -> transcode to
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because they
have been designed for these use cases. 

With best regards,

 Christian





From Borilin@spiritdsp.com  Thu Oct 22 12:40:09 2009
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Michael,

If you are thinking about core network to carry voice in IP between BSCs
- why should you care about any of internet codecs- as this is managed
network normally?

Or I misunderstand you goal?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Christian Hoene
Sent: Thursday, October 22, 2009 11:33 PM
To: 'Michael Knappe'
Cc: codec@ietf.org
Subject: Re: [codec] codec tandeming/codec-requirements

Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec ->
transcode to
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because
they
have been designed for these use cases.=20

With best regards,

 Christian




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

From mknappe@juniper.net  Thu Oct 22 12:43:46 2009
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From: Michael Knappe <mknappe@juniper.net>
To: "hoene@uni-tuebingen.de" <hoene@uni-tuebingen.de>
Date: Thu, 22 Oct 2009 12:42:19 -0700
Thread-Topic: [codec]  codec tandeming/codec-requirements
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Christian,=20

I am not looking for a particular codec type, only pointing out that any pr=
oposed codec may end up receiving traffic that at some point has already be=
en compressed/uncompressed and lost some speech information redundancy in t=
he uncompressed speech as it hits the internet codec. We should at least un=
derstand how the proposed codec will perform when receiving that previously=
 compressed traffic.

Mike

----- Original Message -----
From: Christian Hoene <hoene@uni-tuebingen.de>
To: Michael Knappe
Cc: codec@ietf.org <codec@ietf.org>
Sent: Thu Oct 22 15:32:46 2009=0A=
Subject: RE: [codec]  codec tandeming/codec-requirements

Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec -> transcode t=
o
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because they
have been designed for these use cases.=20

With best regards,

 Christian





From jean-marc.valin@octasic.com  Thu Oct 22 12:45:07 2009
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Hi Slava,

Slava Borilin wrote:
> Again - my point is that router does NOT have to know that something 
> specifically is the codec, or specific codec- the only goal is to let 
> router aware that the stream (and every packet) is LAYERED, and that it 
> is legitimate to strip out some layers if necessary due to congestion.
> 
> The content of such layered can be any  voice codec, video codec, 
> audio codec, any type of content.

One of the issues here is that not only does the router not know about the 
codec, but most (all?) routers don't even know about anything other than 
layer 3 (IP). It doesn't even know whether the packet transports UDP or 
TCP, much less whether there any media or anything. Routers are several 
orders of magnitude too dumb to strip some bits off a codec payload.

Cheers,

	Jean-Marc

From Borilin@spiritdsp.com  Thu Oct 22 12:49:08 2009
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From: "Slava Borilin" <Borilin@spiritdsp.com>
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Sure this is what we have now.
Question is when it comes to congestion - it is worth "teaching
routers", or this does not make sense from overall economy.

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 11:45 PM
To: Slava Borilin
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Hi Slava,

Slava Borilin wrote:
> Again - my point is that router does NOT have to know that something=20
> specifically is the codec, or specific codec- the only goal is to let=20
> router aware that the stream (and every packet) is LAYERED, and that
it=20
> is legitimate to strip out some layers if necessary due to congestion.
>=20
> The content of such "layered" can be any - voice codec, video codec,=20
> audio codec, any type of content.

One of the issues here is that not only does the router not know about
the=20
codec, but most (all?) routers don't even know about anything other than

layer 3 (IP). It doesn't even know whether the packet transports UDP or=20
TCP, much less whether there any media or anything. Routers are several=20
orders of magnitude too dumb to strip some bits off a codec payload.

Cheers,

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

From petithug@acm.org  Thu Oct 22 12:52:43 2009
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Jean-Marc Valin wrote:
> Hi Slava,
> 
> Slava Borilin wrote:
>> Again - my point is that router does NOT have to know that something
>> specifically is the codec, or specific codec- the only goal is to let
>> router aware that the stream (and every packet) is LAYERED, and that
>> it is legitimate to strip out some layers if necessary due to congestion.
>>
>> The content of such layered can be any  voice codec, video codec,
>> audio codec, any type of content.
> 
> One of the issues here is that not only does the router not know about
> the codec, but most (all?) routers don't even know about anything other
> than layer 3 (IP). It doesn't even know whether the packet transports
> UDP or TCP, much less whether there any media or anything. Routers are
> several orders of magnitude too dumb to strip some bits off a codec
> payload.

Right, but one can imagine an IPv6 header that contains one or more offsets
related to the payload that would let a router aware of this extension that it
can shorten the packet to one of this offset if needed.

-- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
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From stefan.sayer@iptego.com  Thu Oct 22 13:08:26 2009
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To: Christian Hoene <hoene@uni-tuebingen.de>
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o Christian Hoene [10/22/09 21:32]:
> Dear Michael Knappe,
> 
>> Cell phone 1 codec type 'A' -> transcode to internet codec -> transcode to
>> cell phone 2 codec type 'B'
> 
> Honestly, in this case I would recommend you to use ITU codecs because they
> have been designed for these use cases. 

I don't think so. IMO bringing applications to cellular networks by good 
quality tandeming with wideband from cellular networks will be one of 
the major application areas for IWAC. Things like

Cell phone 1 codec type 'A' -> transcode to IWAC -> media/application 
server in the internet.

Cell phone 2 codec type 'B' -> transcode to IWAC -> same 
media/application server in the internet.

VoIP phone 3 on IWAC/internet -> same media/application server in the 
internet.

etc.

Stefan

> 
> With best regards,
> 
>  Christian
> 
> 
> 
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec

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From csp@csperkins.org  Thu Oct 22 13:15:38 2009
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From: Colin Perkins <csp@csperkins.org>
To: Slava Borilin <Borilin@spiritdsp.com>
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Slava,

RTP has the concept of a translator, which performs the functions =20
you're looking for in an application-layer middlebox, where it can be =20=

part of the signalling and/or security context. This is not something =20=

a router should do.

Colin




On 22 Oct 2009, at 20:22, Slava Borilin wrote:
> Again - my point is that router does NOT have to know that something =20=

> specifically is the codec, or specific codec- the only goal is to =20
> let router aware that the stream (and every packet) is LAYERED, and =20=

> that it is legitimate to strip out some layers if necessary due to =20
> congestion.
> The content of such =93layered=94 can be any =96 voice codec, video =
codec, =20
> audio codec, any type of content.
>
> And I agree with Gregory =96 that having current infrastructure using =20=

> layers in separate streams will require too much overhead (packets) =20=

> which does not make sense as well =96 and I am thinking about one =20
> stream of packets comtaining all the layers, which, however, can be =20=

> modified on the fly =93in the middle=94.
> So we may think about more generic concept of =93declared layered =20
> content=94 which does NOT loose the information (well, most of it) by =20=

> stripping out portions of the packet and reducing the traffic, so =20
> router can strip out any type of layered content, without going deep =20=

> into specific codec.
>
> Again- this is =93na=EFve=94 suggestions from me (very limited =
engineer) =96 =20
> and definitely I am looking for objections even more then for =20
> agreement =96 for the sake of either finding the idea useful or making =
=20
> judged deny of it, and I agree that this concept is only one =20
> possible way of working with congestion.
>
> best regards,
> Slava Borilin
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On =20
> Behalf OfSlava Borilin
> Sent: Thursday, October 22, 2009 11:13 PM
> To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
> Henry,
> With all of my respect (you know) =96 this is too high-level =20
> consideration and too generic statement which doesn=92t answer MPLS =20=

> issue for example.
>
> best regards,
> Slava Borilin
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> Sent: Thursday, October 22, 2009 11:10 PM
> To: Slava Borilin; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
> Slava,
>
> >I am not that sure that routers should not be aware of the codecs. =20=

> Why?
> >If we are building MPLS and other systems that is content-=20
> dependent, why
> >not let router understand the codec?
>
> The essence of the Dumb Network (Internet) and its routers is that =20
> it knows nothing about the applications.
> All applications that nobody planned for can run over it.
> The tragedy of the telephone network was is knew everything about =20
> its only one application.
>
> Henry
>
>
> On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
>
> I am not that sure that routers should not be aware of the codecs. =20
> Why?
> If we are building MPLS and other systems that is content-dependent, =20=

> why
> not let router understand the codec?
>
> Especially if we have hordes of voice channels going thru router, it =20=

> can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?
>
> Same applies to H.264SVC for example - at the end of the day, router =20=

> do
> not need to understand anything except that it is LAYERED streams =20
> (even
> don't know whether its voice, video or audio or anything else of =20
> layered
> nature, and router "has the right" to strip out layers in certain =20
> order
> to prevent congestion?
>
> best regards,
> Slava Borilin
>
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Thursday, October 22, 2009 5:45 PM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
> Lars Eggert wrote:
> > Also, what we've learned from media over DCCP is that if you don't
> > involve the codec in some form, performance/quality will be bad. =20
> Some
> > codecs need time to adjust the rate. Some can only adjust it in =20
> pretty
>
> > coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> > involving the codec in some form, you might get very good TCP
> > friendliness, but maybe quality will suffer.
>
> I'm totally in favour of "involving the codec" in here. I just want to
> be
> careful not to include in the codec things that don't belong there =20
> (e.g.
> a
> DCCP implementation or a whole jitter buffer).
>
> > (And there is always the ramping up part - congestion control will
> never
> > attempt to probe for more capacity unless the codec generates more
> data,
> > and why would the codec do that if it's converged to transmit at =20
> some
> > lower rate.)
>
> Well, the codec transmits at the rate you ask it to use. If you want =20=

> to
> probe for more capacity, then you tell the codec that it can use a
> higher
> rate. Now, if there's a special way in which you want to ask the =20
> codec,
> then maybe that can become an additional feature for the codec.
>
> > By no means should routers have to understand codec formats! Routers
> > drop or mark IP packets. It is up to the application sending those =20=

> IP
> > packets to react appropriately to the marks or losses. In other =20
> words,
> a
> > layered codec will see losses/marks to packets sent on all layers, =20=

> and
>
> > will then need to figure out whether to strip off a layer or not in
> > response to whatever congestion level those losses/marks indicate.
>
> I definitely agree that the routers shouldn't know about codecs and =20=

> this
> is
> why I was saying that IMO layered codecs aren't too useful for
> congestion
> control (though they have other uses). If there's nothing in the =20
> middle
> of
> the network to strip off the layers, then you might as well not have
> layers
> at all. All you need is to ask the sender to reduce the rate. You =20
> don't
> need layers to have a multi-rate codec. Layering just means that you =20=

> can
>
> strip off bits and still have the result sound OK.
>
> >> OK, I may be wrong here, but my understanding is that how much the
> rate
> >> should change depending on congestion is actually not that
> >> codec-specific.
> >
> > It *can* be non-codec-specific, see DCCP for example. But quality =20=

> will
>
> > suffer IMO.
>
> Well, we need to figure out the aspects that should be codec-specific
> and
> implement just those. Does anyone have a list of the issues codecs
> should
> be aware of?
>
> > Most Internet congestion control is based on *creating* losses/marks
> to
> > determine capacity. People have tried to create congestion control
> > schemes that avoid causing losses (delay-based schemed, for =20
> example),
> > but they are perceived to be more fragile.
>
> Yes, I can imagine the problem not being trivial at all.
>
> >> Once you figure this out, you
> >> can just tell the codec to use that rate.
> >
> > It's not that easy, because rates in the Internet are never stable -
> the
> > rate computation is usually clocked by the RTT. You can obviously
> dampen
> > this somehow, but you can never say "now I have found a rate that I
> can
> > safely use forever."
>
> Oh, I never said we should set a rate forever. I was just saying that
> you
> can have one module that determines in real-time the best rate to =20
> use at
>
> that moment and then it tells the codec what to use. If it needs help
> from
> the codec, then we can define special functionality.
>
> >> On top of that, the codec may be
> >> able to do a bit better by providing a "service" to the other =20
> layers.
> On
> >> example here (also included in the draft) is VBR, which means that
> the
> >> application specifies an average rate and the codec decides how to
> >> vary the
> >> rate to achieve that (subject to some rate control constraints).
> Would
> >> other such mechanisms be useful to deal with congestion?
> >
> > I'm not sure this would work, because congestion control will
> basically
> > change the VBR input parameter continuously.
>
> Changing the VBR parameters continuously is not a problem. VBR just
> means
> that you'd like to have an average of N kb/s for now, but you're OK if
> there are variations from frame to frame.
>
>         Jean-Marc
> _______________________________________________
> 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
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec



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




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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Slava,<div><br></div><div>RTP =
has the concept of a translator, which performs the functions you're =
looking for in an application-layer middlebox, where it can be part of =
the signalling and/or security context. This is not something a router =
should =
do.&nbsp;</div><div><br></div><div>Colin</div><div><br></div><div><br></di=
v><div><br></div><div><br><div><div>On 22 Oct 2009, at 20:22, Slava =
Borilin wrote:</div><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; font-size: medium; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div lang=3D"RU" link=3D"blue" =
vlink=3D"blue"><div class=3D"Section1"><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; ">Again - my point is that router does =
NOT have to know that something specifically is the codec, or specific =
codec- the only goal is to let router aware that the stream (and every =
packet) is LAYERED, and that it is legitimate to strip out some layers =
if necessary due to congestion.<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Arial; color: navy; ">The content =
of such =93layered=94 can be any =96 voice codec, video codec, audio =
codec, any type of content.<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Arial; color: navy; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; ">And I agree with Gregory =96 that =
having current infrastructure using layers in separate streams will =
require too much overhead (packets) which does not make sense as well =96 =
and I am thinking about one stream of packets comtaining all the layers, =
which, however, can be modified on the fly =93in the =
middle=94.<o:p></o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; ">So we may think about more generic =
concept of =93declared layered content=94 which does NOT loose the =
information (well, most of it) by stripping out portions of the packet =
and reducing the traffic, so router can strip out any type of layered =
content, without going deep into specific =
codec.<o:p></o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; ">Again- this is =93na=EFve=94 =
suggestions from me (very limited engineer) =96 and definitely I am =
looking for objections even more then for agreement =96 for the sake of =
either finding the idea useful or making judged deny of it, and I agree =
that this concept is only one possible way of working with =
congestion.<o:p></o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; =
"><o:p>&nbsp;</o:p></span></font></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; =
font-size: 12pt; font-family: 'Times New Roman'; "><font size=3D"2" =
color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: =
10pt; font-family: Arial; color: navy; ">best =
regards,</span></font><font color=3D"navy"><span lang=3D"EN-US" =
style=3D"color: navy; "><o:p></o:p></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Arial; color: navy; ">Slava =
Borilin</span></font><span =
lang=3D"EN-US"><o:p></o:p></span></div></div><div><div class=3D"MsoNormal"=
 align=3D"center" style=3D"margin-top: 0cm; margin-right: 0cm; =
margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: =
'Times New Roman'; text-align: center; "><font size=3D"3" face=3D"Times =
New Roman"><span lang=3D"EN-US" style=3D"font-size: 12pt; "><hr size=3D"2"=
 width=3D"100%" align=3D"center" tabindex=3D"-1"></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Tahoma; font-weight: bold; =
">From:</span></font></b><font size=3D"2" face=3D"Tahoma"><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Tahoma; "><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a =
href=3D"mailto:codec-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">mailto:codec-bounces@ietf.org</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><b><span =
style=3D"font-weight: bold; ">On Behalf Of</span></b>Slava =
Borilin<br><b><span style=3D"font-weight: bold; ">Sent:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Thursday, October 22, 2009 =
11:13 PM<br><b><span style=3D"font-weight: bold; ">To:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Henry Sinnreich; Jean-Marc =
Valin; Lars Eggert<br><b><span style=3D"font-weight: bold; =
">Cc:</span></b><span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:codec@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">codec@ietf.org</a><br><b><span style=3D"font-weight: bold; =
">Subject:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [codec] updated charter =
proposal</span></font><span =
lang=3D"EN-US"><o:p></o:p></span></div></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; =
font-size: 12pt; font-family: 'Times New Roman'; "><font size=3D"3" =
face=3D"Times New Roman"><span style=3D"font-size: 12pt; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; =
">Henry,<o:p></o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; ">With all of my respect (you know) =96 =
this is too high-level consideration and too generic statement which =
doesn=92t answer MPLS issue for =
example.<o:p></o:p></span></font></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: Arial; color: navy; =
"><o:p>&nbsp;</o:p></span></font></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; =
font-size: 12pt; font-family: 'Times New Roman'; "><font size=3D"2" =
color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" style=3D"font-size: =
10pt; font-family: Arial; color: navy; ">best =
regards,</span></font><font color=3D"navy"><span lang=3D"EN-US" =
style=3D"color: navy; "><o:p></o:p></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"navy" face=3D"Arial"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Arial; color: navy; ">Slava =
Borilin</span></font><span =
lang=3D"EN-US"><o:p></o:p></span></div></div><div><div class=3D"MsoNormal"=
 align=3D"center" style=3D"margin-top: 0cm; margin-right: 0cm; =
margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: =
'Times New Roman'; text-align: center; "><font size=3D"3" face=3D"Times =
New Roman"><span lang=3D"EN-US" style=3D"font-size: 12pt; "><hr size=3D"2"=
 width=3D"100%" align=3D"center" tabindex=3D"-1"></span></font></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: Tahoma; font-weight: bold; =
">From:</span></font></b><font size=3D"2" face=3D"Tahoma"><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Tahoma; "><span =
class=3D"Apple-converted-space">&nbsp;</span>Henry Sinnreich [<a =
href=3D"mailto:hsinnrei@adobe.com" style=3D"color: blue; =
text-decoration: underline; ">mailto:hsinnrei@adobe.com</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b><span =
style=3D"font-weight: bold; ">Sent:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Thursday, October 22, 2009 =
11:10 PM<br><b><span style=3D"font-weight: bold; ">To:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Slava Borilin; Jean-Marc =
Valin; Lars Eggert<br><b><span style=3D"font-weight: bold; =
">Cc:</span></b><span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:codec@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">codec@ietf.org</a><br><b><span style=3D"font-weight: bold; =
">Subject:</span></b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [codec] updated charter =
proposal</span></font><span =
lang=3D"EN-US"><o:p></o:p></span></div></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; =
font-size: 12pt; font-family: 'Times New Roman'; "><font size=3D"3" =
face=3D"Times New Roman"><span style=3D"font-size: 12pt; =
"><o:p>&nbsp;</o:p></span></font></div><p class=3D"MsoNormal" =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 12pt; =
margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"5" face=3D"Calibri"><span style=3D"font-size: 18pt; =
font-family: Calibri; ">Slava,<br><br>&gt;I am not that sure that =
routers should not be aware of the codecs. Why?<br>&gt;If we are =
building MPLS and other systems that is content-dependent, =
why<br>&gt;not let router understand the codec?<br><br>The essence of =
the Dumb Network (Internet) and its routers is that it knows nothing =
about the applications.<br>All applications that nobody planned for can =
run over it.<br>The tragedy of the telephone network was is knew =
everything about its only one application.<br><br>Henry<br><br><br>On =
10/22/09 1:23 PM, "Slava Borilin" &lt;<a href=3D"Borilin@spiritdsp.com" =
style=3D"color: blue; text-decoration: underline; =
">Borilin@spiritdsp.com</a>&gt; wrote:</span></font><o:p></o:p></p><p =
class=3D"MsoNormal" style=3D"margin-top: 0cm; margin-right: 0cm; =
margin-bottom: 12pt; margin-left: 0cm; font-size: 12pt; font-family: =
'Times New Roman'; "><font size=3D"5" face=3D"Calibri"><span =
style=3D"font-size: 18pt; font-family: Calibri; ">I am not that sure =
that routers should not be aware of the codecs. Why?<br>If we are =
building MPLS and other systems that is content-dependent, why<br>not =
let router understand the codec?<br><br>Especially if we have hordes of =
voice channels going thru router, it can<br>be too hard and too long to =
tell each sender to reduce the rate - why<br>not to leave this freedom =
to router as this may be more efficient for<br>congestion =
recovery?<br><br>Same applies to H.264SVC for example - at the end of =
the day, router do<br>not need to understand anything except that it is =
LAYERED streams (even<br>don't know whether its voice, video or audio or =
anything else of layered<br>nature, and router "has the right" to strip =
out layers in certain order<br>to prevent congestion?<br><br>best =
regards,<br>Slava Borilin<br><br>-----Original =
Message-----<br>From:<span class=3D"Apple-converted-space">&nbsp;</span><a=
 href=3D"codec-bounces@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">codec-bounces@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>[<a =
href=3D"mailto:codec-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">mailto:codec-bounces@ietf.org</a>] On =
Behalf<br>Of Jean-Marc Valin<br>Sent: Thursday, October 22, 2009 5:45 =
PM<br>To: Lars Eggert<br>Cc:<span =
class=3D"Apple-converted-space">&nbsp;</span><a href=3D"codec@ietf.org" =
style=3D"color: blue; text-decoration: underline; =
">codec@ietf.org</a><br>Subject: Re: [codec] updated charter =
proposal<br><br>Lars Eggert wrote:<br>&gt; Also, what we've learned from =
media over DCCP is that if you don't<br>&gt; involve the codec in some =
form, performance/quality will be bad. Some<br>&gt; codecs need time to =
adjust the rate. Some can only adjust it in pretty<br><br>&gt; coarse =
steps. Etc. If you let DCCP adjust the rate blindly, without<br>&gt; =
involving the codec in some form, you might get very good TCP<br>&gt; =
friendliness, but maybe quality will suffer.<br><br>I'm totally in =
favour of "involving the codec" in here. I just want to<br>be<br>careful =
not to include in the codec things that don't belong there =
(e.g.<br>a<br>DCCP implementation or a whole jitter buffer).<br><br>&gt; =
(And there is always the ramping up part - congestion control =
will<br>never<br>&gt; attempt to probe for more capacity unless the =
codec generates more<br>data,<br>&gt; and why would the codec do that if =
it's converged to transmit at some<br>&gt; lower rate.)<br><br>Well, the =
codec transmits at the rate you ask it to use. If you want to<br>probe =
for more capacity, then you tell the codec that it can use =
a<br>higher<br>rate. Now, if there's a special way in which you want to =
ask the codec,<br>then maybe that can become an additional feature for =
the codec.<br><br>&gt; By no means should routers have to understand =
codec formats! Routers<br>&gt; drop or mark IP packets. It is up to the =
application sending those IP<br>&gt; packets to react appropriately to =
the marks or losses. In other words,<br>a<br>&gt; layered codec will see =
losses/marks to packets sent on all layers, and<br><br>&gt; will then =
need to figure out whether to strip off a layer or not in<br>&gt; =
response to whatever congestion level those losses/marks =
indicate.<br><br>I definitely agree that the routers shouldn't know =
about codecs and this<br>is<br>why I was saying that IMO layered codecs =
aren't too useful for<br>congestion<br>control (though they have other =
uses). If there's nothing in the middle<br>of<br>the network to strip =
off the layers, then you might as well not have<br>layers<br>at all. All =
you need is to ask the sender to reduce the rate. You don't<br>need =
layers to have a multi-rate codec. Layering just means that you =
can<br><br>strip off bits and still have the result sound =
OK.<br><br>&gt;&gt; OK, I may be wrong here, but my understanding is =
that how much the<br>rate<br>&gt;&gt; should change depending on =
congestion is actually not that<br>&gt;&gt; =
codec-specific.<br>&gt;<br>&gt; It *can* be non-codec-specific, see DCCP =
for example. But quality will<br><br>&gt; suffer IMO.<br><br>Well, we =
need to figure out the aspects that should be =
codec-specific<br>and<br>implement just those. Does anyone have a list =
of the issues codecs<br>should<br>be aware of?<br><br>&gt; Most Internet =
congestion control is based on *creating* losses/marks<br>to<br>&gt; =
determine capacity. People have tried to create congestion =
control<br>&gt; schemes that avoid causing losses (delay-based schemed, =
for example),<br>&gt; but they are perceived to be more =
fragile.<br><br>Yes, I can imagine the problem not being trivial at =
all.<br><br>&gt;&gt; Once you figure this out, you<br>&gt;&gt; can just =
tell the codec to use that rate.<br>&gt;<br>&gt; It's not that easy, =
because rates in the Internet are never stable -<br>the<br>&gt; rate =
computation is usually clocked by the RTT. You can =
obviously<br>dampen<br>&gt; this somehow, but you can never say "now I =
have found a rate that I<br>can<br>&gt; safely use forever."<br><br>Oh, =
I never said we should set a rate forever. I was just saying =
that<br>you<br>can have one module that determines in real-time the best =
rate to use at<br><br>that moment and then it tells the codec what to =
use. If it needs help<br>from<br>the codec, then we can define special =
functionality.<br><br>&gt;&gt; On top of that, the codec may =
be<br>&gt;&gt; able to do a bit better by providing a "service" to the =
other layers.<br>On<br>&gt;&gt; example here (also included in the =
draft) is VBR, which means that<br>the<br>&gt;&gt; application specifies =
an average rate and the codec decides how to<br>&gt;&gt; vary =
the<br>&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>Would<br>&gt;&gt; other such mechanisms be useful to =
deal with congestion?<br>&gt;<br>&gt; I'm not sure this would work, =
because congestion control will<br>basically<br>&gt; change the VBR =
input parameter continuously.<br><br>Changing the VBR parameters =
continuously is not a problem. VBR just<br>means<br>that you'd like to =
have an average of N kb/s for now, but you're OK if<br>there are =
variations from frame to =
frame.<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br=
>_______________________________________________<br>codec mailing =
list<br><a href=3D"codec@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">codec@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/codec" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/codec</a><br>_____________________=
__________________________<br>codec mailing list<br><a =
href=3D"codec@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">codec@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/codec" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/codec</a></span></font><o:p></o:p>=
</p></div>_______________________________________________<br>codec =
mailing list<br><a href=3D"mailto:codec@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">codec@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/codec" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/codec</a><br></div></span></blockq=
uote></div><br><div apple-content-edited=3D"true"> <span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></div></span> =
</div><br></div></body></html>=

--Apple-Mail-5-909952088--

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Subject: Re: [codec] updated charter proposal
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Hi,

The problem with congestion control protocols is that they have limited 
implementation on consumer level hardware.

We've tried something as simple as setting the IP TOS byte to try to 
prioritize voice packets, and in practice very few home routers honor 
it, and there are quite a few wireless routers which silently discard 
frames with the TOS byte set. The reason we even tried this is that 
there is nothing we can do about our own stream to help the latency or 
congestion; the problem is usually that the user is also uploading quite 
a few torrents, so their uplink is completely full.

For most networks, the bottleneck is that first/last hop; it's the home 
users DSL/Cable connection, or a WiFi connection that has to compete 
with other networks on the same channel, or it could be the mobile 
users's connection to the tower. Beyond that point there is usually 
sufficient bandwidth that congestion doesn't occur.

I don't think this problem is going to improve; I can't think of any 
congestion control which can make my neighbour's WiFi network back off 
when I want to transmit high priority packets. So, while interfacing 
with congestion protocols and teaching routers about codecs may be 
something we'd like in an ideal world, it's more important to create a 
codec that will work in real world networks, which are congested and 
buggy, and are configured and used by non-technical people.

Best Regards,
Thorvald


From mknappe@juniper.net  Thu Oct 22 14:54:51 2009
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From: Michael Knappe <mknappe@juniper.net>
To: Slava Borilin <Borilin@spiritdsp.com>, Christian Hoene <hoene@uni-tuebingen.de>
Date: Thu, 22 Oct 2009 14:54:20 -0700
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The rather extreme example I was thinking was a cell phone coming in via TD=
M to an IP PBX based branch office, then, being forwarded via SIP trunk usi=
ng the proposed internet codec to another IP PBX branch where it happens to=
 get routed off the PBX via TDM and the PSTN to another cell network and ph=
one. This is a particularly problematic multiple tandem scenario with sever=
al passes through G.711, but even a double tandem scenario of a cell phone =
codec transcoded through G.711 and TDM then transcoded again to reach an IP=
 network endpoint on a fabric using the proposed codec would be relatively =
common in the near term.

>From a 40,000 ft level, this all becomes a multi-variate quality balance of=
 the different non-linear human perception impact thresholds (one-way delay=
, distortion, noise, residual echo, etc). It's difficult from the quality p=
erspective to separate a single codec out of the end-to-end quality equatio=
n due to the non-linearity of the human perception curves. For example, the=
 perceived quality impact of one-way network delay stays relatively low and=
 constant below 150 ms, but rises dramatically by the time you hit 250 ms).=
 In the case of one-way delay, you can have two codecs that by themselves e=
asily meet the 150 ms quality criteria on their own, but once in-situ and t=
andemed, they cross the knee of the non-linear quality impact curve with a =
noticeable impact on quality. This can easily be heard on many cell-phone t=
o cell-phone calls across disparate cell networks.

That said, we most definitely should have specific quality goals and measur=
ement methodologies specified for the codec on its own. Just recommending f=
rom my own experience that we define a few more complex topologies where we=
 look at end-to-end quality with the new codec in-situ.

Mike


On 10/22/09 12:40 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

Michael,

If you are thinking about core network to carry voice in IP between BSCs
- why should you care about any of internet codecs- as this is managed
network normally?

Or I misunderstand you goal?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Christian Hoene
Sent: Thursday, October 22, 2009 11:33 PM
To: 'Michael Knappe'
Cc: codec@ietf.org
Subject: Re: [codec] codec tandeming/codec-requirements

Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec ->
transcode to
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because
they
have been designed for these use cases.

With best regards,

 Christian




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


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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Michael Knappe <mknappe@juniper.net>, Slava Borilin <Borilin@spiritdsp.com>, Christian Hoene <hoene@uni-tuebingen.de>
Date: Thu, 22 Oct 2009 17:07:03 -0700
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--_000_C7065DD768D5hsinnreiadobecom_
Content-Type: text/plain; charset="iso-8859-1"
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Mike,

>The rather extreme example I was thinking was a cell phone coming in via T=
DM to an IP PBX based branch >office, then, being forwarded via SIP trunk u=
sing the proposed internet codec to another IP PBX branch where >it happens=
 to get routed off the PBX via TDM and the PSTN to another cell network and=
 phone. This is a >particularly problematic multiple tandem scenario with s=
everal passes through G.711, but even a double >tandem scenario of a cell p=
hone codec transcoded through G.711 and TDM then transcoded again to reach =
an
>IP network endpoint on a fabric using the proposed codec would be relative=
ly common in the near term.

How about using the Internet e2e topology?

Henry


On 10/22/09 4:54 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

The rather extreme example I was thinking was a cell phone coming in via TD=
M to an IP PBX based branch office, then, being forwarded via SIP trunk usi=
ng the proposed internet codec to another IP PBX branch where it happens to=
 get routed off the PBX via TDM and the PSTN to another cell network and ph=
one. This is a particularly problematic multiple tandem scenario with sever=
al passes through G.711, but even a double tandem scenario of a cell phone =
codec transcoded through G.711 and TDM then transcoded again to reach an IP=
 network endpoint on a fabric using the proposed codec would be relatively =
common in the near term.

>From a 40,000 ft level, this all becomes a multi-variate quality balance of=
 the different non-linear human perception impact thresholds (one-way delay=
, distortion, noise, residual echo, etc). It's difficult from the quality p=
erspective to separate a single codec out of the end-to-end quality equatio=
n due to the non-linearity of the human perception curves. For example, the=
 perceived quality impact of one-way network delay stays relatively low and=
 constant below 150 ms, but rises dramatically by the time you hit 250 ms).=
 In the case of one-way delay, you can have two codecs that by themselves e=
asily meet the 150 ms quality criteria on their own, but once in-situ and t=
andemed, they cross the knee of the non-linear quality impact curve with a =
noticeable impact on quality. This can easily be heard on many cell-phone t=
o cell-phone calls across disparate cell networks.

That said, we most definitely should have specific quality goals and measur=
ement methodologies specified for the codec on its own. Just recommending f=
rom my own experience that we define a few more complex topologies where we=
 look at end-to-end quality with the new codec in-situ.

Mike


On 10/22/09 12:40 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

Michael,

If you are thinking about core network to carry voice in IP between BSCs
- why should you care about any of internet codecs- as this is managed
network normally?

Or I misunderstand you goal?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Christian Hoene
Sent: Thursday, October 22, 2009 11:33 PM
To: 'Michael Knappe'
Cc: codec@ietf.org
Subject: Re: [codec] codec tandeming/codec-requirements

Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec ->
transcode to
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because
they
have been designed for these use cases.

With best regards,

 Christian




_______________________________________________
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


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Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [codec] codec tandeming/codec-requirements</TITLE>
</HEAD>
<BODY>
<FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN ST=
YLE=3D'font-size:14pt'>Mike,<BR>
<BR>
&gt;The rather extreme example I was thinking was a cell phone coming in vi=
a TDM to an IP PBX based branch &gt;office, then, being forwarded via SIP t=
runk using the proposed internet codec to another IP PBX branch where &gt;i=
t happens to get routed off the PBX via TDM and the PSTN to another cell ne=
twork and phone. This is a &gt;particularly problematic multiple tandem sce=
nario with several passes through G.711, but even a double &gt;tandem scena=
rio of a cell phone codec transcoded through G.711 and TDM then transcoded =
again to reach an <BR>
&gt;IP network endpoint on a fabric using the proposed codec would be relat=
ively common in the near term.<BR>
<BR>
How about using the Internet e2e topology?<BR>
<BR>
Henry<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPA=
N STYLE=3D'font-size:18pt'><BR>
<BR>
On 10/22/09 4:54 PM, &quot;Michael Knappe&quot; &lt;<a href=3D"mknappe@juni=
per.net">mknappe@juniper.net</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>The rather extreme example I was thinking w=
as a cell phone coming in via TDM to an IP PBX based branch office, then, b=
eing forwarded via SIP trunk using the proposed internet codec to another I=
P PBX branch where it happens to get routed off the PBX via TDM and the PST=
N to another cell network and phone. This is a particularly problematic mul=
tiple tandem scenario with several passes through G.711, but even a double =
tandem scenario of a cell phone codec transcoded through G.711 and TDM then=
 transcoded again to reach an IP network endpoint on a fabric using the pro=
posed codec would be relatively common in the near term.<BR>
<BR>
>From a 40,000 ft level, this all becomes a multi-variate quality balance of=
 the different non-linear human perception impact thresholds (one-way delay=
, distortion, noise, residual echo, etc). It's difficult from the quality p=
erspective to separate a single codec out of the end-to-end quality equatio=
n due to the non-linearity of the human perception curves. For example, the=
 perceived quality impact of one-way network delay stays relatively low and=
 constant below 150 ms, but rises dramatically by the time you hit 250 ms).=
 In the case of one-way delay, you can have two codecs that by themselves e=
asily meet the 150 ms quality criteria on their own, but once in-situ and t=
andemed, they cross the knee of the non-linear quality impact curve with a =
noticeable impact on quality. This can easily be heard on many cell-phone t=
o cell-phone calls across disparate cell networks.<BR>
<BR>
That said, we most definitely should have specific quality goals and measur=
ement methodologies specified for the codec on its own. Just recommending f=
rom my own experience that we define a few more complex topologies where we=
 look at end-to-end quality with the new codec in-situ.<BR>
<BR>
Mike<BR>
<BR>
<BR>
On 10/22/09 12:40 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spir=
itdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
<BR>
Michael,<BR>
<BR>
If you are thinking about core network to carry voice in IP between BSCs<BR=
>
- why should you care about any of internet codecs- as this is managed<BR>
network normally?<BR>
<BR>
Or I misunderstand you goal?<BR>
<BR>
best regards,<BR>
Slava Borilin<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a hre=
f=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] On B=
ehalf<BR>
Of Christian Hoene<BR>
Sent: Thursday, October 22, 2009 11:33 PM<BR>
To: 'Michael Knappe'<BR>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
Subject: Re: [codec] codec tandeming/codec-requirements<BR>
<BR>
Dear Michael Knappe,<BR>
<BR>
&gt; Cell phone 1 codec type 'A' -&gt; transcode to internet codec -&gt;<BR=
>
transcode to<BR>
&gt; cell phone 2 codec type 'B'<BR>
<BR>
Honestly, in this case I would recommend you to use ITU codecs because<BR>
they<BR>
have been designed for these use cases.<BR>
<BR>
With best regards,<BR>
<BR>
&nbsp;Christian<BR>
<BR>
<BR>
<BR>
<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C7065DD768D5hsinnreiadobecom_--

From swmike@swm.pp.se  Thu Oct 22 22:49:31 2009
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From: Mikael Abrahamsson <swmike@swm.pp.se>
To: Slava Borilin <Borilin@spiritdsp.com>
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On Thu, 22 Oct 2009, Slava Borilin wrote:

> Question is when it comes to congestion - it is worth "teaching
> routers", or this does not make sense from overall economy.

IP was successful because the "network" didn't need to handle 
applications. Getting application specific tech into routers is a 5+ year 
waiting period (ECN still isn't there, and that's quite simple) and the 
direction routers are taking right now is bigger, faster and simpler, not 
the other way around.

So any proposal that relies on "the Internet" doing something special to 
your packets outside of what it already can do and what's in the pipe, 
will delay deployment for a long time.

Relying on ECN (optionally) for congestion signals might work though, 
that's been in the pipe for 5+ years and some adoption might happen.

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

From Borilin@spiritdsp.com  Fri Oct 23 00:24:24 2009
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, "Michael Knappe" <mknappe@juniper.net>, "Christian Hoene" <hoene@uni-tuebingen.de>
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Henry, e2e (or p2p) is cool but what about Legal interception for
example in this case, which every cellco has to implemenet?

=20

best regards,

Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Friday, October 23, 2009 4:07 AM
To: Michael Knappe; Slava Borilin; Christian Hoene
Cc: codec@ietf.org
Subject: Re: [codec] codec tandeming/codec-requirements

=20

Mike,

>The rather extreme example I was thinking was a cell phone coming in
via TDM to an IP PBX based branch >office, then, being forwarded via SIP
trunk using the proposed internet codec to another IP PBX branch where
>it happens to get routed off the PBX via TDM and the PSTN to another
cell network and phone. This is a >particularly problematic multiple
tandem scenario with several passes through G.711, but even a double
>tandem scenario of a cell phone codec transcoded through G.711 and TDM
then transcoded again to reach an=20
>IP network endpoint on a fabric using the proposed codec would be
relatively common in the near term.

How about using the Internet e2e topology?

Henry


On 10/22/09 4:54 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

The rather extreme example I was thinking was a cell phone coming in via
TDM to an IP PBX based branch office, then, being forwarded via SIP
trunk using the proposed internet codec to another IP PBX branch where
it happens to get routed off the PBX via TDM and the PSTN to another
cell network and phone. This is a particularly problematic multiple
tandem scenario with several passes through G.711, but even a double
tandem scenario of a cell phone codec transcoded through G.711 and TDM
then transcoded again to reach an IP network endpoint on a fabric using
the proposed codec would be relatively common in the near term.

>From a 40,000 ft level, this all becomes a multi-variate quality balance
of the different non-linear human perception impact thresholds (one-way
delay, distortion, noise, residual echo, etc). It's difficult from the
quality perspective to separate a single codec out of the end-to-end
quality equation due to the non-linearity of the human perception
curves. For example, the perceived quality impact of one-way network
delay stays relatively low and constant below 150 ms, but rises
dramatically by the time you hit 250 ms). In the case of one-way delay,
you can have two codecs that by themselves easily meet the 150 ms
quality criteria on their own, but once in-situ and tandemed, they cross
the knee of the non-linear quality impact curve with a noticeable impact
on quality. This can easily be heard on many cell-phone to cell-phone
calls across disparate cell networks.

That said, we most definitely should have specific quality goals and
measurement methodologies specified for the codec on its own. Just
recommending from my own experience that we define a few more complex
topologies where we look at end-to-end quality with the new codec
in-situ.

Mike


On 10/22/09 12:40 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

Michael,

If you are thinking about core network to carry voice in IP between BSCs
- why should you care about any of internet codecs- as this is managed
network normally?

Or I misunderstand you goal?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Christian Hoene
Sent: Thursday, October 22, 2009 11:33 PM
To: 'Michael Knappe'
Cc: codec@ietf.org
Subject: Re: [codec] codec tandeming/codec-requirements

Dear Michael Knappe,

> Cell phone 1 codec type 'A' -> transcode to internet codec ->
transcode to
> cell phone 2 codec type 'B'

Honestly, in this case I would recommend you to use ITU codecs because
they
have been designed for these use cases.

With best regards,

 Christian




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

_______________________________________________
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Henry, e2e (or =
p2p) is
cool but what about Legal interception for example in this case, which =
every
cellco has to implemenet?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>best regards,</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Henry Sinnreich [mailto:hsinnrei@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Friday, October 23, =
2009
4:07 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Michael Knappe; Slava =
Borilin;
Christian Hoene<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
codec
tandeming/codec-requirements</span></font><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D4 =
face=3DCalibri><span
style=3D'font-size:14.0pt;font-family:Calibri'>Mike,<br>
<br>
&gt;The rather extreme example I was thinking was a cell phone coming in =
via
TDM to an IP PBX based branch &gt;office, then, being forwarded via SIP =
trunk
using the proposed internet codec to another IP PBX branch where &gt;it =
happens
to get routed off the PBX via TDM and the PSTN to another cell network =
and
phone. This is a &gt;particularly problematic multiple tandem scenario =
with
several passes through G.711, but even a double &gt;tandem scenario of a =
cell
phone codec transcoded through G.711 and TDM then transcoded again to =
reach an <br>
&gt;IP network endpoint on a fabric using the proposed codec would be
relatively common in the near term.<br>
<br>
How about using the Internet e2e topology?<br>
<br>
Henry<br>
</span></font><font size=3D5 face=3DCalibri><span =
style=3D'font-size:18.0pt;
font-family:Calibri'><br>
<br>
On 10/22/09 4:54 PM, &quot;Michael Knappe&quot; &lt;<a
href=3D"mknappe@juniper.net">mknappe@juniper.net</a>&gt; =
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 =
face=3DCalibri><span
style=3D'font-size:18.0pt;font-family:Calibri'>The rather extreme =
example I was
thinking was a cell phone coming in via TDM to an IP PBX based branch =
office,
then, being forwarded via SIP trunk using the proposed internet codec to
another IP PBX branch where it happens to get routed off the PBX via TDM =
and
the PSTN to another cell network and phone. This is a particularly =
problematic
multiple tandem scenario with several passes through G.711, but even a =
double
tandem scenario of a cell phone codec transcoded through G.711 and TDM =
then
transcoded again to reach an IP network endpoint on a fabric using the =
proposed
codec would be relatively common in the near term.<br>
<br>
>From a 40,000 ft level, this all becomes a multi-variate quality balance =
of the
different non-linear human perception impact thresholds (one-way delay,
distortion, noise, residual echo, etc). It's difficult from the quality
perspective to separate a single codec out of the end-to-end quality =
equation
due to the non-linearity of the human perception curves. For example, =
the
perceived quality impact of one-way network delay stays relatively low =
and
constant below 150 ms, but rises dramatically by the time you hit 250 =
ms). In
the case of one-way delay, you can have two codecs that by themselves =
easily
meet the 150 ms quality criteria on their own, but once in-situ and =
tandemed,
they cross the knee of the non-linear quality impact curve with a =
noticeable
impact on quality. This can easily be heard on many cell-phone to =
cell-phone
calls across disparate cell networks.<br>
<br>
That said, we most definitely should have specific quality goals and
measurement methodologies specified for the codec on its own. Just =
recommending
from my own experience that we define a few more complex topologies =
where we
look at end-to-end quality with the new codec in-situ.<br>
<br>
Mike<br>
<br>
<br>
On 10/22/09 12:40 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
<br>
Michael,<br>
<br>
If you are thinking about core network to carry voice in IP between =
BSCs<br>
- why should you care about any of internet codecs- as this is =
managed<br>
network normally?<br>
<br>
Or I misunderstand you goal?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Christian Hoene<br>
Sent: Thursday, October 22, 2009 11:33 PM<br>
To: 'Michael Knappe'<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] codec tandeming/codec-requirements<br>
<br>
Dear Michael Knappe,<br>
<br>
&gt; Cell phone 1 codec type 'A' -&gt; transcode to internet codec =
-&gt;<br>
transcode to<br>
&gt; cell phone 2 codec type 'B'<br>
<br>
Honestly, in this case I would recommend you to use ITU codecs =
because<br>
they<br>
have been designed for these use cases.<br>
<br>
With best regards,<br>
<br>
&nbsp;Christian<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a></span></font><o:p></o:p></p>

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Date: Fri, 23 Oct 2009 10:34:18 +0300
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References: <4ADE2ED9.2030101@stpeter.im><F28F5303-A99E-4C40-9FE3-9ADA233B9B76@nokia.com><4ADF0C41.3010808@octasic.com><030726E0-55AC-4828-B6D9-926550CF70A9@nokia.com> <4AE061F5.7080301@octasic.com>, <AA5A65FC22B6F145830AC0EAC7586A6C0513E874@mail-srv.spiritcorp.com> <BCB3F026FAC4C145A4A3330806FEFDA93A2B50766F@EMBX01-HQ.jnpr.net> <4AE0BFA7.1040106@natvig.com>
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Hi,

On 2009-10-22, at 23:25, Thorvald Natvig wrote:
> The problem with congestion control protocols is that they have  
> limited
> implementation on consumer level hardware.
>
> We've tried something as simple as setting the IP TOS byte to try to
> prioritize voice packets

please understand that setting or honoring DSCPs is *not* congestion  
control. Congestion control requires a sender to lower the rate of a  
flow when that flow experiences packet loss (or ECN marks).

Lars
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From ron@debian.org  Fri Oct 23 02:48:30 2009
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From: Ron <ron@debian.org>
To: codec@ietf.org
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Subject: Re: [codec] codec tandeming/codec-requirements
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On Thu, Oct 22, 2009 at 10:08:33PM +0200, Stefan Sayer wrote:
> o Christian Hoene [10/22/09 21:32]:
> >Dear Michael Knappe,
> >
> >>Cell phone 1 codec type 'A' -> transcode to internet codec -> transcode to
> >>cell phone 2 codec type 'B'
> >
> >Honestly, in this case I would recommend you to use ITU codecs because they
> >have been designed for these use cases.
> 
> I don't think so. IMO bringing applications to cellular networks by
> good quality tandeming with wideband from cellular networks will be
> one of the major application areas for IWAC. Things like
> 
> Cell phone 1 codec type 'A' -> transcode to IWAC ->
> media/application server in the internet.
> 
> Cell phone 2 codec type 'B' -> transcode to IWAC -> same
> media/application server in the internet.
> 
> VoIP phone 3 on IWAC/internet -> same media/application server in
> the internet.

By the time this codec is widely deployed will there be (m)any
mobile devices without technology to do IP directly?

If "codec type 'A'" is a terrible codec for this purpose, how far
should we compromise to get acceptable results with things that
are "off the internet"?

I agree this raises interesting issues to consider, but requiring
compatibility with particular legacy codecs seems more like a SHOULD
than a MUST.  I think I'd rather change the string between my tin
cans than have this codec constrained to its current best passband.

 Ron



From bmschwar@fas.harvard.edu  Fri Oct 23 06:29:24 2009
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Slava Borilin wrote:
> Henry, e2e (or p2p) is cool but what about Legal interception for
> example in this case, which every cellco has to implemenet?

Whether or not we want to encourage it, interception of VoIP traffic does=

not require a re-encode.  The encoded packets can just be copied to the
additional receiver.

To address the "tandeming" issue, I think the following text should be
added to the "Basic Requirements" or "Additional Considerations":

"""
Audio Corruption Robustness

The systems providing input to the encoder and output from the decoder ar=
e
likely to be far from ideal in actual use.  Input and output audio stream=
s
may be corrupted by artifacts from analog hardware and digital processing=
=2E
 The codecs to be developed should be tested to ensure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion.
"""


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From: Henry Sinnreich <hsinnrei@adobe.com>
To: "bens@alum.mit.edu" <bens@alum.mit.edu>, Slava Borilin <Borilin@spiritdsp.com>
Date: Fri, 23 Oct 2009 07:18:17 -0700
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--_000_C70725596920hsinnreiadobecom_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

>Whether or not we want to encourage it, interception of VoIP traffic does =
not require a re-encode.
>The encoded packets can just be copied to the additional receiver.

This is correct IMO.
There is also traffic analysis and many other tools - not the right topic f=
or this list.
Henry


On 10/23/09 8:29 AM, "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu> wrot=
e:

Slava Borilin wrote:
> Henry, e2e (or p2p) is cool but what about Legal interception for
> example in this case, which every cellco has to implemenet?

Whether or not we want to encourage it, interception of VoIP traffic does
not require a re-encode.  The encoded packets can just be copied to the
additional receiver.

To address the "tandeming" issue, I think the following text should be
added to the "Basic Requirements" or "Additional Considerations":

"""
Audio Corruption Robustness

The systems providing input to the encoder and output from the decoder are
likely to be far from ideal in actual use.  Input and output audio streams
may be corrupted by artifacts from analog hardware and digital processing.
 The codecs to be developed should be tested to ensure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion.
"""



--_000_C70725596920hsinnreiadobecom_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [codec] codec tandeming/codec-requirements</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>&gt;Whether or not we want to encourage it, interception of VoIP traf=
fic does not require a re-encode. &nbsp;<BR>
&gt;The encoded packets can just be copied to the additional receiver.<BR>
<BR>
This is correct IMO. <BR>
There is also traffic analysis and many other tools &#8211; not the right t=
opic for this list.<BR>
Henry<BR>
<BR>
<BR>
On 10/23/09 8:29 AM, &quot;Benjamin M. Schwartz&quot; &lt;<a href=3D"bmschw=
ar@fas.harvard.edu">bmschwar@fas.harvard.edu</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Slava Borilin wrote:<BR>
&gt; Henry, e2e (or p2p) is cool but what about Legal interception for<BR>
&gt; example in this case, which every cellco has to implemenet?<BR>
<BR>
Whether or not we want to encourage it, interception of VoIP traffic does<B=
R>
not require a re-encode. &nbsp;The encoded packets can just be copied to th=
e<BR>
additional receiver.<BR>
<BR>
To address the &quot;tandeming&quot; issue, I think the following text shou=
ld be<BR>
added to the &quot;Basic Requirements&quot; or &quot;Additional Considerati=
ons&quot;:<BR>
<BR>
&quot;&quot;&quot;<BR>
Audio Corruption Robustness<BR>
<BR>
The systems providing input to the encoder and output from the decoder are<=
BR>
likely to be far from ideal in actual use. &nbsp;Input and output audio str=
eams<BR>
may be corrupted by artifacts from analog hardware and digital processing.<=
BR>
&nbsp;The codecs to be developed should be tested to ensure that they degra=
de<BR>
gracefully under adverse audio conditions. &nbsp;Types of digital corruptio=
n<BR>
that may be tested include tandem encoding with similar and different<BR>
codecs, low-quality resampling, and digital clipping. &nbsp;Types of analog=
<BR>
corruption that may be tested include microphones with substantial<BR>
background noise, analog clipping, and loudspeaker distortion.<BR>
&quot;&quot;&quot;<BR>
<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C70725596920hsinnreiadobecom_--

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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Mikael Abrahamsson <swmike@swm.pp.se>, Slava Borilin <Borilin@spiritdsp.com>
Date: Fri, 23 Oct 2009 07:22:13 -0700
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>So any proposal that relies on "the Internet" doing something special to y=
our packets
>outside of what it already can do and what's in the pipe, will delay deplo=
yment for a long time.

Actually for ever. See the Internet stats and the link to the NANOG present=
ed slides:

http://www.telco2.net/blog/2009/10/perhaps_the_most_important_cha_1.html

Henry


On 10/23/09 12:49 AM, "Mikael Abrahamsson" <swmike@swm.pp.se> wrote:

On Thu, 22 Oct 2009, Slava Borilin wrote:

> Question is when it comes to congestion - it is worth "teaching
> routers", or this does not make sense from overall economy.

IP was successful because the "network" didn't need to handle
applications. Getting application specific tech into routers is a 5+ year
waiting period (ECN still isn't there, and that's quite simple) and the
direction routers are taking right now is bigger, faster and simpler, not
the other way around.

So any proposal that relies on "the Internet" doing something special to
your packets outside of what it already can do and what's in the pipe,
will delay deployment for a long time.

Relying on ECN (optionally) for congestion signals might work though,
that's been in the pipe for 5+ years and some adoption might happen.

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


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<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>&gt;So any proposal that relies on &quot;the Internet&quot; doing som=
ething special to your packets <BR>
&gt;outside of what it already can do and what's in the pipe, will delay de=
ployment for a long time.<BR>
<BR>
Actually for ever. See the Internet stats and the link to the NANOG present=
ed slides:<BR>
<BR>
<a href=3D"http://www.telco2.net/blog/2009/10/perhaps_the_most_important_ch=
a_1.html">http://www.telco2.net/blog/2009/10/perhaps_the_most_important_cha=
_1.html</a><BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/23/09 12:49 AM, &quot;Mikael Abrahamsson&quot; &lt;<a href=3D"swmike@=
swm.pp.se">swmike@swm.pp.se</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>On Thu, 22 Oct 2009, Slava Borilin wrote:<B=
R>
<BR>
&gt; Question is when it comes to congestion - it is worth &quot;teaching<B=
R>
&gt; routers&quot;, or this does not make sense from overall economy.<BR>
<BR>
IP was successful because the &quot;network&quot; didn't need to handle<BR>
applications. Getting application specific tech into routers is a 5+ year<B=
R>
waiting period (ECN still isn't there, and that's quite simple) and the<BR>
direction routers are taking right now is bigger, faster and simpler, not<B=
R>
the other way around.<BR>
<BR>
So any proposal that relies on &quot;the Internet&quot; doing something spe=
cial to<BR>
your packets outside of what it already can do and what's in the pipe,<BR>
will delay deployment for a long time.<BR>
<BR>
Relying on ECN (optionally) for congestion signals might work though,<BR>
that's been in the pipe for 5+ years and some adoption might happen.<BR>
<BR>
--<BR>
Mikael Abrahamsson &nbsp;&nbsp;&nbsp;email: <a href=3D"swmike@swm.pp.se">sw=
mike@swm.pp.se</a><BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Slava Borilin'" <Borilin@spiritdsp.com>, "'Henry Sinnreich'" <hsinnrei@adobe.com>, "'Jean-Marc Valin'" <jean-marc.valin@octasic.com>, "'Lars Eggert'" <lars.eggert@nokia.com>
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Slava,

In SVC the terms for it is media-aware network elements  (MANEs) so =
adaption
can be done by the source, destination or by MANEs. Note that they need =
to
understand the structure of the payload and relations between the layers =
to
strip the right ones.

Roni Even

=20

=20

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of
Slava Borilin
Sent: Thursday, October 22, 2009 9:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Again - my point is that router does NOT have to know that something
specifically is the codec, or specific codec- the only goal is to let =
router
aware that the stream (and every packet) is LAYERED, and that it is
legitimate to strip out some layers if necessary due to congestion.

The content of such =93layered=94 can be any =96 voice codec, video =
codec, audio
codec, any type of content.

=20

And I agree with Gregory =96 that having current infrastructure using =
layers
in separate streams will require too much overhead (packets) which does =
not
make sense as well =96 and I am thinking about one stream of packets
comtaining all the layers, which, however, can be modified on the fly =
=93in
the middle=94.

So we may think about more generic concept of =93declared layered =
content=94
which does NOT loose the information (well, most of it) by stripping out
portions of the packet and reducing the traffic, so router can strip out =
any
type of layered content, without going deep into specific codec.

=20

Again- this is =93na=EFve=94 suggestions from me (very limited engineer) =
=96 and
definitely I am looking for objections even more then for agreement =96 =
for
the sake of either finding the idea useful or making judged deny of it, =
and
I agree that this concept is only one possible way of working with
congestion.

=20

best regards,

Slava Borilin

  _____ =20

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of
Slava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Henry,

With all of my respect (you know) =96 this is too high-level =
consideration and
too generic statement which doesn=92t answer MPLS issue for example.

=20

best regards,

Slava Borilin

  _____ =20

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, =
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its =
only
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:

I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

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

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In SVC the terms for it is media-aware network =
elements=A0 (MANEs)
so adaption can be done by the source, destination or by MANEs. Note =
that they
need to understand the structure of the payload and relations between =
the
layers to strip the right ones.<o:p></o:p></span></p>

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

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

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

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0in 4.0pt'>

<div>

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

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b>On Behalf Of =
</b>Slava
Borilin<br>
<b>Sent:</b> Thursday, October 22, 2009 9:23 PM<br>
<b>To:</b> Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars =
Eggert<br>
<b>Cc:</b> codec@ietf.org<br>
<b>Subject:</b> Re: [codec] updated charter =
proposal<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Again - my point is that router does NOT have to know that
something specifically is the codec, or specific codec- the only goal is =
to let
router aware that the stream (and every packet) is LAYERED, and that it =
is
legitimate to strip out some layers if necessary due to =
congestion.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>The content of such &#8220;layered&#8221; can be any &#8211; =
voice codec, video
codec, audio codec, any type of content.<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>And I agree with Gregory &#8211; that having current =
infrastructure using
layers in separate streams will require too much overhead (packets) =
which does
not make sense as well &#8211; and I am thinking about one stream of =
packets
comtaining all the layers, which, however, can be modified on the fly =
&#8220;in the
middle&#8221;.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>So we may think about more generic concept of =
&#8220;declared layered
content&#8221; which does NOT loose the information (well, most of it) =
by stripping
out portions of the packet and reducing the traffic, so router can strip =
out
any type of layered content, without going deep into specific =
codec.<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Again- this is &#8220;na=EFve&#8221; suggestions from me =
(very limited engineer)
&#8211; and definitely I am looking for objections even more then for =
agreement &#8211; for
the sake of either finding the idea useful or making judged deny of it, =
and I
agree that this concept is only one possible way of working with =
congestion.<o:p></o:p></span></p>

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

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>best regards,</span><span =
style=3D'color:navy'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Slava Borilin</span><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<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>Slava
Borilin<br>
<b>Sent:</b> Thursday, October 22, 2009 11:13 PM<br>
<b>To:</b> Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b>Cc:</b> codec@ietf.org<br>
<b>Subject:</b> Re: [codec] updated charter =
proposal</span><o:p></o:p></p>

</div>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Henry,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>With all of my respect (you know) &#8211; this is too =
high-level
consideration and too generic statement which doesn&#8217;t answer MPLS =
issue for
example.<o:p></o:p></span></p>

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

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>best regards,</span><span =
style=3D'color:navy'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Slava Borilin</span><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<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"'> Henry =
Sinnreich
[mailto:hsinnrei@adobe.com] <br>
<b>Sent:</b> Thursday, October 22, 2009 11:10 PM<br>
<b>To:</b> Slava Borilin; Jean-Marc Valin; Lars Eggert<br>
<b>Cc:</b> codec@ietf.org<br>
<b>Subject:</b> Re: [codec] updated charter =
proposal</span><o:p></o:p></p>

</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span lang=3DRU =
style=3D'font-size:
18.0pt;font-family:"Calibri","sans-serif"'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; =
wrote:</span><span
lang=3DRU><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span lang=3DRU =
style=3D'font-size:
18.0pt;font-family:"Calibri","sans-serif"'>I am not that sure that =
routers
should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org=
/mailman/listinfo/codec</a></span><span
lang=3DRU><o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DRU><br>
<br>
__________ Information from ESET NOD32 Antivirus, version of virus =
signature
database 4532 (20091022) __________<br>
<br>
The message was checked by ESET NOD32 Antivirus.<br>
<br>
<a =
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From koen.vos@skype.net  Sun Oct 25 06:38:46 2009
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Date: Sun, 25 Oct 2009 06:38:46 -0700
From: Koen Vos <koen.vos@skype.net>
To: Slava Borilin <Borilin@spiritdsp.com>
References: <AA5A65FC22B6F145830AC0EAC7586A6C0513E874@mail-srv.spiritcorp.com><C706182F.68A0%hsinnrei@adobe.com> <AA5A65FC22B6F145830AC0EAC7586A6C0513E89B@mail-srv.spiritcorp.com> <AA5A65FC22B6F145830AC0EAC7586A6C0513E89D@mail-srv.spiritcorp.com>
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Subject: Re: [codec] updated charter proposal
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At a given quality level, the bitrate for encoding speech/audio with =20
layers is typically 10~50% higher than without layers.  So layered =20
coding may in fact cause congestion, and reducing the bitrate at the =20
encoder is probably better.

koen.


Quoting Slava Borilin:

> Again - my point is that router does NOT have to know that something =20
> specifically is the codec, or specific codec- the only goal is to =20
> let router aware that the stream (and every packet) is LAYERED, and =20
> that it is legitimate to strip out some layers if necessary due to =20
> congestion.
>
> The content of such "layered" can be any - voice codec, video codec, =20
> audio codec, any type of content.
>
>
>
> And I agree with Gregory - that having current infrastructure using =20
> layers in separate streams will require too much overhead (packets) =20
> which does not make sense as well - and I am thinking about one =20
> stream of packets comtaining all the layers, which, however, can be =20
> modified on the fly "in the middle".
>
> So we may think about more generic concept of "declared layered =20
> content" which does NOT loose the information (well, most of it) by =20
> stripping out portions of the packet and reducing the traffic, so =20
> router can strip out any type of layered content, without going deep =20
> into specific codec.
>
>
>
> Again- this is "na=EFve" suggestions from me (very limited engineer) - =20
> and definitely I am looking for objections even more then for =20
> agreement - for the sake of either finding the idea useful or making =20
> judged deny of it, and I agree that this concept is only one =20
> possible way of working with congestion.
>
>
>
> best regards,
>
> Slava Borilin
>
> ________________________________
>
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On =20
> Behalf Of Slava Borilin
> Sent: Thursday, October 22, 2009 11:13 PM
> To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
>
>
> Henry,
>
> With all of my respect (you know) - this is too high-level =20
> consideration and too generic statement which doesn't answer MPLS =20
> issue for example.
>
>
>
> best regards,
>
> Slava Borilin
>
> ________________________________
>
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> Sent: Thursday, October 22, 2009 11:10 PM
> To: Slava Borilin; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
>
>
> Slava,
>
>> I am not that sure that routers should not be aware of the codecs. Why?
>> If we are building MPLS and other systems that is content-dependent, why
>> not let router understand the codec?
>
> The essence of the Dumb Network (Internet) and its routers is that =20
> it knows nothing about the applications.
> All applications that nobody planned for can run over it.
> The tragedy of the telephone network was is knew everything about =20
> its only one application.
>
> Henry
>
>
> On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
>
> I am not that sure that routers should not be aware of the codecs. Why?
> If we are building MPLS and other systems that is content-dependent, why
> not let router understand the codec?
>
> Especially if we have hordes of voice channels going thru router, it can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?
>
> Same applies to H.264SVC for example - at the end of the day, router do
> not need to understand anything except that it is LAYERED streams (even
> don't know whether its voice, video or audio or anything else of layered
> nature, and router "has the right" to strip out layers in certain order
> to prevent congestion?
>
> best regards,
> Slava Borilin
>
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Thursday, October 22, 2009 5:45 PM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>
> Lars Eggert wrote:
>> Also, what we've learned from media over DCCP is that if you don't
>> involve the codec in some form, performance/quality will be bad. Some
>> codecs need time to adjust the rate. Some can only adjust it in pretty
>
>> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
>> involving the codec in some form, you might get very good TCP
>> friendliness, but maybe quality will suffer.
>
> I'm totally in favour of "involving the codec" in here. I just want to
> be
> careful not to include in the codec things that don't belong there (e.g.
> a
> DCCP implementation or a whole jitter buffer).
>
>> (And there is always the ramping up part - congestion control will
> never
>> attempt to probe for more capacity unless the codec generates more
> data,
>> and why would the codec do that if it's converged to transmit at some
>> lower rate.)
>
> Well, the codec transmits at the rate you ask it to use. If you want to
> probe for more capacity, then you tell the codec that it can use a
> higher
> rate. Now, if there's a special way in which you want to ask the codec,
> then maybe that can become an additional feature for the codec.
>
>> By no means should routers have to understand codec formats! Routers
>> drop or mark IP packets. It is up to the application sending those IP
>> packets to react appropriately to the marks or losses. In other words,
> a
>> layered codec will see losses/marks to packets sent on all layers, and
>
>> will then need to figure out whether to strip off a layer or not in
>> response to whatever congestion level those losses/marks indicate.
>
> I definitely agree that the routers shouldn't know about codecs and this
> is
> why I was saying that IMO layered codecs aren't too useful for
> congestion
> control (though they have other uses). If there's nothing in the middle
> of
> the network to strip off the layers, then you might as well not have
> layers
> at all. All you need is to ask the sender to reduce the rate. You don't
> need layers to have a multi-rate codec. Layering just means that you can
>
> strip off bits and still have the result sound OK.
>
>>> OK, I may be wrong here, but my understanding is that how much the
> rate
>>> should change depending on congestion is actually not that
>>> codec-specific.
>>
>> It *can* be non-codec-specific, see DCCP for example. But quality will
>
>> suffer IMO.
>
> Well, we need to figure out the aspects that should be codec-specific
> and
> implement just those. Does anyone have a list of the issues codecs
> should
> be aware of?
>
>> Most Internet congestion control is based on *creating* losses/marks
> to
>> determine capacity. People have tried to create congestion control
>> schemes that avoid causing losses (delay-based schemed, for example),
>> but they are perceived to be more fragile.
>
> Yes, I can imagine the problem not being trivial at all.
>
>>> Once you figure this out, you
>>> can just tell the codec to use that rate.
>>
>> It's not that easy, because rates in the Internet are never stable -
> the
>> rate computation is usually clocked by the RTT. You can obviously
> dampen
>> this somehow, but you can never say "now I have found a rate that I
> can
>> safely use forever."
>
> Oh, I never said we should set a rate forever. I was just saying that
> you
> can have one module that determines in real-time the best rate to use at
>
> that moment and then it tells the codec what to use. If it needs help
> from
> the codec, then we can define special functionality.
>
>>> On top of that, the codec may be
>>> able to do a bit better by providing a "service" to the other layers.
> On
>>> example here (also included in the draft) is VBR, which means that
> the
>>> application specifies an average rate and the codec decides how to
>>> vary the
>>> rate to achieve that (subject to some rate control constraints).
> Would
>>> other such mechanisms be useful to deal with congestion?
>>
>> I'm not sure this would work, because congestion control will
> basically
>> change the VBR input parameter continuously.
>
> Changing the VBR parameters continuously is not a problem. VBR just
> means
> that you'd like to have an average of N kb/s for now, but you're OK if
> there are variations from frame to frame.
>
>         Jean-Marc
> _______________________________________________
> 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
>
>


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From: Jozsef Vass <jovass@adobe.com>
To: Slava Borilin <Borilin@spiritdsp.com>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
Date: Mon, 26 Oct 2009 10:55:47 -0700
Thread-Topic: [codec] updated charter proposal
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I do not like the idea of sending multiple layers in a packet either. I fee=
l very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but delivering semantically modified packet is not expected.

Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.

Jozsef

________________________________
From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something specif=
ically is the codec, or specific codec- the only goal is to let router awar=
e that the stream (and every packet) is LAYERED, and that it is legitimate =
to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, audio =
codec, any type of content.

And I agree with Gregory - that having current infrastructure using layers =
in separate streams will require too much overhead (packets) which does not=
 make sense as well - and I am thinking about one stream of packets comtain=
ing all the layers, which, however, can be modified on the fly "in the midd=
le".
So we may think about more generic concept of "declared layered content" wh=
ich does NOT loose the information (well, most of it) by stripping out port=
ions of the packet and reducing the traffic, so router can strip out any ty=
pe of layered content, without going deep into specific codec.

Again- this is "na=EFve" suggestions from me (very limited engineer) - and =
definitely I am looking for objections even more then for agreement - for t=
he sake of either finding the idea useful or making judged deny of it, and =
I agree that this concept is only one possible way of working with congesti=
on.

best regards,
Slava Borilin
________________________________
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of S=
lava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration an=
d too generic statement which doesn't answer MPLS issue for example.

best regards,
Slava Borilin
________________________________
From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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

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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do not like the idea of sending mult=
iple
layers in a packet either. I feel very uncomfortable of routers modifying
packets. It is OK not to deliver, but delivering semantically modified pack=
et is
not expected.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Using layers for video totally makes s=
ense
where the rates are much higher and layers can be transported in separate
packets.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Bo=
rilin
[mailto:Borilin@spiritdsp.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Again - my point is that router does N=
OT
have to know that something specifically is the codec, or specific codec- t=
he
only goal is to let router aware that the stream (and every packet) is LAYE=
RED,
and that it is legitimate to strip out some layers if necessary due to
congestion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The content of such &#8220;layered&#82=
21; can be any &#8211;
voice codec, video codec, audio codec, any type of content.<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>And I agree with Gregory &#8211; that =
having
current infrastructure using layers in separate streams will require too mu=
ch
overhead (packets) which does not make sense as well &#8211; and I am think=
ing about
one stream of packets comtaining all the layers, which, however, can be
modified on the fly &#8220;in the middle&#8221;.<o:p></o:p></span></font></=
p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>So we may think about more generic con=
cept
of &#8220;declared layered content&#8221; which does NOT loose the informat=
ion (well, most
of it) by stripping out portions of the packet and reducing the traffic, so
router can strip out any type of layered content, without going deep into
specific codec.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Again- this is &#8220;na=EFve&#8221; s=
uggestions from me
(very limited engineer) &#8211; and definitely I am looking for objections =
even more
then for agreement &#8211; for the sake of either finding the idea useful o=
r making
judged deny of it, and I agree that this concept is only one possible way o=
f
working with congestion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>best regards,</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Slava Borilin</span></font><o:p></o:p>=
</p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; Jean-Ma=
rc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span lang=3DR=
U
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Henry,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>With all of my respect (you know) &#82=
11; this
is too high-level consideration and too generic statement which doesn&#8217=
;t answer
MPLS issue for example.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>best regards,</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Slava Borilin</span></font><o:p></o:p>=
</p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Henry Si=
nnreich
[mailto:hsinnrei@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span lang=3DR=
U
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 face=3DC=
alibri><span
lang=3DRU style=3D'font-size:18.0pt;font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<br>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its only =
one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:</span><=
/font><span
lang=3DRU><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D5 face=3DC=
alibri><span
lang=3DRU style=3D'font-size:18.0pt;font-family:Calibri'>I am not that sure=
 that
routers should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, why<br=
>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it can<br=
>
be too hard and too long to tell each sender to reduce the rate - why<br>
not to leave this freedom to router as this may be more efficient for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router do<br>
not need to understand anything except that it is LAYERED streams (even<br>
don't know whether its voice, video or audio or anything else of layered<br=
>
nature, and router &quot;has the right&quot; to strip out layers in certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] O=
n
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you don't<br>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
br>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<br=
>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt
to<br>
be<br>
careful not to include in the codec things that don't belong there (e.g.<br=
>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at some<=
br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! Routers<b=
r>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
br>
&gt; packets to react appropriately to the marks or losses. In other words,=
<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not in<br=
>
&gt; response to whatever congestion level those losses/marks indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and this<br=
>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You don't<br>
need layers to have a multi-rate codec. Layering just means that you can<br=
>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* losses/marks<b=
r>
to<br>
&gt; determine capacity. People have tried to create congestion control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable -<b=
r>
the<br>
&gt; rate computation is usually clocked by the RTT. You can obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying that<br>
you<br>
can have one module that determines in real-time the best rate to use at<br=
>
<br>
that moment and then it tells the codec what to use. If it needs help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how to=
<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<b=
r>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a></span></font><span
lang=3DRU><o:p></o:p></span></p>

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From stewe@stewe.org  Mon Oct 26 11:15:12 2009
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Date: Mon, 26 Oct 2009 11:15:00 -0700
From: Stephan Wenger <stewe@stewe.org>
To: Jozsef Vass <jovass@adobe.com>, Slava Borilin <Borilin@spiritdsp.com>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
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Thread-Topic: [codec] updated charter proposal
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Subject: Re: [codec] updated charter proposal
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I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is
they don=B9t modify IP packets.  Boxes that look deep into the IP packets, an=
d
(potentially) modify them, are higher layer devices.  After long discussion=
s
in AVT at the time, RFC 3984 and its successors call those boxes =B3Media
Aware Network Elements=B2, MANEs.  MANEs modify packets at the application
layer, for example to remove layers, re-package H.264 NAL units to adapt to
different MTU sizes, and do all kinds of other things that violate the
end-to-end principle.  A MANE has business in looking into, and potentially
modifying, IP/UDP/RTP packets, a router has not.

When MANEs are in the picture, it=B9s IMO totally acceptable (from both an
abstract layering theory viewpoint and in practice) that packets are being
opened and the content is repackaged.  It=B9s actually common practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:

> I do not like the idea of sending multiple layers in a packet either. I f=
eel
> very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but
> delivering semantically modified packet is not expected.
> =20
> Using layers for video totally makes sense where the rates are much highe=
r and
> layers can be transported in separate packets.
> =20
> Jozsef
> =20
>=20
>=20
> From: Slava Borilin [mailto:Borilin@spiritdsp.com]
> Sent: Thursday, October 22, 2009 12:23 PM
> To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> =20
> Again - my point is that router does NOT have to know that something
> specifically is the codec, or specific codec- the only goal is to let rou=
ter
> aware that the stream (and every packet) is LAYERED, and that it is legit=
imate
> to strip out some layers if necessary due to congestion.
> The content of such =B3layered=B2 can be any =AD voice codec, video codec, audi=
o
> codec, any type of content.
> =20
> And I agree with Gregory =AD that having current infrastructure using layer=
s in
> separate streams will require too much overhead (packets) which does not =
make
> sense as well =AD and I am thinking about one stream of packets comtaining =
all
> the layers, which, however, can be modified on the fly =B3in the middle=B2.
> So we may think about more generic concept of =B3declared layered content=B2 =
which
> does NOT loose the information (well, most of it) by stripping out portio=
ns of
> the packet and reducing the traffic, so router can strip out any type of
> layered content, without going deep into specific codec.
> =20
> Again- this is =B3na=EFve=B2 suggestions from me (very limited engineer) =AD and
> definitely I am looking for objections even more then for agreement =AD for=
 the
> sake of either finding the idea useful or making judged deny of it, and I
> agree that this concept is only one possible way of working with congesti=
on.
> =20
>=20
> best regards,
> Slava Borilin
>=20
>=20
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
> Slava Borilin
> Sent: Thursday, October 22, 2009 11:13 PM
> To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> =20
> Henry,
> With all of my respect (you know) =AD this is too high-level consideration =
and
> too generic statement which doesn=B9t answer MPLS issue for example.
> =20
>=20
> best regards,
> Slava Borilin
>=20
>=20
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> Sent: Thursday, October 22, 2009 11:10 PM
> To: Slava Borilin; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> =20
> Slava,
>=20
>> >I am not that sure that routers should not be aware of the codecs. Why?
>> >If we are building MPLS and other systems that is content-dependent, wh=
y
>> >not let router understand the codec?
>=20
> The essence of the Dumb Network (Internet) and its routers is that it kno=
ws
> nothing about the applications.
> All applications that nobody planned for can run over it.
> The tragedy of the telephone network was is knew everything about its onl=
y one
> application.
>=20
> Henry
>=20
>=20
> On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
> I am not that sure that routers should not be aware of the codecs. Why?
> If we are building MPLS and other systems that is content-dependent, why
> not let router understand the codec?
>=20
> Especially if we have hordes of voice channels going thru router, it can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?
>=20
> Same applies to H.264SVC for example - at the end of the day, router do
> not need to understand anything except that it is LAYERED streams (even
> don't know whether its voice, video or audio or anything else of layered
> nature, and router "has the right" to strip out layers in certain order
> to prevent congestion?
>=20
> best regards,
> Slava Borilin
>=20
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Thursday, October 22, 2009 5:45 PM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Lars Eggert wrote:
>> > Also, what we've learned from media over DCCP is that if you don't
>> > involve the codec in some form, performance/quality will be bad. Some
>> > codecs need time to adjust the rate. Some can only adjust it in pretty
>=20
>> > coarse steps. Etc. If you let DCCP adjust the rate blindly, without
>> > involving the codec in some form, you might get very good TCP
>> > friendliness, but maybe quality will suffer.
>=20
> I'm totally in favour of "involving the codec" in here. I just want to
> be
> careful not to include in the codec things that don't belong there (e.g.
> a
> DCCP implementation or a whole jitter buffer).
>=20
>> > (And there is always the ramping up part - congestion control will
> never
>> > attempt to probe for more capacity unless the codec generates more
> data,
>> > and why would the codec do that if it's converged to transmit at some
>> > lower rate.)
>=20
> Well, the codec transmits at the rate you ask it to use. If you want to
> probe for more capacity, then you tell the codec that it can use a
> higher
> rate. Now, if there's a special way in which you want to ask the codec,
> then maybe that can become an additional feature for the codec.
>=20
>> > By no means should routers have to understand codec formats! Routers
>> > drop or mark IP packets. It is up to the application sending those IP
>> > packets to react appropriately to the marks or losses. In other words,
> a
>> > layered codec will see losses/marks to packets sent on all layers, and
>=20
>> > will then need to figure out whether to strip off a layer or not in
>> > response to whatever congestion level those losses/marks indicate.
>=20
> I definitely agree that the routers shouldn't know about codecs and this
> is
> why I was saying that IMO layered codecs aren't too useful for
> congestion
> control (though they have other uses). If there's nothing in the middle
> of
> the network to strip off the layers, then you might as well not have
> layers
> at all. All you need is to ask the sender to reduce the rate. You don't
> need layers to have a multi-rate codec. Layering just means that you can
>=20
> strip off bits and still have the result sound OK.
>=20
>>> >> OK, I may be wrong here, but my understanding is that how much the
> rate
>>> >> should change depending on congestion is actually not that
>>> >> codec-specific.
>> >
>> > It *can* be non-codec-specific, see DCCP for example. But quality will
>=20
>> > suffer IMO.
>=20
> Well, we need to figure out the aspects that should be codec-specific
> and
> implement just those. Does anyone have a list of the issues codecs
> should
> be aware of?
>=20
>> > Most Internet congestion control is based on *creating* losses/marks
> to
>> > determine capacity. People have tried to create congestion control
>> > schemes that avoid causing losses (delay-based schemed, for example),
>> > but they are perceived to be more fragile.
>=20
> Yes, I can imagine the problem not being trivial at all.
>=20
>>> >> Once you figure this out, you
>>> >> can just tell the codec to use that rate.
>> >
>> > It's not that easy, because rates in the Internet are never stable -
> the
>> > rate computation is usually clocked by the RTT. You can obviously
> dampen
>> > this somehow, but you can never say "now I have found a rate that I
> can
>> > safely use forever."
>=20
> Oh, I never said we should set a rate forever. I was just saying that
> you
> can have one module that determines in real-time the best rate to use at
>=20
> that moment and then it tells the codec what to use. If it needs help
> from
> the codec, then we can define special functionality.
>=20
>>> >> On top of that, the codec may be
>>> >> able to do a bit better by providing a "service" to the other layers=
.
> On
>>> >> example here (also included in the draft) is VBR, which means that
> the
>>> >> application specifies an average rate and the codec decides how to
>>> >> vary the
>>> >> rate to achieve that (subject to some rate control constraints).
> Would
>>> >> other such mechanisms be useful to deal with congestion?
>> >
>> > I'm not sure this would work, because congestion control will
> basically
>> > change the VBR input parameter continuously.
>=20
> Changing the VBR parameters continuously is not a problem. VBR just
> means
> that you'd like to have an average of N kb/s for now, but you're OK if
> there are variations from frame to frame.
>=20
>         Jean-Marc
> _______________________________________________
> 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
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


--B_3339400508_2129852
Content-type: text/html;
	charset="ISO-8859-1"
Content-transfer-encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:11pt=
'>I think that we have a terminology problem here.<BR>
<BR>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP pac=
kets, and (potentially) modify them, are higher layer devices. &nbsp;After l=
ong discussions in AVT at the time, RFC 3984 and its successors call those b=
oxes &#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify p=
ackets at the application layer, for example to remove layers, re-package H.=
264 NAL units to adapt to different MTU sizes, and do all kinds of other thi=
ngs that violate the end-to-end principle. &nbsp;A MANE has business in look=
ing into, and potentially modifying, IP/UDP/RTP packets, a router has not.<B=
R>
<BR>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from both=
 an abstract layering theory viewpoint and in practice) that packets are bei=
ng opened and the content is repackaged. &nbsp;It&#8217;s actually common pr=
actice.<BR>
<BR>
Stephan<BR>
<BR>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.com=
">jovass@adobe.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"=
Arial"><SPAN STYLE=3D'font-size:10pt'>I do not like the idea of sending multip=
le layers in a packet either. I feel very uncomfortable of routers modifying=
 packets. It is OK not to deliver, but delivering semantically modified pack=
et is not expected.<BR>
&nbsp;<BR>
Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.<BR>
&nbsp;<BR>
Jozsef<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> Slava Borilin [<a href=3D"mailto:Borilin@spiritds=
p.com">mailto:Borilin@spiritdsp.com</a>] <BR>
<B>Sent:</B> Thursday, October 22, 2009 12:23 PM<BR>
<B>To:</B> Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'> <BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>Again - my point is that router does NOT have to kno=
w that something specifically is the codec, or specific codec- the only goal=
 is to let router aware that the stream (and every packet) is LAYERED, and t=
hat it is legitimate to strip out some layers if necessary due to congestion=
.<BR>
The content of such &#8220;layered&#8221; can be any &#8211; voice codec, v=
ideo codec, audio codec, any type of content.<BR>
&nbsp;<BR>
And I agree with Gregory &#8211; that having current infrastructure using l=
ayers in separate streams will require too much overhead (packets) which doe=
s not make sense as well &#8211; and I am thinking about one stream of packe=
ts comtaining all the layers, which, however, can be modified on the fly &#8=
220;in the middle&#8221;.<BR>
So we may think about more generic concept of &#8220;declared layered conte=
nt&#8221; which does NOT loose the information (well, most of it) by strippi=
ng out portions of the packet and reducing the traffic, so router can strip =
out any type of layered content, without going deep into specific codec.<BR>
&nbsp;<BR>
Again- this is &#8220;na&iuml;ve&#8221; suggestions from me (very limited e=
ngineer) &#8211; and definitely I am looking for objections even more then f=
or agreement &#8211; for the sake of either finding the idea useful or makin=
g judged deny of it, and I agree that this concept is only one possible way =
of working with congestion.<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>best regards,<BR>
Slava Borilin<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> <a href=3D"codec-bounces@ietf.org">codec-bounces@=
ietf.org</a> [<a href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@i=
etf.org</a>] <B>On Behalf Of </B>Slava Borilin<BR>
<B>Sent:</B> Thursday, October 22, 2009 11:13 PM<BR>
<B>To:</B> Henry Sinnreich; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'> <BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>Henry,<BR>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion and too generic statement which doesn&#8217;t answer MPLS issue for exam=
ple.<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>best regards,<BR>
Slava Borilin<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> Henry Sinnreich [<a href=3D"mailto:hsinnrei@adobe=
.com">mailto:hsinnrei@adobe.com</a>] <BR>
<B>Sent:</B> Thursday, October 22, 2009 11:10 PM<BR>
<B>To:</B> Slava Borilin; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'> <BR>
</SPAN></FONT><FONT SIZE=3D"5"><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial=
"><SPAN STYLE=3D'font-size:18pt'>Slava,<BR>
<BR>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<BR>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<BR>
&gt;not let router understand the codec?<BR>
<BR>
The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.<BR>
All applications that nobody planned for can run over it.<BR>
The tragedy of the telephone network was is knew everything about its only =
one application.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spiritd=
sp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
I am not that sure that routers should not be aware of the codecs. Why?<BR>
If we are building MPLS and other systems that is content-dependent, why<BR=
>
not let router understand the codec?<BR>
<BR>
Especially if we have hordes of voice channels going thru router, it can<BR=
>
be too hard and too long to tell each sender to reduce the rate - why<BR>
not to leave this freedom to router as this may be more efficient for<BR>
congestion recovery?<BR>
<BR>
Same applies to H.264SVC for example - at the end of the day, router do<BR>
not need to understand anything except that it is LAYERED streams (even<BR>
don't know whether its voice, video or audio or anything else of layered<BR=
>
nature, and router &quot;has the right&quot; to strip out layers in certain=
 order<BR>
to prevent congestion?<BR>
<BR>
best regards,<BR>
Slava Borilin<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a href=3D=
"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] On Behalf=
<BR>
Of Jean-Marc Valin<BR>
Sent: Thursday, October 22, 2009 5:45 PM<BR>
To: Lars Eggert<BR>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
Subject: Re: [codec] updated charter proposal<BR>
<BR>
Lars Eggert wrote:<BR>
&gt; Also, what we've learned from media over DCCP is that if you don't<BR>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
BR>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<BR>
<BR>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<BR=
>
&gt; involving the codec in some form, you might get very good TCP<BR>
&gt; friendliness, but maybe quality will suffer.<BR>
<BR>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt to<BR>
be<BR>
careful not to include in the codec things that don't belong there (e.g.<BR=
>
a<BR>
DCCP implementation or a whole jitter buffer).<BR>
<BR>
&gt; (And there is always the ramping up part - congestion control will<BR>
never<BR>
&gt; attempt to probe for more capacity unless the codec generates more<BR>
data,<BR>
&gt; and why would the codec do that if it's converged to transmit at some<=
BR>
&gt; lower rate.)<BR>
<BR>
Well, the codec transmits at the rate you ask it to use. If you want to<BR>
probe for more capacity, then you tell the codec that it can use a<BR>
higher<BR>
rate. Now, if there's a special way in which you want to ask the codec,<BR>
then maybe that can become an additional feature for the codec.<BR>
<BR>
&gt; By no means should routers have to understand codec formats! Routers<B=
R>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
BR>
&gt; packets to react appropriately to the marks or losses. In other words,=
<BR>
a<BR>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<BR>
<BR>
&gt; will then need to figure out whether to strip off a layer or not in<BR=
>
&gt; response to whatever congestion level those losses/marks indicate.<BR>
<BR>
I definitely agree that the routers shouldn't know about codecs and this<BR=
>
is<BR>
why I was saying that IMO layered codecs aren't too useful for<BR>
congestion<BR>
control (though they have other uses). If there's nothing in the middle<BR>
of<BR>
the network to strip off the layers, then you might as well not have<BR>
layers<BR>
at all. All you need is to ask the sender to reduce the rate. You don't<BR>
need layers to have a multi-rate codec. Layering just means that you can<BR=
>
<BR>
strip off bits and still have the result sound OK.<BR>
<BR>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<BR>
rate<BR>
&gt;&gt; should change depending on congestion is actually not that<BR>
&gt;&gt; codec-specific.<BR>
&gt;<BR>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<BR>
<BR>
&gt; suffer IMO.<BR>
<BR>
Well, we need to figure out the aspects that should be codec-specific<BR>
and<BR>
implement just those. Does anyone have a list of the issues codecs<BR>
should<BR>
be aware of?<BR>
<BR>
&gt; Most Internet congestion control is based on *creating* losses/marks<B=
R>
to<BR>
&gt; determine capacity. People have tried to create congestion control<BR>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
BR>
&gt; but they are perceived to be more fragile.<BR>
<BR>
Yes, I can imagine the problem not being trivial at all.<BR>
<BR>
&gt;&gt; Once you figure this out, you<BR>
&gt;&gt; can just tell the codec to use that rate.<BR>
&gt;<BR>
&gt; It's not that easy, because rates in the Internet are never stable -<B=
R>
the<BR>
&gt; rate computation is usually clocked by the RTT. You can obviously<BR>
dampen<BR>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<BR>
can<BR>
&gt; safely use forever.&quot;<BR>
<BR>
Oh, I never said we should set a rate forever. I was just saying that<BR>
you<BR>
can have one module that determines in real-time the best rate to use at<BR=
>
<BR>
that moment and then it tells the codec what to use. If it needs help<BR>
from<BR>
the codec, then we can define special functionality.<BR>
<BR>
&gt;&gt; On top of that, the codec may be<BR>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the =
other layers.<BR>
On<BR>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<BR>
the<BR>
&gt;&gt; application specifies an average rate and the codec decides how to=
<BR>
&gt;&gt; vary the<BR>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<B=
R>
Would<BR>
&gt;&gt; other such mechanisms be useful to deal with congestion?<BR>
&gt;<BR>
&gt; I'm not sure this would work, because congestion control will<BR>
basically<BR>
&gt; change the VBR input parameter continuously.<BR>
<BR>
Changing the VBR parameters continuously is not a problem. VBR just<BR>
means<BR>
that you'd like to have an average of N kb/s for now, but you're OK if<BR>
there are variations from frame to frame.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org/=
mailman/listinfo/codec</a><BR>
_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org/=
mailman/listinfo/codec</a><BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:11pt'><BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"95%"></SPAN></FONT><FONT SIZE=3D"2"><FONT FA=
CE=3D"Consolas, Courier New, Courier"><SPAN STYLE=3D'font-size:10pt'>___________=
____________________________________<BR>
codec mailing list<BR>
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<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org/=
mailman/listinfo/codec</a><BR>
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From: Michael Knappe <mknappe@juniper.net>
To: "bens@alum.mit.edu" <bens@alum.mit.edu>, Slava Borilin <Borilin@spiritdsp.com>
Date: Mon, 26 Oct 2009 13:47:12 -0700
Thread-Topic: [codec] codec tandeming/codec-requirements
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Benjamin,

Well worded paragraph, thanks. I would add just some minor edits and a scop=
ing sentence at the end. My proposed changes are in italics.

Cheers,

Mike

Audio Corruption Robustness

The systems providing input to the encoder and s//receiving/ output from th=
e decoder s/are
likely to/may/ be far from ideal in actual use.  Input and output audio str=
eams
may be corrupted by s//compounding non-linear/ artifacts from analog hardwa=
re and digital processing. The codecs to be developed should be tested to e=
nsure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion. s//No specif=
ic end-to-end quality requirements are mandated for use with the proposed c=
odec. It is advisable, however, that several typical in-situ environments/p=
rocessing chains be specified for the purpose of benchmarking end-to-end qu=
ality with the proposed codec./
"""

"




On 10/23/09 6:29 AM, "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu> wrot=
e:

Slava Borilin wrote:
> Henry, e2e (or p2p) is cool but what about Legal interception for
> example in this case, which every cellco has to implemenet?

Whether or not we want to encourage it, interception of VoIP traffic does
not require a re-encode.  The encoded packets can just be copied to the
additional receiver.

To address the "tandeming" issue, I think the following text should be
added to the "Basic Requirements" or "Additional Considerations":

"""
Audio Corruption Robustness

The systems providing input to the encoder and output from the decoder are
likely to be far from ideal in actual use.  Input and output audio streams
may be corrupted by artifacts from analog hardware and digital processing.
 The codecs to be developed should be tested to ensure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion.
"""



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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Stephan Wenger" <stewe@stewe.org>, "Jozsef Vass" <jovass@adobe.com>, "Henry Sinnreich" <hsinnrei@adobe.com>, "Jean-Marc Valin" <jean-marc.valin@octasic.com>, "Lars Eggert" <lars.eggert@nokia.com>
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Well, if we change "routers" to MANEs, the whole story of layering start =
to make more sense?

=20

best regards,

Slava Borilin

________________________________

From: Stephan Wenger [mailto:stewe@stewe.org]=20
Sent: Monday, October 26, 2009 9:15 PM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars =
Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they don't modify IP packets.  Boxes that look deep into the IP packets, =
and (potentially) modify them, are higher layer devices.  After long =
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes "Media Aware Network Elements", MANEs.  MANEs modify packets at =
the application layer, for example to remove layers, re-package H.264 =
NAL units to adapt to different MTU sizes, and do all kinds of other =
things that violate the end-to-end principle.  A MANE has business in =
looking into, and potentially modifying, IP/UDP/RTP packets, a router =
has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an =
abstract layering theory viewpoint and in practice) that packets are =
being opened and the content is repackaged.  It's actually common =
practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:

I do not like the idea of sending multiple layers in a packet either. I =
feel very uncomfortable of routers modifying packets. It is OK not to =
deliver, but delivering semantically modified packet is not expected.
=20
Using layers for video totally makes sense where the rates are much =
higher and layers can be transported in separate packets.
=20
Jozsef
=20

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]=20
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something =
specifically is the codec, or specific codec- the only goal is to let =
router aware that the stream (and every packet) is LAYERED, and that it =
is legitimate to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, =
audio codec, any type of content.
=20
And I agree with Gregory - that having current infrastructure using =
layers in separate streams will require too much overhead (packets) =
which does not make sense as well - and I am thinking about one stream =
of packets comtaining all the layers, which, however, can be modified on =
the fly "in the middle".
So we may think about more generic concept of "declared layered content" =
which does NOT loose the information (well, most of it) by stripping out =
portions of the packet and reducing the traffic, so router can strip out =
any type of layered content, without going deep into specific codec.
=20
Again- this is "na=EFve" suggestions from me (very limited engineer) - =
and definitely I am looking for objections even more then for agreement =
- for the sake of either finding the idea useful or making judged deny =
of it, and I agree that this concept is only one possible way of working =
with congestion.
=20

best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Slava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration =
and too generic statement which doesn't answer MPLS issue for example.
=20

best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, =
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it =
knows nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its =
only one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Well, if we =
change &#8220;routers&#8221;
to MANEs, the whole story of layering start to make more =
sense?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,</span></font><font
color=3Dnavy><span lang=3DEN-US =
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Stephan Wenger [mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, =
2009
9:15 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava =
Borilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri'>I think that we have a =
terminology
problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP =
packets,
and (potentially) modify them, are higher layer devices. &nbsp;After =
long
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify =
packets
at the application layer, for example to remove layers, re-package H.264 =
NAL
units to adapt to different MTU sizes, and do all kinds of other things =
that
violate the end-to-end principle. &nbsp;A MANE has business in looking =
into,
and potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from =
both an
abstract layering theory viewpoint and in practice) that packets are =
being
opened and the content is repackaged. &nbsp;It&#8217;s actually common
practice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a =
href=3D"jovass@adobe.com">jovass@adobe.com</a>&gt;
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do not like the idea of sending =
multiple
layers in a packet either. I feel very uncomfortable of routers =
modifying
packets. It is OK not to deliver, but delivering semantically modified =
packet
is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much =
higher and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Borilin [<a
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>] =
<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'>Again - my point is that router does NOT have to know =
that
something specifically is the codec, or specific codec- the only goal is =
to let
router aware that the stream (and every packet) is LAYERED, and that it =
is
legitimate to strip out some layers if necessary due to congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice =
codec, video
codec, audio codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure =
using
layers in separate streams will require too much overhead (packets) =
which does
not make sense as well &#8211; and I am thinking about one stream of =
packets
comtaining all the layers, which, however, can be modified on the fly =
&#8220;in
the middle&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered
content&#8221; which does NOT loose the information (well, most of it) =
by
stripping out portions of the packet and reducing the traffic, so router =
can
strip out any type of layered content, without going deep into specific =
codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited =
engineer)
&#8211; and definitely I am looking for objections even more then for =
agreement
&#8211; for the sake of either finding the idea useful or making judged =
deny of
it, and I agree that this concept is only one possible way of working =
with
congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> <a =
href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a>
[<a =
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level =
consideration
and too generic statement which doesn&#8217;t answer MPLS issue for =
example.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Henry
Sinnreich [<a =
href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</a>] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><br>
<font size=3D5 face=3DCalibri><span =
style=3D'font-size:18.0pt;font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. =
Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
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_______________________________________________<br>
codec mailing list<br>
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/mailman/listinfo/codec</a></span></font><font
size=3D2 face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:Calibri'><o:p></o:p></span></font><=
/p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D2
face=3DCalibri><span style=3D'font-size:11.0pt;font-family:Calibri'>

<hr size=3D3 width=3D"95%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal><font size=3D2 face=3DConsolas><span =
style=3D'font-size:10.0pt;
font-family:Consolas'>_______________________________________________<br>=

codec mailing list<br>
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From mknappe@juniper.net  Mon Oct 26 14:00:18 2009
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From: Michael Knappe <mknappe@juniper.net>
To: Michael Knappe <mknappe@juniper.net>, "bens@alum.mit.edu" <bens@alum.mit.edu>, Slava Borilin <Borilin@spiritdsp.com>
Date: Mon, 26 Oct 2009 13:58:22 -0700
Thread-Topic: [codec] codec tandeming/codec-requirements
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Subject: Re: [codec] codec tandeming/codec-requirements
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For ease of reading, incorporating the edits it would read as:

The systems providing input to the encoder and receiving output from the de=
coder may be far from ideal in actual use.  Input and output audio streams =
may be corrupted by compounding non-linear artifacts from analog hardware a=
nd digital processing. The codecs to be developed should be tested to ensur=
e that they degrade gracefully under adverse audio conditions.  Types of di=
gital corruption that may be tested include tandem encoding with similar an=
d different codecs, low-quality resampling, and digital clipping.  Types of=
 analog corruption that may be tested include microphones with substantial =
background noise, analog clipping, and loudspeaker distortion. No specific =
end-to-end quality requirements are mandated for use with the proposed code=
c. It is advisable, however, that several typical in-situ environments/proc=
essing chains be specified for the purpose of benchmarking end-to-end quali=
ty with the proposed codec.


On 10/26/09 1:47 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

Benjamin,

Well worded paragraph, thanks. I would add just some minor edits and a scop=
ing sentence at the end. My proposed changes are in italics.

Cheers,

Mike

Audio Corruption Robustness

The systems providing input to the encoder and s//receiving/ output from th=
e decoder s/are
likely to/may/ be far from ideal in actual use.  Input and output audio str=
eams
may be corrupted by s//compounding non-linear/ artifacts from analog hardwa=
re and digital processing. The codecs to be developed should be tested to e=
nsure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion. s//No specif=
ic end-to-end quality requirements are mandated for use with the proposed c=
odec. It is advisable, however, that several typical in-situ environments/p=
rocessing chains be specified for the purpose of benchmarking end-to-end qu=
ality with the proposed codec./
"""

"




On 10/23/09 6:29 AM, "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu> wrot=
e:

Slava Borilin wrote:
> Henry, e2e (or p2p) is cool but what about Legal interception for
> example in this case, which every cellco has to implemenet?

Whether or not we want to encourage it, interception of VoIP traffic does
not require a re-encode.  The encoded packets can just be copied to the
additional receiver.

To address the "tandeming" issue, I think the following text should be
added to the "Basic Requirements" or "Additional Considerations":

"""
Audio Corruption Robustness

The systems providing input to the encoder and output from the decoder are
likely to be far from ideal in actual use.  Input and output audio streams
may be corrupted by artifacts from analog hardware and digital processing.
 The codecs to be developed should be tested to ensure that they degrade
gracefully under adverse audio conditions.  Types of digital corruption
that may be tested include tandem encoding with similar and different
codecs, low-quality resampling, and digital clipping.  Types of analog
corruption that may be tested include microphones with substantial
background noise, analog clipping, and loudspeaker distortion.
"""


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


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From: Jozsef Vass <jovass@adobe.com>
To: Stephan Wenger <stewe@stewe.org>, Slava Borilin <Borilin@spiritdsp.com>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
Date: Mon, 26 Oct 2009 14:09:01 -0700
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In rfc 3984 processing packets by mane has some clear benefits, e.g., disca=
rding packets that the decoder would describe anyways (section 7.3), droppi=
ng packets in case of congestion, etc.

Here, we mostly talking about reducing packet size, and I am not sure how b=
eneficial it is regarding the overall system. Consider speex. Quality level=
 six, wideband and narrowband encoder produces 52 bytes and 28 bytes of pay=
load, respectively. Adding IP/UDP/RTP header, you go from 92 bytes to 68 by=
tes. Since packet rate is the same, I do not think it will affect congestio=
n too much. Please also consider that processing packets by intermediary, a=
dds additional delay.

Jozsef

________________________________
From: Stephan Wenger [mailto:stewe@stewe.org]
Sent: Monday, October 26, 2009 11:15 AM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Egge=
rt
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey don't modify IP packets.  Boxes that look deep into the IP packets, and =
(potentially) modify them, are higher layer devices.  After long discussion=
s in AVT at the time, RFC 3984 and its successors call those boxes "Media A=
ware Network Elements", MANEs.  MANEs modify packets at the application lay=
er, for example to remove layers, re-package H.264 NAL units to adapt to di=
fferent MTU sizes, and do all kinds of other things that violate the end-to=
-end principle.  A MANE has business in looking into, and potentially modif=
ying, IP/UDP/RTP packets, a router has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an ab=
stract layering theory viewpoint and in practice) that packets are being op=
ened and the content is repackaged.  It's actually common practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:
I do not like the idea of sending multiple layers in a packet either. I fee=
l very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but delivering semantically modified packet is not expected.

Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.

Jozsef

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something specif=
ically is the codec, or specific codec- the only goal is to let router awar=
e that the stream (and every packet) is LAYERED, and that it is legitimate =
to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, audio =
codec, any type of content.

And I agree with Gregory - that having current infrastructure using layers =
in separate streams will require too much overhead (packets) which does not=
 make sense as well - and I am thinking about one stream of packets comtain=
ing all the layers, which, however, can be modified on the fly "in the midd=
le".
So we may think about more generic concept of "declared layered content" wh=
ich does NOT loose the information (well, most of it) by stripping out port=
ions of the packet and reducing the traffic, so router can strip out any ty=
pe of layered content, without going deep into specific codec.

Again- this is "na=EFve" suggestions from me (very limited engineer) - and =
definitely I am looking for objections even more then for agreement - for t=
he sake of either finding the idea useful or making judged deny of it, and =
I agree that this concept is only one possible way of working with congesti=
on.


best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of S=
lava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration an=
d too generic statement which doesn't answer MPLS issue for example.


best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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

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

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>In rfc 3984 processing packets by mane=
 has
some clear benefits, e.g., discarding packets that the decoder would descri=
be
anyways (section 7.3), dropping packets in case of congestion, etc.<o:p></o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Here, we mostly talking about reducing
packet size, and I am not sure how beneficial it is regarding the overall s=
ystem.
Consider speex. Quality level six, wideband and narrowband encoder produces=
 52 bytes
and 28 bytes of payload, respectively. Adding IP/UDP/RTP header, you go fro=
m 92
bytes to 68 bytes. Since packet rate is the same, I do not think it will af=
fect
congestion too much. Please also consider that processing packets by interm=
ediary,
adds additional delay.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Stephan =
Wenger
[mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, 20=
09
11:15 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava Boril=
in;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 face=3DC=
alibri><span
style=3D'font-size:11.0pt;font-family:Calibri'>I think that we have a termi=
nology
problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP packe=
ts,
and (potentially) modify them, are higher layer devices. &nbsp;After long
discussions in AVT at the time, RFC 3984 and its successors call those boxe=
s
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify packe=
ts
at the application layer, for example to remove layers, re-package H.264 NA=
L units
to adapt to different MTU sizes, and do all kinds of other things that viol=
ate
the end-to-end principle. &nbsp;A MANE has business in looking into, and
potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from both=
 an
abstract layering theory viewpoint and in practice) that packets are being
opened and the content is repackaged. &nbsp;It&#8217;s actually common
practice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.c=
om">jovass@adobe.com</a>&gt;
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do not like the idea of sending mult=
iple
layers in a packet either. I feel very uncomfortable of routers modifying
packets. It is OK not to deliver, but delivering semantically modified pack=
et
is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much higher =
and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Borilin [<a
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>] <br=
>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Again - my point is that router does NOT have to know tha=
t
something specifically is the codec, or specific codec- the only goal is to=
 let
router aware that the stream (and every packet) is LAYERED, and that it is
legitimate to strip out some layers if necessary due to congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice codec, v=
ideo
codec, audio codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure using
layers in separate streams will require too much overhead (packets) which d=
oes
not make sense as well &#8211; and I am thinking about one stream of packet=
s
comtaining all the layers, which, however, can be modified on the fly &#822=
0;in
the middle&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered
content&#8221; which does NOT loose the information (well, most of it) by s=
tripping
out portions of the packet and reducing the traffic, so router can strip ou=
t
any type of layered content, without going deep into specific codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited engi=
neer)
&#8211; and definitely I am looking for objections even more then for agree=
ment
&#8211; for the sake of either finding the idea useful or making judged den=
y of
it, and I agree that this concept is only one possible way of working with
congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> <a href=3D"codec-bounces@iet=
f.org">codec-bounces@ietf.org</a>
[<a href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a=
>] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; Jean-Ma=
rc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion
and too generic statement which doesn&#8217;t answer MPLS issue for example=
.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span>=
</font></b><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>=
 Henry
Sinnreich [<a href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com<=
/a>] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D5 face=3DCalibri><span style=3D'font-size:18.0pt;font-family:C=
alibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<br>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its only =
one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, why<br=
>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it can<br=
>
be too hard and too long to tell each sender to reduce the rate - why<br>
not to leave this freedom to router as this may be more efficient for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router do<br>
not need to understand anything except that it is LAYERED streams (even<br>
don't know whether its voice, video or audio or anything else of layered<br=
>
nature, and router &quot;has the right&quot; to strip out layers in certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] O=
n
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you don't<br>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
br>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<br=
>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt
to<br>
be<br>
careful not to include in the codec things that don't belong there (e.g.<br=
>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at some<=
br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! Routers<b=
r>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
br>
&gt; packets to react appropriately to the marks or losses. In other words,=
<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not in<br=
>
&gt; response to whatever congestion level those losses/marks indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and this<br=
>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You don't<br>
need layers to have a multi-rate codec. Layering just means that you can<br=
>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* losses/marks<b=
r>
to<br>
&gt; determine capacity. People have tried to create congestion control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable -<b=
r>
the<br>
&gt; rate computation is usually clocked by the RTT. You can obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying that<br>
you<br>
can have one module that determines in real-time the best rate to use at<br=
>
<br>
that moment and then it tells the codec what to use. If it needs help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how to=
<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<b=
r>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
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g/mailman/listinfo/codec</a><br>
_______________________________________________<br>
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Jozsef Vass" <jovass@adobe.com>, "Stephan Wenger" <stewe@stewe.org>, "Henry Sinnreich" <hsinnrei@adobe.com>, "Jean-Marc Valin" <jean-marc.valin@octasic.com>, "Lars Eggert" <lars.eggert@nokia.com>
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Speex is not layered, isn't it?

=20

best regards,

Slava Borilin

________________________________

From: Jozsef Vass [mailto:jovass@adobe.com]=20
Sent: Tuesday, October 27, 2009 12:09 AM
To: Stephan Wenger; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; =
Lars Eggert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

=20

In rfc 3984 processing packets by mane has some clear benefits, e.g., =
discarding packets that the decoder would describe anyways (section =
7.3), dropping packets in case of congestion, etc.

=20

Here, we mostly talking about reducing packet size, and I am not sure =
how beneficial it is regarding the overall system. Consider speex. =
Quality level six, wideband and narrowband encoder produces 52 bytes and =
28 bytes of payload, respectively. Adding IP/UDP/RTP header, you go from =
92 bytes to 68 bytes. Since packet rate is the same, I do not think it =
will affect congestion too much. Please also consider that processing =
packets by intermediary, adds additional delay.

=20

Jozsef

=20

________________________________

From: Stephan Wenger [mailto:stewe@stewe.org]=20
Sent: Monday, October 26, 2009 11:15 AM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars =
Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they don't modify IP packets.  Boxes that look deep into the IP packets, =
and (potentially) modify them, are higher layer devices.  After long =
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes "Media Aware Network Elements", MANEs.  MANEs modify packets at =
the application layer, for example to remove layers, re-package H.264 =
NAL units to adapt to different MTU sizes, and do all kinds of other =
things that violate the end-to-end principle.  A MANE has business in =
looking into, and potentially modifying, IP/UDP/RTP packets, a router =
has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an =
abstract layering theory viewpoint and in practice) that packets are =
being opened and the content is repackaged.  It's actually common =
practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:

I do not like the idea of sending multiple layers in a packet either. I =
feel very uncomfortable of routers modifying packets. It is OK not to =
deliver, but delivering semantically modified packet is not expected.
=20
Using layers for video totally makes sense where the rates are much =
higher and layers can be transported in separate packets.
=20
Jozsef
=20

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]=20
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something =
specifically is the codec, or specific codec- the only goal is to let =
router aware that the stream (and every packet) is LAYERED, and that it =
is legitimate to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, =
audio codec, any type of content.
=20
And I agree with Gregory - that having current infrastructure using =
layers in separate streams will require too much overhead (packets) =
which does not make sense as well - and I am thinking about one stream =
of packets comtaining all the layers, which, however, can be modified on =
the fly "in the middle".
So we may think about more generic concept of "declared layered content" =
which does NOT loose the information (well, most of it) by stripping out =
portions of the packet and reducing the traffic, so router can strip out =
any type of layered content, without going deep into specific codec.
=20
Again- this is "na=EFve" suggestions from me (very limited engineer) - =
and definitely I am looking for objections even more then for agreement =
- for the sake of either finding the idea useful or making judged deny =
of it, and I agree that this concept is only one possible way of working =
with congestion.
=20

best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Slava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration =
and too generic statement which doesn't answer MPLS issue for example.
=20

best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, =
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it =
knows nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its =
only one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
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lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Speex is not =
layered, isn&#8217;t
it?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>best regards,</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Jozsef Vass [mailto:jovass@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, October =
27, 2009
12:09 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Stephan Wenger; Slava =
Borilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>In rfc 3984 =
processing
packets by mane has some clear benefits, e.g., discarding packets that =
the
decoder would describe anyways (section 7.3), dropping packets in case =
of
congestion, etc.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Here, we mostly =
talking
about reducing packet size, and I am not sure how beneficial it is =
regarding
the overall system. Consider speex. Quality level six, wideband and =
narrowband
encoder produces 52 bytes and 28 bytes of payload, respectively. Adding
IP/UDP/RTP header, you go from 92 bytes to 68 bytes. Since packet rate =
is the
same, I do not think it will affect congestion too much. Please also =
consider
that processing packets by intermediary, adds additional =
delay.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Stephan Wenger [mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, =
2009
11:15 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava =
Borilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
lang=3DEN-US
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DCalibri><span
lang=3DEN-US style=3D'font-size:11.0pt;font-family:Calibri'>I think that =
we have a
terminology problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP =
packets,
and (potentially) modify them, are higher layer devices. &nbsp;After =
long
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify =
packets
at the application layer, for example to remove layers, re-package H.264 =
NAL
units to adapt to different MTU sizes, and do all kinds of other things =
that
violate the end-to-end principle. &nbsp;A MANE has business in looking =
into,
and potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from =
both an
abstract layering theory viewpoint and in practice) that packets are =
being
opened and the content is repackaged. &nbsp;It&#8217;s actually common
practice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a =
href=3D"jovass@adobe.com">jovass@adobe.com</a>&gt;
wrote:</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>I do not like =
the idea of
sending multiple layers in a packet either. I feel very uncomfortable of
routers modifying packets. It is OK not to deliver, but delivering =
semantically
modified packet is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much =
higher and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava
Borilin [<a =
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>]
<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D2 color=3Dnavy face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Again - my point is that router =
does NOT
have to know that something specifically is the codec, or specific =
codec- the
only goal is to let router aware that the stream (and every packet) is =
LAYERED,
and that it is legitimate to strip out some layers if necessary due to
congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice =
codec, video
codec, audio codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure =
using
layers in separate streams will require too much overhead (packets) =
which does
not make sense as well &#8211; and I am thinking about one stream of =
packets
comtaining all the layers, which, however, can be modified on the fly =
&#8220;in
the middle&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered
content&#8221; which does NOT loose the information (well, most of it) =
by
stripping out portions of the packet and reducing the traffic, so router =
can
strip out any type of layered content, without going deep into specific =
codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited =
engineer)
&#8211; and definitely I am looking for objections even more then for =
agreement
&#8211; for the sake of either finding the idea useful or making judged =
deny of
it, and I agree that this concept is only one possible way of working =
with
congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,<br>
Slava Borilin</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'> <a
href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D2 color=3Dnavy face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level =
consideration and
too generic statement which doesn&#8217;t answer MPLS issue for =
example.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,<br>
Slava Borilin</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Henry Sinnreich [<a =
href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</a>]
<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D5 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:18.0pt;
font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. =
Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
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/mailman/listinfo/codec</a><br>
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/mailman/listinfo/codec</a></span></font><font
size=3D2 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:Calibri'><o:p></o:p></span></font><=
/p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
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<p class=3DMsoNormal><font size=3D2 face=3DConsolas><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Consolas'>_________________________=
______________________<br>
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From jovass@adobe.com  Mon Oct 26 14:24:34 2009
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From: Jozsef Vass <jovass@adobe.com>
To: Slava Borilin <Borilin@spiritdsp.com>, Stephan Wenger <stewe@stewe.org>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
Date: Mon, 26 Oct 2009 14:23:31 -0700
Thread-Topic: [codec] updated charter proposal
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>From speex manual (http://www.speex.org/docs/manual/speex-manual/node4.html=
#SECTION00410000000000000000):
Integration of narrowband and wideband using an embedded bit-stream

Jozsef

________________________________
From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Monday, October 26, 2009 2:14 PM
To: Jozsef Vass; Stephan Wenger; Henry Sinnreich; Jean-Marc Valin; Lars Egg=
ert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

Speex is not layered, isn't it?

best regards,
Slava Borilin
________________________________
From: Jozsef Vass [mailto:jovass@adobe.com]
Sent: Tuesday, October 27, 2009 12:09 AM
To: Stephan Wenger; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars E=
ggert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

In rfc 3984 processing packets by mane has some clear benefits, e.g., disca=
rding packets that the decoder would describe anyways (section 7.3), droppi=
ng packets in case of congestion, etc.

Here, we mostly talking about reducing packet size, and I am not sure how b=
eneficial it is regarding the overall system. Consider speex. Quality level=
 six, wideband and narrowband encoder produces 52 bytes and 28 bytes of pay=
load, respectively. Adding IP/UDP/RTP header, you go from 92 bytes to 68 by=
tes. Since packet rate is the same, I do not think it will affect congestio=
n too much. Please also consider that processing packets by intermediary, a=
dds additional delay.

Jozsef

________________________________
From: Stephan Wenger [mailto:stewe@stewe.org]
Sent: Monday, October 26, 2009 11:15 AM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Egge=
rt
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey don't modify IP packets.  Boxes that look deep into the IP packets, and =
(potentially) modify them, are higher layer devices.  After long discussion=
s in AVT at the time, RFC 3984 and its successors call those boxes "Media A=
ware Network Elements", MANEs.  MANEs modify packets at the application lay=
er, for example to remove layers, re-package H.264 NAL units to adapt to di=
fferent MTU sizes, and do all kinds of other things that violate the end-to=
-end principle.  A MANE has business in looking into, and potentially modif=
ying, IP/UDP/RTP packets, a router has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an ab=
stract layering theory viewpoint and in practice) that packets are being op=
ened and the content is repackaged.  It's actually common practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:
I do not like the idea of sending multiple layers in a packet either. I fee=
l very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but delivering semantically modified packet is not expected.

Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.

Jozsef

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something specif=
ically is the codec, or specific codec- the only goal is to let router awar=
e that the stream (and every packet) is LAYERED, and that it is legitimate =
to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, audio =
codec, any type of content.

And I agree with Gregory - that having current infrastructure using layers =
in separate streams will require too much overhead (packets) which does not=
 make sense as well - and I am thinking about one stream of packets comtain=
ing all the layers, which, however, can be modified on the fly "in the midd=
le".
So we may think about more generic concept of "declared layered content" wh=
ich does NOT loose the information (well, most of it) by stripping out port=
ions of the packet and reducing the traffic, so router can strip out any ty=
pe of layered content, without going deep into specific codec.

Again- this is "na=EFve" suggestions from me (very limited engineer) - and =
definitely I am looking for objections even more then for agreement - for t=
he sake of either finding the idea useful or making judged deny of it, and =
I agree that this concept is only one possible way of working with congesti=
on.


best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of S=
lava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration an=
d too generic statement which doesn't answer MPLS issue for example.


best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>From speex manual (<a
href=3D"http://www.speex.org/docs/manual/speex-manual/node4.html#SECTION004=
10000000000000000">http://www.speex.org/docs/manual/speex-manual/node4.html=
#SECTION00410000000000000000</a>):
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Integration of narrowband<a name=3D104></a> and wideband<a name=3D1=
05></a>
using an embedded bit-stream</span></font><font size=3D2 color=3Dnavy face=
=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span><=
/font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Bo=
rilin
[mailto:Borilin@spiritdsp.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, 20=
09
2:14 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Stephan Wen=
ger;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Speex is not layered, isn&#8217;t it?<=
o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span lang=3D=
RU
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best regards,</span=
></font><font
color=3Dnavy><span lang=3DRU style=3D'color:navy'><o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span lang=3D=
RU
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava Borilin</span=
></font><span
lang=3DRU><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Jozsef V=
ass
[mailto:jovass@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, October 27, 2=
009
12:09 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Stephan Wenger; Slava Bo=
rilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span lang=3DR=
U
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>In rfc 3984 processing packets by mane=
 has
some clear benefits, e.g., discarding packets that the decoder would descri=
be
anyways (section 7.3), dropping packets in case of congestion, etc.<o:p></o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Here, we mostly talking about reducing
packet size, and I am not sure how beneficial it is regarding the overall
system. Consider speex. Quality level six, wideband and narrowband encoder
produces 52 bytes and 28 bytes of payload, respectively. Adding IP/UDP/RTP
header, you go from 92 bytes to 68 bytes. Since packet rate is the same, I =
do
not think it will affect congestion too much. Please also consider that
processing packets by intermediary, adds additional delay.<o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Stephan =
Wenger
[mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, 20=
09
11:15 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava Boril=
in; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 face=3DC=
alibri><span
style=3D'font-size:11.0pt;font-family:Calibri'>I think that we have a termi=
nology
problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP packe=
ts, and
(potentially) modify them, are higher layer devices. &nbsp;After long
discussions in AVT at the time, RFC 3984 and its successors call those boxe=
s
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify packe=
ts at the
application layer, for example to remove layers, re-package H.264 NAL units=
 to
adapt to different MTU sizes, and do all kinds of other things that violate=
 the
end-to-end principle. &nbsp;A MANE has business in looking into, and
potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from both=
 an
abstract layering theory viewpoint and in practice) that packets are being
opened and the content is repackaged. &nbsp;It&#8217;s actually common prac=
tice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.c=
om">jovass@adobe.com</a>&gt;
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do not like the idea of sending mult=
iple
layers in a packet either. I feel very uncomfortable of routers modifying
packets. It is OK not to deliver, but delivering semantically modified pack=
et
is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much higher =
and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Borilin [<a
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>] <br=
>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Again - my point is that router does NOT have to know tha=
t
something specifically is the codec, or specific codec- the only goal is to=
 let
router aware that the stream (and every packet) is LAYERED, and that it is =
legitimate
to strip out some layers if necessary due to congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice codec, v=
ideo codec, audio
codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure using l=
ayers in
separate streams will require too much overhead (packets) which does not ma=
ke
sense as well &#8211; and I am thinking about one stream of packets comtain=
ing all
the layers, which, however, can be modified on the fly &#8220;in the middle=
&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered conte=
nt&#8221; which
does NOT loose the information (well, most of it) by stripping out portions=
 of
the packet and reducing the traffic, so router can strip out any type of
layered content, without going deep into specific codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited engi=
neer) &#8211; and
definitely I am looking for objections even more then for agreement &#8211;=
 for the
sake of either finding the idea useful or making judged deny of it, and I a=
gree
that this concept is only one possible way of working with congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> <a href=3D"codec-bounces@iet=
f.org">codec-bounces@ietf.org</a>
[<a href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a=
>] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; Jean-Ma=
rc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion and
too generic statement which doesn&#8217;t answer MPLS issue for example.<br=
>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span>=
</font></b><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>=
 Henry
Sinnreich [<a href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com<=
/a>] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D5 face=3DCalibri><span style=3D'font-size:18.0pt;font-family:C=
alibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<br>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its only =
one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, why<br=
>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it can<br=
>
be too hard and too long to tell each sender to reduce the rate - why<br>
not to leave this freedom to router as this may be more efficient for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router do<br>
not need to understand anything except that it is LAYERED streams (even<br>
don't know whether its voice, video or audio or anything else of layered<br=
>
nature, and router &quot;has the right&quot; to strip out layers in certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] O=
n
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you don't<br>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
br>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<br=
>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt
to<br>
be<br>
careful not to include in the codec things that don't belong there (e.g.<br=
>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at some<=
br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! Routers<b=
r>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
br>
&gt; packets to react appropriately to the marks or losses. In other words,=
<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not in<br=
>
&gt; response to whatever congestion level those losses/marks indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and this<br=
>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You don't<br>
need layers to have a multi-rate codec. Layering just means that you can<br=
>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* losses/marks<b=
r>
to<br>
&gt; determine capacity. People have tried to create congestion control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable -<b=
r>
the<br>
&gt; rate computation is usually clocked by the RTT. You can obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying that<br>
you<br>
can have one module that determines in real-time the best rate to use at<br=
>
<br>
that moment and then it tells the codec what to use. If it needs help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the =
other
layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how to=
<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<b=
r>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a></span></font><font
size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt;font-family:Calibri=
'><o:p></o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D2
face=3DCalibri><span style=3D'font-size:11.0pt;font-family:Calibri'>

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</span></font></div>

<p class=3DMsoNormal><font size=3D2 face=3DConsolas><span style=3D'font-siz=
e:10.0pt;
font-family:Consolas'>_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
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From: "Slava Borilin" <Borilin@spiritdsp.com>
To: "Jozsef Vass" <jovass@adobe.com>, "Stephan Wenger" <stewe@stewe.org>, "Henry Sinnreich" <hsinnrei@adobe.com>, "Jean-Marc Valin" <jean-marc.valin@octasic.com>, "Lars Eggert" <lars.eggert@nokia.com>
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Subject: Re: [codec] updated charter proposal
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I also think that we are talking not only about packet size, but only =
packet rate as well?

=20

best regards,

Slava Borilin

________________________________

From: Jozsef Vass [mailto:jovass@adobe.com]=20
Sent: Tuesday, October 27, 2009 12:09 AM
To: Stephan Wenger; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; =
Lars Eggert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

=20

In rfc 3984 processing packets by mane has some clear benefits, e.g., =
discarding packets that the decoder would describe anyways (section =
7.3), dropping packets in case of congestion, etc.

=20

Here, we mostly talking about reducing packet size, and I am not sure =
how beneficial it is regarding the overall system. Consider speex. =
Quality level six, wideband and narrowband encoder produces 52 bytes and =
28 bytes of payload, respectively. Adding IP/UDP/RTP header, you go from =
92 bytes to 68 bytes. Since packet rate is the same, I do not think it =
will affect congestion too much. Please also consider that processing =
packets by intermediary, adds additional delay.

=20

Jozsef

=20

________________________________

From: Stephan Wenger [mailto:stewe@stewe.org]=20
Sent: Monday, October 26, 2009 11:15 AM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars =
Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

=20

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they don't modify IP packets.  Boxes that look deep into the IP packets, =
and (potentially) modify them, are higher layer devices.  After long =
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes "Media Aware Network Elements", MANEs.  MANEs modify packets at =
the application layer, for example to remove layers, re-package H.264 =
NAL units to adapt to different MTU sizes, and do all kinds of other =
things that violate the end-to-end principle.  A MANE has business in =
looking into, and potentially modifying, IP/UDP/RTP packets, a router =
has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an =
abstract layering theory viewpoint and in practice) that packets are =
being opened and the content is repackaged.  It's actually common =
practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:

I do not like the idea of sending multiple layers in a packet either. I =
feel very uncomfortable of routers modifying packets. It is OK not to =
deliver, but delivering semantically modified packet is not expected.
=20
Using layers for video totally makes sense where the rates are much =
higher and layers can be transported in separate packets.
=20
Jozsef
=20

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]=20
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something =
specifically is the codec, or specific codec- the only goal is to let =
router aware that the stream (and every packet) is LAYERED, and that it =
is legitimate to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, =
audio codec, any type of content.
=20
And I agree with Gregory - that having current infrastructure using =
layers in separate streams will require too much overhead (packets) =
which does not make sense as well - and I am thinking about one stream =
of packets comtaining all the layers, which, however, can be modified on =
the fly "in the middle".
So we may think about more generic concept of "declared layered content" =
which does NOT loose the information (well, most of it) by stripping out =
portions of the packet and reducing the traffic, so router can strip out =
any type of layered content, without going deep into specific codec.
=20
Again- this is "na=EFve" suggestions from me (very limited engineer) - =
and definitely I am looking for objections even more then for agreement =
- for the sake of either finding the idea useful or making judged deny =
of it, and I agree that this concept is only one possible way of working =
with congestion.
=20

best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Slava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration =
and too generic statement which doesn't answer MPLS issue for example.
=20

best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, =
why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it =
knows nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its =
only one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>I also think =
that we are talking
not only about packet size, but only packet rate as =
well?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>best regards,</span></font><font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><o:p></o:p></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Jozsef Vass [mailto:jovass@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, October =
27, 2009
12:09 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Stephan Wenger; Slava =
Borilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>In rfc 3984 =
processing
packets by mane has some clear benefits, e.g., discarding packets that =
the
decoder would describe anyways (section 7.3), dropping packets in case =
of
congestion, etc.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Here, we mostly =
talking
about reducing packet size, and I am not sure how beneficial it is =
regarding
the overall system. Consider speex. Quality level six, wideband and =
narrowband
encoder produces 52 bytes and 28 bytes of payload, respectively. Adding
IP/UDP/RTP header, you go from 92 bytes to 68 bytes. Since packet rate =
is the
same, I do not think it will affect congestion too much. Please also =
consider
that processing packets by intermediary, adds additional =
delay.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Stephan Wenger [mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, =
2009
11:15 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava =
Borilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
lang=3DEN-US
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DCalibri><span
lang=3DEN-US style=3D'font-size:11.0pt;font-family:Calibri'>I think that =
we have a
terminology problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP =
packets,
and (potentially) modify them, are higher layer devices. &nbsp;After =
long
discussions in AVT at the time, RFC 3984 and its successors call those =
boxes
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify =
packets
at the application layer, for example to remove layers, re-package H.264 =
NAL
units to adapt to different MTU sizes, and do all kinds of other things =
that
violate the end-to-end principle. &nbsp;A MANE has business in looking =
into,
and potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from =
both an
abstract layering theory viewpoint and in practice) that packets are =
being
opened and the content is repackaged. &nbsp;It&#8217;s actually common
practice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a =
href=3D"jovass@adobe.com">jovass@adobe.com</a>&gt;
wrote:</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>I do not like =
the idea of
sending multiple layers in a packet either. I feel very uncomfortable of
routers modifying packets. It is OK not to deliver, but delivering =
semantically
modified packet is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much =
higher and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava
Borilin [<a =
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>]
<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D2 color=3Dnavy face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Again - my point is that router =
does NOT
have to know that something specifically is the codec, or specific =
codec- the
only goal is to let router aware that the stream (and every packet) is =
LAYERED,
and that it is legitimate to strip out some layers if necessary due to
congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice =
codec, video
codec, audio codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure =
using
layers in separate streams will require too much overhead (packets) =
which does
not make sense as well &#8211; and I am thinking about one stream of =
packets
comtaining all the layers, which, however, can be modified on the fly =
&#8220;in
the middle&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered
content&#8221; which does NOT loose the information (well, most of it) =
by
stripping out portions of the packet and reducing the traffic, so router =
can
strip out any type of layered content, without going deep into specific =
codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited =
engineer)
&#8211; and definitely I am looking for objections even more then for =
agreement
&#8211; for the sake of either finding the idea useful or making judged =
deny of
it, and I agree that this concept is only one possible way of working =
with
congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,<br>
Slava Borilin</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'> <a
href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D2 color=3Dnavy face=3DArial><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level =
consideration
and too generic statement which doesn&#8217;t answer MPLS issue for =
example.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:11.0pt;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,<br>
Slava Borilin</span></font><span lang=3DEN-US><o:p></o:p></span></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
Henry Sinnreich [<a =
href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</a>]
<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October =
22, 2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; =
Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a =
href=3D"codec@ietf.org">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] =
updated
charter proposal<br>
</span></font><span lang=3DEN-US><br>
</span><font size=3D5 face=3DCalibri><span lang=3DEN-US =
style=3D'font-size:18.0pt;
font-family:Calibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. =
Why?<br>
&gt;If we are building MPLS and other systems that is content-dependent, =
why<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it =
knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its =
only one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. =
Why?<br>
If we are building MPLS and other systems that is content-dependent, =
why<br>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it =
can<br>
be too hard and too long to tell each sender to reduce the rate - =
why<br>
not to leave this freedom to router as this may be more efficient =
for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router =
do<br>
not need to understand anything except that it is LAYERED streams =
(even<br>
don't know whether its voice, video or audio or anything else of =
layered<br>
nature, and router &quot;has the right&quot; to strip out layers in =
certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>]=
 On
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you =
don't<br>
&gt; involve the codec in some form, performance/quality will be bad. =
Some<br>
&gt; codecs need time to adjust the rate. Some can only adjust it in =
pretty<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, =
without<br>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just =
want
to<br>
be<br>
careful not to include in the codec things that don't belong there =
(e.g.<br>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control =
will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates =
more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at =
some<br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want =
to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the =
codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! =
Routers<br>
&gt; drop or mark IP packets. It is up to the application sending those =
IP<br>
&gt; packets to react appropriately to the marks or losses. In other =
words,<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, =
and<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not =
in<br>
&gt; response to whatever congestion level those losses/marks =
indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and =
this<br>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the =
middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You =
don't<br>
need layers to have a multi-rate codec. Layering just means that you =
can<br>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much =
the<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality =
will<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be =
codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* =
losses/marks<br>
to<br>
&gt; determine capacity. People have tried to create congestion =
control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for =
example),<br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable =
-<br>
the<br>
&gt; rate computation is usually clocked by the RTT. You can =
obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate =
that I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying =
that<br>
you<br>
can have one module that determines in real-time the best rate to use =
at<br>
<br>
that moment and then it tells the codec what to use. If it needs =
help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to =
the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means =
that<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how =
to<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control =
constraints).<br>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK =
if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
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From stewe@stewe.org  Mon Oct 26 14:28:56 2009
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Date: Mon, 26 Oct 2009 14:28:44 -0700
From: Stephan Wenger <stewe@stewe.org>
To: Jozsef Vass <jovass@adobe.com>, Slava Borilin <Borilin@spiritdsp.com>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
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Thread-Topic: [codec] updated charter proposal
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I think that these arguments boil down to the discussion of measurement of
congestion in the packet count domain or in the bitrate domain.  I don=B9t
think there=B9s a verdict yet which of the two are really important.  It may
very well depend on your viewpoint and on what your company makes money of.

When operating in low delay environments (as it is the assumption here), I
agree that there is not much point in attempting to reduce the packet
count---doing so would increase delay and would presumably be unacceptable
to the application.  Therefore, congestion friendliness, when congestion is
measured in the packet count domain, seems to be independent from the codec
characteristics; as long as the codec is optimized for low delay.

However, if we start from the assumption that congestion is measured in the
bitrate domain, then the ballgame is somewhat different.  Yes, when we drop
a layer from a hypothetical layered audio codec, we would not safe a lot of
bits in absolute numbers, and yes, the percentage would be even more
disappointing considering the packetization overhead.  But still, we would
reduce the overall bitrate.  If you have thousands or millions of sessions
on the same fiber, dropping layers could be argued to make a difference.

Stephan=20


On 10/26/09 2:09 PM, "Jozsef Vass" <jovass@adobe.com> wrote:

> In rfc 3984 processing packets by mane has some clear benefits, e.g.,
> discarding packets that the decoder would describe anyways (section 7.3),
> dropping packets in case of congestion, etc.
> =20
> Here, we mostly talking about reducing packet size, and I am not sure how
> beneficial it is regarding the overall system. Consider speex. Quality le=
vel
> six, wideband and narrowband encoder produces 52 bytes and 28 bytes of
> payload, respectively. Adding IP/UDP/RTP header, you go from 92 bytes to =
68
> bytes. Since packet rate is the same, I do not think it will affect conge=
stion
> too much. Please also consider that processing packets by intermediary, a=
dds
> additional delay.
> =20
> Jozsef
> =20
>=20
>=20
> From: Stephan Wenger [mailto:stewe@stewe.org]
> Sent: Monday, October 26, 2009 11:15 AM
> To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eg=
gert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> =20
> I think that we have a terminology problem here.
>=20
> Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is =
they
> don=B9t modify IP packets.  Boxes that look deep into the IP packets, and
> (potentially) modify them, are higher layer devices.  After long discussi=
ons
> in AVT at the time, RFC 3984 and its successors call those boxes =B3Media A=
ware
> Network Elements=B2, MANEs.  MANEs modify packets at the application layer,=
 for
> example to remove layers, re-package H.264 NAL units to adapt to differen=
t MTU
> sizes, and do all kinds of other things that violate the end-to-end princ=
iple.
> A MANE has business in looking into, and potentially modifying, IP/UDP/RT=
P
> packets, a router has not.
>=20
> When MANEs are in the picture, it=B9s IMO totally acceptable (from both an
> abstract layering theory viewpoint and in practice) that packets are bein=
g
> opened and the content is repackaged.  It=B9s actually common practice.
>=20
> Stephan
>=20
> On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:
> I do not like the idea of sending multiple layers in a packet either. I f=
eel
> very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but
> delivering semantically modified packet is not expected.
> =20
> Using layers for video totally makes sense where the rates are much highe=
r and
> layers can be transported in separate packets.
> =20
> Jozsef
> =20
>=20
>=20
> From: Slava Borilin [mailto:Borilin@spiritdsp.com]
> Sent: Thursday, October 22, 2009 12:23 PM
> To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Again - my point is that router does NOT have to know that something
> specifically is the codec, or specific codec- the only goal is to let rou=
ter
> aware that the stream (and every packet) is LAYERED, and that it is legit=
imate
> to strip out some layers if necessary due to congestion.
> The content of such =B3layered=B2 can be any =AD voice codec, video codec, audi=
o
> codec, any type of content.
> =20
> And I agree with Gregory =AD that having current infrastructure using layer=
s in
> separate streams will require too much overhead (packets) which does not =
make
> sense as well =AD and I am thinking about one stream of packets comtaining =
all
> the layers, which, however, can be modified on the fly =B3in the middle=B2.
> So we may think about more generic concept of =B3declared layered content=B2 =
which
> does NOT loose the information (well, most of it) by stripping out portio=
ns of
> the packet and reducing the traffic, so router can strip out any type of
> layered content, without going deep into specific codec.
> =20
> Again- this is =B3na=EFve=B2 suggestions from me (very limited engineer) =AD and
> definitely I am looking for objections even more then for agreement =AD for=
 the
> sake of either finding the idea useful or making judged deny of it, and I
> agree that this concept is only one possible way of working with congesti=
on.
> =20
>=20
> best regards,
> Slava Borilin
>=20
>=20
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
> Slava Borilin
> Sent: Thursday, October 22, 2009 11:13 PM
> To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Henry,
> With all of my respect (you know) =AD this is too high-level consideration =
and
> too generic statement which doesn=B9t answer MPLS issue for example.
> =20
>=20
> best regards,
> Slava Borilin
>=20
>=20
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> Sent: Thursday, October 22, 2009 11:10 PM
> To: Slava Borilin; Jean-Marc Valin; Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Slava,
>=20
>> >I am not that sure that routers should not be aware of the codecs. Why?
>> >If we are building MPLS and other systems that is content-dependent, wh=
y
>> >not let router understand the codec?
>=20
> The essence of the Dumb Network (Internet) and its routers is that it kno=
ws
> nothing about the applications.
> All applications that nobody planned for can run over it.
> The tragedy of the telephone network was is knew everything about its onl=
y one
> application.
>=20
> Henry
>=20
>=20
> On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
> I am not that sure that routers should not be aware of the codecs. Why?
> If we are building MPLS and other systems that is content-dependent, why
> not let router understand the codec?
>=20
> Especially if we have hordes of voice channels going thru router, it can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?
>=20
> Same applies to H.264SVC for example - at the end of the day, router do
> not need to understand anything except that it is LAYERED streams (even
> don't know whether its voice, video or audio or anything else of layered
> nature, and router "has the right" to strip out layers in certain order
> to prevent congestion?
>=20
> best regards,
> Slava Borilin
>=20
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Thursday, October 22, 2009 5:45 PM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
>=20
> Lars Eggert wrote:
>> > Also, what we've learned from media over DCCP is that if you don't
>> > involve the codec in some form, performance/quality will be bad. Some
>> > codecs need time to adjust the rate. Some can only adjust it in pretty
>=20
>> > coarse steps. Etc. If you let DCCP adjust the rate blindly, without
>> > involving the codec in some form, you might get very good TCP
>> > friendliness, but maybe quality will suffer.
>=20
> I'm totally in favour of "involving the codec" in here. I just want to
> be
> careful not to include in the codec things that don't belong there (e.g.
> a
> DCCP implementation or a whole jitter buffer).
>=20
>> > (And there is always the ramping up part - congestion control will
> never
>> > attempt to probe for more capacity unless the codec generates more
> data,
>> > and why would the codec do that if it's converged to transmit at some
>> > lower rate.)
>=20
> Well, the codec transmits at the rate you ask it to use. If you want to
> probe for more capacity, then you tell the codec that it can use a
> higher
> rate. Now, if there's a special way in which you want to ask the codec,
> then maybe that can become an additional feature for the codec.
>=20
>> > By no means should routers have to understand codec formats! Routers
>> > drop or mark IP packets. It is up to the application sending those IP
>> > packets to react appropriately to the marks or losses. In other words,
> a
>> > layered codec will see losses/marks to packets sent on all layers, and
>=20
>> > will then need to figure out whether to strip off a layer or not in
>> > response to whatever congestion level those losses/marks indicate.
>=20
> I definitely agree that the routers shouldn't know about codecs and this
> is
> why I was saying that IMO layered codecs aren't too useful for
> congestion
> control (though they have other uses). If there's nothing in the middle
> of
> the network to strip off the layers, then you might as well not have
> layers
> at all. All you need is to ask the sender to reduce the rate. You don't
> need layers to have a multi-rate codec. Layering just means that you can
>=20
> strip off bits and still have the result sound OK.
>=20
>>> >> OK, I may be wrong here, but my understanding is that how much the
> rate
>>> >> should change depending on congestion is actually not that
>>> >> codec-specific.
>> >
>> > It *can* be non-codec-specific, see DCCP for example. But quality will
>=20
>> > suffer IMO.
>=20
> Well, we need to figure out the aspects that should be codec-specific
> and
> implement just those. Does anyone have a list of the issues codecs
> should
> be aware of?
>=20
>> > Most Internet congestion control is based on *creating* losses/marks
> to
>> > determine capacity. People have tried to create congestion control
>> > schemes that avoid causing losses (delay-based schemed, for example),
>> > but they are perceived to be more fragile.
>=20
> Yes, I can imagine the problem not being trivial at all.
>=20
>>> >> Once you figure this out, you
>>> >> can just tell the codec to use that rate.
>> >
>> > It's not that easy, because rates in the Internet are never stable -
> the
>> > rate computation is usually clocked by the RTT. You can obviously
> dampen
>> > this somehow, but you can never say "now I have found a rate that I
> can
>> > safely use forever."
>=20
> Oh, I never said we should set a rate forever. I was just saying that
> you
> can have one module that determines in real-time the best rate to use at
>=20
> that moment and then it tells the codec what to use. If it needs help
> from
> the codec, then we can define special functionality.
>=20
>>> >> On top of that, the codec may be
>>> >> able to do a bit better by providing a "service" to the other layers=
.
> On
>>> >> example here (also included in the draft) is VBR, which means that
> the
>>> >> application specifies an average rate and the codec decides how to
>>> >> vary the
>>> >> rate to achieve that (subject to some rate control constraints).
> Would
>>> >> other such mechanisms be useful to deal with congestion?
>> >
>> > I'm not sure this would work, because congestion control will
> basically
>> > change the VBR input parameter continuously.
>=20
> Changing the VBR parameters continuously is not a problem. VBR just
> means
> that you'd like to have an average of N kb/s for now, but you're OK if
> there are variations from frame to frame.
>=20
>         Jean-Marc
> _______________________________________________
> 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
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>=20


--B_3339412130_2650127
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	charset="ISO-8859-1"
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<HTML>
<HEAD>
<TITLE>Re: [codec] updated charter proposal</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:11pt=
'>I think that these arguments boil down to the discussion of measurement of=
 congestion in the packet count domain or in the bitrate domain. &nbsp;I don=
&#8217;t think there&#8217;s a verdict yet which of the two are really impor=
tant. &nbsp;It may very well depend on your viewpoint and on what your compa=
ny makes money of.<BR>
<BR>
When operating in low delay environments (as it is the assumption here), I =
agree that there is not much point in attempting to reduce the packet count-=
--doing so would increase delay and would presumably be unacceptable to the =
application. &nbsp;Therefore, congestion friendliness, when congestion is me=
asured in the packet count domain, seems to be independent from the codec ch=
aracteristics; as long as the codec is optimized for low delay.<BR>
<BR>
However, if we start from the assumption that congestion is measured in the=
 bitrate domain, then the ballgame is somewhat different. &nbsp;Yes, when we=
 drop a layer from a hypothetical layered audio codec, we would not safe a l=
ot of bits in absolute numbers, and yes, the percentage would be even more d=
isappointing considering the packetization overhead. &nbsp;But still, we wou=
ld reduce the overall bitrate. &nbsp;If you have thousands or millions of se=
ssions on the same fiber, dropping layers could be argued to make a differen=
ce.<BR>
<BR>
Stephan <BR>
<BR>
<BR>
On 10/26/09 2:09 PM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.com"=
>jovass@adobe.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"=
Arial"><SPAN STYLE=3D'font-size:10pt'>In rfc 3984 processing packets by mane h=
as some clear benefits, e.g., discarding packets that the decoder would desc=
ribe anyways (section 7.3), dropping packets in case of congestion, etc.<BR>
&nbsp;<BR>
Here, we mostly talking about reducing packet size, and I am not sure how b=
eneficial it is regarding the overall system. Consider speex. Quality level =
six, wideband and narrowband encoder produces 52 bytes and 28 bytes of paylo=
ad, respectively. Adding IP/UDP/RTP header, you go from 92 bytes to 68 bytes=
. Since packet rate is the same, I do not think it will affect congestion to=
o much. Please also consider that processing packets by intermediary, adds a=
dditional delay.<BR>
&nbsp;<BR>
Jozsef<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> Stephan Wenger [<a href=3D"mailto:stewe@stewe.org=
">mailto:stewe@stewe.org</a>] <BR>
<B>Sent:</B> Monday, October 26, 2009 11:15 AM<BR>
<B>To:</B> Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; La=
rs Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'> <BR>
</SPAN></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:11pt'>I think that we have a terminology problem here.<BR>
<BR>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP pac=
kets, and (potentially) modify them, are higher layer devices. &nbsp;After l=
ong discussions in AVT at the time, RFC 3984 and its successors call those b=
oxes &#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify p=
ackets at the application layer, for example to remove layers, re-package H.=
264 NAL units to adapt to different MTU sizes, and do all kinds of other thi=
ngs that violate the end-to-end principle. &nbsp;A MANE has business in look=
ing into, and potentially modifying, IP/UDP/RTP packets, a router has not.<B=
R>
<BR>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from both=
 an abstract layering theory viewpoint and in practice) that packets are bei=
ng opened and the content is repackaged. &nbsp;It&#8217;s actually common pr=
actice.<BR>
<BR>
Stephan<BR>
<BR>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.com=
">jovass@adobe.com</a>&gt; wrote:<BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>I do not like the idea of sending multiple layers in=
 a packet either. I feel very uncomfortable of routers modifying packets. It=
 is OK not to deliver, but delivering semantically modified packet is not ex=
pected.<BR>
&nbsp;<BR>
Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.<BR>
&nbsp;<BR>
Jozsef<BR>
&nbsp;
</SPAN></FONT></FONT></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:11pt=
'><BR>
</SPAN></FONT><FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:10pt'><B>From:</B> Slava Borilin [<a href=3D"mailto:Bo=
rilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>] <BR>
<B>Sent:</B> Thursday, October 22, 2009 12:23 PM<BR>
<B>To:</B> Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>Again - my point is that router does NOT have to kno=
w that something specifically is the codec, or specific codec- the only goal=
 is to let router aware that the stream (and every packet) is LAYERED, and t=
hat it is legitimate to strip out some layers if necessary due to congestion=
.<BR>
The content of such &#8220;layered&#8221; can be any &#8211; voice codec, v=
ideo codec, audio codec, any type of content.<BR>
&nbsp;<BR>
And I agree with Gregory &#8211; that having current infrastructure using l=
ayers in separate streams will require too much overhead (packets) which doe=
s not make sense as well &#8211; and I am thinking about one stream of packe=
ts comtaining all the layers, which, however, can be modified on the fly &#8=
220;in the middle&#8221;.<BR>
So we may think about more generic concept of &#8220;declared layered conte=
nt&#8221; which does NOT loose the information (well, most of it) by strippi=
ng out portions of the packet and reducing the traffic, so router can strip =
out any type of layered content, without going deep into specific codec.<BR>
&nbsp;<BR>
Again- this is &#8220;na&iuml;ve&#8221; suggestions from me (very limited e=
ngineer) &#8211; and definitely I am looking for objections even more then f=
or agreement &#8211; for the sake of either finding the idea useful or makin=
g judged deny of it, and I agree that this concept is only one possible way =
of working with congestion.<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>best regards,<BR>
Slava Borilin<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> <a href=3D"codec-bounces@ietf.org">codec-bounces@=
ietf.org</a> [<a href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@i=
etf.org</a>] <B>On Behalf Of </B>Slava Borilin<BR>
<B>Sent:</B> Thursday, October 22, 2009 11:13 PM<BR>
<B>To:</B> Henry Sinnreich; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>Henry,<BR>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion and too generic statement which doesn&#8217;t answer MPLS issue for exam=
ple.<BR>
&nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#000080"><FONT SIZE=3D"2"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:10pt'>best regards,<BR>
Slava Borilin<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>
</SPAN></FONT>
<P ALIGN=3DCENTER>
<FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12pt'><HR ALIGN=3DCENTER =
SIZE=3D"2" WIDTH=3D"100%"></SPAN></FONT>
<P>
<FONT SIZE=3D"2"><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE=3D'=
font-size:10pt'><B>From:</B> Henry Sinnreich [<a href=3D"mailto:hsinnrei@adobe=
.com">mailto:hsinnrei@adobe.com</a>] <BR>
<B>Sent:</B> Thursday, October 22, 2009 11:10 PM<BR>
<B>To:</B> Slava Borilin; Jean-Marc Valin; Lars Eggert<BR>
<B>Cc:</B> <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<B>Subject:</B> Re: [codec] updated charter proposal<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Times New Roman"><SPAN STYLE=3D'font-size:12=
pt'><BR>
</SPAN></FONT><FONT SIZE=3D"5"><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial=
"><SPAN STYLE=3D'font-size:18pt'>Slava,<BR>
<BR>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<BR>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<BR>
&gt;not let router understand the codec?<BR>
<BR>
The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.<BR>
All applications that nobody planned for can run over it.<BR>
The tragedy of the telephone network was is knew everything about its only =
one application.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a href=3D"Borilin@spiritd=
sp.com">Borilin@spiritdsp.com</a>&gt; wrote:<BR>
I am not that sure that routers should not be aware of the codecs. Why?<BR>
If we are building MPLS and other systems that is content-dependent, why<BR=
>
not let router understand the codec?<BR>
<BR>
Especially if we have hordes of voice channels going thru router, it can<BR=
>
be too hard and too long to tell each sender to reduce the rate - why<BR>
not to leave this freedom to router as this may be more efficient for<BR>
congestion recovery?<BR>
<BR>
Same applies to H.264SVC for example - at the end of the day, router do<BR>
not need to understand anything except that it is LAYERED streams (even<BR>
don't know whether its voice, video or audio or anything else of layered<BR=
>
nature, and router &quot;has the right&quot; to strip out layers in certain=
 order<BR>
to prevent congestion?<BR>
<BR>
best regards,<BR>
Slava Borilin<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a href=3D=
"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] On Behalf=
<BR>
Of Jean-Marc Valin<BR>
Sent: Thursday, October 22, 2009 5:45 PM<BR>
To: Lars Eggert<BR>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
Subject: Re: [codec] updated charter proposal<BR>
<BR>
Lars Eggert wrote:<BR>
&gt; Also, what we've learned from media over DCCP is that if you don't<BR>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
BR>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<BR>
<BR>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<BR=
>
&gt; involving the codec in some form, you might get very good TCP<BR>
&gt; friendliness, but maybe quality will suffer.<BR>
<BR>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt to<BR>
be<BR>
careful not to include in the codec things that don't belong there (e.g.<BR=
>
a<BR>
DCCP implementation or a whole jitter buffer).<BR>
<BR>
&gt; (And there is always the ramping up part - congestion control will<BR>
never<BR>
&gt; attempt to probe for more capacity unless the codec generates more<BR>
data,<BR>
&gt; and why would the codec do that if it's converged to transmit at some<=
BR>
&gt; lower rate.)<BR>
<BR>
Well, the codec transmits at the rate you ask it to use. If you want to<BR>
probe for more capacity, then you tell the codec that it can use a<BR>
higher<BR>
rate. Now, if there's a special way in which you want to ask the codec,<BR>
then maybe that can become an additional feature for the codec.<BR>
<BR>
&gt; By no means should routers have to understand codec formats! Routers<B=
R>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
BR>
&gt; packets to react appropriately to the marks or losses. In other words,=
<BR>
a<BR>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<BR>
<BR>
&gt; will then need to figure out whether to strip off a layer or not in<BR=
>
&gt; response to whatever congestion level those losses/marks indicate.<BR>
<BR>
I definitely agree that the routers shouldn't know about codecs and this<BR=
>
is<BR>
why I was saying that IMO layered codecs aren't too useful for<BR>
congestion<BR>
control (though they have other uses). If there's nothing in the middle<BR>
of<BR>
the network to strip off the layers, then you might as well not have<BR>
layers<BR>
at all. All you need is to ask the sender to reduce the rate. You don't<BR>
need layers to have a multi-rate codec. Layering just means that you can<BR=
>
<BR>
strip off bits and still have the result sound OK.<BR>
<BR>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<BR>
rate<BR>
&gt;&gt; should change depending on congestion is actually not that<BR>
&gt;&gt; codec-specific.<BR>
&gt;<BR>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<BR>
<BR>
&gt; suffer IMO.<BR>
<BR>
Well, we need to figure out the aspects that should be codec-specific<BR>
and<BR>
implement just those. Does anyone have a list of the issues codecs<BR>
should<BR>
be aware of?<BR>
<BR>
&gt; Most Internet congestion control is based on *creating* losses/marks<B=
R>
to<BR>
&gt; determine capacity. People have tried to create congestion control<BR>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
BR>
&gt; but they are perceived to be more fragile.<BR>
<BR>
Yes, I can imagine the problem not being trivial at all.<BR>
<BR>
&gt;&gt; Once you figure this out, you<BR>
&gt;&gt; can just tell the codec to use that rate.<BR>
&gt;<BR>
&gt; It's not that easy, because rates in the Internet are never stable -<B=
R>
the<BR>
&gt; rate computation is usually clocked by the RTT. You can obviously<BR>
dampen<BR>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<BR>
can<BR>
&gt; safely use forever.&quot;<BR>
<BR>
Oh, I never said we should set a rate forever. I was just saying that<BR>
you<BR>
can have one module that determines in real-time the best rate to use at<BR=
>
<BR>
that moment and then it tells the codec what to use. If it needs help<BR>
from<BR>
the codec, then we can define special functionality.<BR>
<BR>
&gt;&gt; On top of that, the codec may be<BR>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the =
other layers.<BR>
On<BR>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<BR>
the<BR>
&gt;&gt; application specifies an average rate and the codec decides how to=
<BR>
&gt;&gt; vary the<BR>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<B=
R>
Would<BR>
&gt;&gt; other such mechanisms be useful to deal with congestion?<BR>
&gt;<BR>
&gt; I'm not sure this would work, because congestion control will<BR>
basically<BR>
&gt; change the VBR input parameter continuously.<BR>
<BR>
Changing the VBR parameters continuously is not a problem. VBR just<BR>
means<BR>
that you'd like to have an average of N kb/s for now, but you're OK if<BR>
there are variations from frame to frame.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<BR>
_______________________________________________<BR>
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From jean-marc.valin@usherbrooke.ca  Mon Oct 26 14:30:16 2009
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Date: Mon, 26 Oct 2009 17:30:17 -0400
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Subject: Re: [codec] updated charter proposal
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Speex has an embedded bit-stream with a narrowband base and a wideband
"layer" on top. However, it is not possible to change the rate of the
narrowband or wideband "layers" by removing bits. It is only possible to
remove the entire wideband layer. This has the advantage of not
introducing a large cost in terms of bit-rate, while still making it
easy to decode a wideband stream in narrowband mode. That being said,
I'm aware that SILK (and probably IP-MR and BV32, but I haven't heard
samples) out-perform Speex in terms of quality vs rate.

	Jean-Marc

Slava Borilin a écrit :
> Speex is not layered, isn’t it?
> 
>  
> 
> best regards,
> 
> Slava Borilin
> 
> ------------------------------------------------------------------------
> 
> *From:* Jozsef Vass [mailto:jovass@adobe.com]
> *Sent:* Tuesday, October 27, 2009 12:09 AM
> *To:* Stephan Wenger; Slava Borilin; Henry Sinnreich; Jean-Marc Valin;
> Lars Eggert
> *Cc:* codec@ietf.org
> *Subject:* RE: [codec] updated charter proposal
> 
>  
> 
> In rfc 3984 processing packets by mane has some clear benefits, e.g.,
> discarding packets that the decoder would describe anyways (section
> 7.3), dropping packets in case of congestion, etc.
> 
>  
> 
> Here, we mostly talking about reducing packet size, and I am not sure
> how beneficial it is regarding the overall system. Consider speex.
> Quality level six, wideband and narrowband encoder produces 52 bytes and
> 28 bytes of payload, respectively. Adding IP/UDP/RTP header, you go from
> 92 bytes to 68 bytes. Since packet rate is the same, I do not think it
> will affect congestion too much. Please also consider that processing
> packets by intermediary, adds additional delay.
> 
>  
> 
> Jozsef
> 
>  
> 
> ------------------------------------------------------------------------
> 
> *From:* Stephan Wenger [mailto:stewe@stewe.org]
> *Sent:* Monday, October 26, 2009 11:15 AM
> *To:* Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars
> Eggert
> *Cc:* codec@ietf.org
> *Subject:* Re: [codec] updated charter proposal
> 
>  
> 
> I think that we have a terminology problem here.
> 
> Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is
> they don’t modify IP packets.  Boxes that look deep into the IP packets,
> and (potentially) modify them, are higher layer devices.  After long
> discussions in AVT at the time, RFC 3984 and its successors call those
> boxes “Media Aware Network Elements”, MANEs.  MANEs modify packets at
> the application layer, for example to remove layers, re-package H.264
> NAL units to adapt to different MTU sizes, and do all kinds of other
> things that violate the end-to-end principle.  A MANE has business in
> looking into, and potentially modifying, IP/UDP/RTP packets, a router
> has not.
> 
> When MANEs are in the picture, it’s IMO totally acceptable (from both an
> abstract layering theory viewpoint and in practice) that packets are
> being opened and the content is repackaged.  It’s actually common practice.
> 
> Stephan
> 
> On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:
> 
> I do not like the idea of sending multiple layers in a packet either. I
> feel very uncomfortable of routers modifying packets. It is OK not to
> deliver, but delivering semantically modified packet is not expected.
>  
> Using layers for video totally makes sense where the rates are much
> higher and layers can be transported in separate packets.
>  
> Jozsef
>  
> 
> ------------------------------------------------------------------------
> 
> *From:* Slava Borilin [mailto:Borilin@spiritdsp.com]
> *Sent:* Thursday, October 22, 2009 12:23 PM
> *To:* Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> *Cc:* codec@ietf.org
> *Subject:* Re: [codec] updated charter proposal
> 
> Again - my point is that router does NOT have to know that something
> specifically is the codec, or specific codec- the only goal is to let
> router aware that the stream (and every packet) is LAYERED, and that it
> is legitimate to strip out some layers if necessary due to congestion.
> The content of such “layered” can be any – voice codec, video codec,
> audio codec, any type of content.
>  
> And I agree with Gregory – that having current infrastructure using
> layers in separate streams will require too much overhead (packets)
> which does not make sense as well – and I am thinking about one stream
> of packets comtaining all the layers, which, however, can be modified on
> the fly “in the middle”.
> So we may think about more generic concept of “declared layered content”
> which does NOT loose the information (well, most of it) by stripping out
> portions of the packet and reducing the traffic, so router can strip out
> any type of layered content, without going deep into specific codec.
>  
> Again- this is “naïve” suggestions from me (very limited engineer) – and
> definitely I am looking for objections even more then for agreement –
> for the sake of either finding the idea useful or making judged deny of
> it, and I agree that this concept is only one possible way of working
> with congestion.
>  
> 
> best regards,
> Slava Borilin
> 
> ------------------------------------------------------------------------
> 
> *From:* codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] *On
> Behalf Of *Slava Borilin
> *Sent:* Thursday, October 22, 2009 11:13 PM
> *To:* Henry Sinnreich; Jean-Marc Valin; Lars Eggert
> *Cc:* codec@ietf.org
> *Subject:* Re: [codec] updated charter proposal
> 
> Henry,
> With all of my respect (you know) – this is too high-level consideration
> and too generic statement which doesn’t answer MPLS issue for example.
>  
> 
> best regards,
> Slava Borilin
> 
> ------------------------------------------------------------------------
> 
> *From:* Henry Sinnreich [mailto:hsinnrei@adobe.com]
> *Sent:* Thursday, October 22, 2009 11:10 PM
> *To:* Slava Borilin; Jean-Marc Valin; Lars Eggert
> *Cc:* codec@ietf.org
> *Subject:* Re: [codec] updated charter proposal
> 
> Slava,
> 
>>I am not that sure that routers should not be aware of the codecs. Why?
>>If we are building MPLS and other systems that is content-dependent, why
>>not let router understand the codec?
> 
> The essence of the Dumb Network (Internet) and its routers is that it
> knows nothing about the applications.
> All applications that nobody planned for can run over it.
> The tragedy of the telephone network was is knew everything about its
> only one application.
> 
> Henry
> 
> 
> On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
> I am not that sure that routers should not be aware of the codecs. Why?
> If we are building MPLS and other systems that is content-dependent, why
> not let router understand the codec?
> 
> Especially if we have hordes of voice channels going thru router, it can
> be too hard and too long to tell each sender to reduce the rate - why
> not to leave this freedom to router as this may be more efficient for
> congestion recovery?
> 
> Same applies to H.264SVC for example - at the end of the day, router do
> not need to understand anything except that it is LAYERED streams (even
> don't know whether its voice, video or audio or anything else of layered
> nature, and router "has the right" to strip out layers in certain order
> to prevent congestion?
> 
> best regards,
> Slava Borilin
> 
> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Thursday, October 22, 2009 5:45 PM
> To: Lars Eggert
> Cc: codec@ietf.org
> Subject: Re: [codec] updated charter proposal
> 
> Lars Eggert wrote:
>> Also, what we've learned from media over DCCP is that if you don't
>> involve the codec in some form, performance/quality will be bad. Some
>> codecs need time to adjust the rate. Some can only adjust it in pretty
> 
>> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
>> involving the codec in some form, you might get very good TCP
>> friendliness, but maybe quality will suffer.
> 
> I'm totally in favour of "involving the codec" in here. I just want to
> be
> careful not to include in the codec things that don't belong there (e.g.
> a
> DCCP implementation or a whole jitter buffer).
> 
>> (And there is always the ramping up part - congestion control will
> never
>> attempt to probe for more capacity unless the codec generates more
> data,
>> and why would the codec do that if it's converged to transmit at some
>> lower rate.)
> 
> Well, the codec transmits at the rate you ask it to use. If you want to
> probe for more capacity, then you tell the codec that it can use a
> higher
> rate. Now, if there's a special way in which you want to ask the codec,
> then maybe that can become an additional feature for the codec.
> 
>> By no means should routers have to understand codec formats! Routers
>> drop or mark IP packets. It is up to the application sending those IP
>> packets to react appropriately to the marks or losses. In other words,
> a
>> layered codec will see losses/marks to packets sent on all layers, and
> 
>> will then need to figure out whether to strip off a layer or not in
>> response to whatever congestion level those losses/marks indicate.
> 
> I definitely agree that the routers shouldn't know about codecs and this
> is
> why I was saying that IMO layered codecs aren't too useful for
> congestion
> control (though they have other uses). If there's nothing in the middle
> of
> the network to strip off the layers, then you might as well not have
> layers
> at all. All you need is to ask the sender to reduce the rate. You don't
> need layers to have a multi-rate codec. Layering just means that you can
> 
> strip off bits and still have the result sound OK.
> 
>>> OK, I may be wrong here, but my understanding is that how much the
> rate
>>> should change depending on congestion is actually not that
>>> codec-specific.
>>
>> It *can* be non-codec-specific, see DCCP for example. But quality will
> 
>> suffer IMO.
> 
> Well, we need to figure out the aspects that should be codec-specific
> and
> implement just those. Does anyone have a list of the issues codecs
> should
> be aware of?
> 
>> Most Internet congestion control is based on *creating* losses/marks
> to
>> determine capacity. People have tried to create congestion control
>> schemes that avoid causing losses (delay-based schemed, for example),
>> but they are perceived to be more fragile.
> 
> Yes, I can imagine the problem not being trivial at all.
> 
>>> Once you figure this out, you
>>> can just tell the codec to use that rate.
>>
>> It's not that easy, because rates in the Internet are never stable -
> the
>> rate computation is usually clocked by the RTT. You can obviously
> dampen
>> this somehow, but you can never say "now I have found a rate that I
> can
>> safely use forever."
> 
> Oh, I never said we should set a rate forever. I was just saying that
> you
> can have one module that determines in real-time the best rate to use at
> 
> that moment and then it tells the codec what to use. If it needs help
> from
> the codec, then we can define special functionality.
> 
>>> On top of that, the codec may be
>>> able to do a bit better by providing a "service" to the other layers.
> On
>>> example here (also included in the draft) is VBR, which means that
> the
>>> application specifies an average rate and the codec decides how to
>>> vary the
>>> rate to achieve that (subject to some rate control constraints).
> Would
>>> other such mechanisms be useful to deal with congestion?
>>
>> I'm not sure this would work, because congestion control will
> basically
>> change the VBR input parameter continuously.
> 
> Changing the VBR parameters continuously is not a problem. VBR just
> means
> that you'd like to have an average of N kb/s for now, but you're OK if
> there are variations from frame to frame.
> 
>         Jean-Marc
> _______________________________________________
> 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
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> codec mailing list
> codec@ietf.org
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From jovass@adobe.com  Mon Oct 26 14:32:57 2009
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From: Jozsef Vass <jovass@adobe.com>
To: Slava Borilin <Borilin@spiritdsp.com>, Stephan Wenger <stewe@stewe.org>, Henry Sinnreich <hsinnrei@adobe.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>, Lars Eggert <lars.eggert@nokia.com>
Date: Mon, 26 Oct 2009 14:28:02 -0700
Thread-Topic: [codec] updated charter proposal
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] updated charter proposal
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Sorry, my link was stripped. As per speex documentation and my experience, =
one of the features of speex is "Integration of narrowband and wideband usi=
ng an embedded bit-stream"

Jozsef

________________________________
From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Monday, October 26, 2009 2:14 PM
To: Jozsef Vass; Stephan Wenger; Henry Sinnreich; Jean-Marc Valin; Lars Egg=
ert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

Speex is not layered, isn't it?

best regards,
Slava Borilin
________________________________
From: Jozsef Vass [mailto:jovass@adobe.com]
Sent: Tuesday, October 27, 2009 12:09 AM
To: Stephan Wenger; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars E=
ggert
Cc: codec@ietf.org
Subject: RE: [codec] updated charter proposal

In rfc 3984 processing packets by mane has some clear benefits, e.g., disca=
rding packets that the decoder would describe anyways (section 7.3), droppi=
ng packets in case of congestion, etc.

Here, we mostly talking about reducing packet size, and I am not sure how b=
eneficial it is regarding the overall system. Consider speex. Quality level=
 six, wideband and narrowband encoder produces 52 bytes and 28 bytes of pay=
load, respectively. Adding IP/UDP/RTP header, you go from 92 bytes to 68 by=
tes. Since packet rate is the same, I do not think it will affect congestio=
n too much. Please also consider that processing packets by intermediary, a=
dds additional delay.

Jozsef

________________________________
From: Stephan Wenger [mailto:stewe@stewe.org]
Sent: Monday, October 26, 2009 11:15 AM
To: Jozsef Vass; Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Egge=
rt
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

I think that we have a terminology problem here.

Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey don't modify IP packets.  Boxes that look deep into the IP packets, and =
(potentially) modify them, are higher layer devices.  After long discussion=
s in AVT at the time, RFC 3984 and its successors call those boxes "Media A=
ware Network Elements", MANEs.  MANEs modify packets at the application lay=
er, for example to remove layers, re-package H.264 NAL units to adapt to di=
fferent MTU sizes, and do all kinds of other things that violate the end-to=
-end principle.  A MANE has business in looking into, and potentially modif=
ying, IP/UDP/RTP packets, a router has not.

When MANEs are in the picture, it's IMO totally acceptable (from both an ab=
stract layering theory viewpoint and in practice) that packets are being op=
ened and the content is repackaged.  It's actually common practice.

Stephan

On 10/26/09 10:55 AM, "Jozsef Vass" <jovass@adobe.com> wrote:
I do not like the idea of sending multiple layers in a packet either. I fee=
l very uncomfortable of routers modifying packets. It is OK not to deliver,=
 but delivering semantically modified packet is not expected.

Using layers for video totally makes sense where the rates are much higher =
and layers can be transported in separate packets.

Jozsef

________________________________

From: Slava Borilin [mailto:Borilin@spiritdsp.com]
Sent: Thursday, October 22, 2009 12:23 PM
To: Slava Borilin; Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Again - my point is that router does NOT have to know that something specif=
ically is the codec, or specific codec- the only goal is to let router awar=
e that the stream (and every packet) is LAYERED, and that it is legitimate =
to strip out some layers if necessary due to congestion.
The content of such "layered" can be any - voice codec, video codec, audio =
codec, any type of content.

And I agree with Gregory - that having current infrastructure using layers =
in separate streams will require too much overhead (packets) which does not=
 make sense as well - and I am thinking about one stream of packets comtain=
ing all the layers, which, however, can be modified on the fly "in the midd=
le".
So we may think about more generic concept of "declared layered content" wh=
ich does NOT loose the information (well, most of it) by stripping out port=
ions of the packet and reducing the traffic, so router can strip out any ty=
pe of layered content, without going deep into specific codec.

Again- this is "na=EFve" suggestions from me (very limited engineer) - and =
definitely I am looking for objections even more then for agreement - for t=
he sake of either finding the idea useful or making judged deny of it, and =
I agree that this concept is only one possible way of working with congesti=
on.


best regards,
Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of S=
lava Borilin
Sent: Thursday, October 22, 2009 11:13 PM
To: Henry Sinnreich; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Henry,
With all of my respect (you know) - this is too high-level consideration an=
d too generic statement which doesn't answer MPLS issue for example.


best regards,
Slava Borilin

________________________________

From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Thursday, October 22, 2009 11:10 PM
To: Slava Borilin; Jean-Marc Valin; Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Slava,

>I am not that sure that routers should not be aware of the codecs. Why?
>If we are building MPLS and other systems that is content-dependent, why
>not let router understand the codec?

The essence of the Dumb Network (Internet) and its routers is that it knows=
 nothing about the applications.
All applications that nobody planned for can run over it.
The tragedy of the telephone network was is knew everything about its only =
one application.

Henry


On 10/22/09 1:23 PM, "Slava Borilin" <Borilin@spiritdsp.com> wrote:
I am not that sure that routers should not be aware of the codecs. Why?
If we are building MPLS and other systems that is content-dependent, why
not let router understand the codec?

Especially if we have hordes of voice channels going thru router, it can
be too hard and too long to tell each sender to reduce the rate - why
not to leave this freedom to router as this may be more efficient for
congestion recovery?

Same applies to H.264SVC for example - at the end of the day, router do
not need to understand anything except that it is LAYERED streams (even
don't know whether its voice, video or audio or anything else of layered
nature, and router "has the right" to strip out layers in certain order
to prevent congestion?

best regards,
Slava Borilin

-----Original Message-----
From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
Of Jean-Marc Valin
Sent: Thursday, October 22, 2009 5:45 PM
To: Lars Eggert
Cc: codec@ietf.org
Subject: Re: [codec] updated charter proposal

Lars Eggert wrote:
> Also, what we've learned from media over DCCP is that if you don't
> involve the codec in some form, performance/quality will be bad. Some
> codecs need time to adjust the rate. Some can only adjust it in pretty

> coarse steps. Etc. If you let DCCP adjust the rate blindly, without
> involving the codec in some form, you might get very good TCP
> friendliness, but maybe quality will suffer.

I'm totally in favour of "involving the codec" in here. I just want to
be
careful not to include in the codec things that don't belong there (e.g.
a
DCCP implementation or a whole jitter buffer).

> (And there is always the ramping up part - congestion control will
never
> attempt to probe for more capacity unless the codec generates more
data,
> and why would the codec do that if it's converged to transmit at some
> lower rate.)

Well, the codec transmits at the rate you ask it to use. If you want to
probe for more capacity, then you tell the codec that it can use a
higher
rate. Now, if there's a special way in which you want to ask the codec,
then maybe that can become an additional feature for the codec.

> By no means should routers have to understand codec formats! Routers
> drop or mark IP packets. It is up to the application sending those IP
> packets to react appropriately to the marks or losses. In other words,
a
> layered codec will see losses/marks to packets sent on all layers, and

> will then need to figure out whether to strip off a layer or not in
> response to whatever congestion level those losses/marks indicate.

I definitely agree that the routers shouldn't know about codecs and this
is
why I was saying that IMO layered codecs aren't too useful for
congestion
control (though they have other uses). If there's nothing in the middle
of
the network to strip off the layers, then you might as well not have
layers
at all. All you need is to ask the sender to reduce the rate. You don't
need layers to have a multi-rate codec. Layering just means that you can

strip off bits and still have the result sound OK.

>> OK, I may be wrong here, but my understanding is that how much the
rate
>> should change depending on congestion is actually not that
>> codec-specific.
>
> It *can* be non-codec-specific, see DCCP for example. But quality will

> suffer IMO.

Well, we need to figure out the aspects that should be codec-specific
and
implement just those. Does anyone have a list of the issues codecs
should
be aware of?

> Most Internet congestion control is based on *creating* losses/marks
to
> determine capacity. People have tried to create congestion control
> schemes that avoid causing losses (delay-based schemed, for example),
> but they are perceived to be more fragile.

Yes, I can imagine the problem not being trivial at all.

>> Once you figure this out, you
>> can just tell the codec to use that rate.
>
> It's not that easy, because rates in the Internet are never stable -
the
> rate computation is usually clocked by the RTT. You can obviously
dampen
> this somehow, but you can never say "now I have found a rate that I
can
> safely use forever."

Oh, I never said we should set a rate forever. I was just saying that
you
can have one module that determines in real-time the best rate to use at

that moment and then it tells the codec what to use. If it needs help
from
the codec, then we can define special functionality.

>> On top of that, the codec may be
>> able to do a bit better by providing a "service" to the other layers.
On
>> example here (also included in the draft) is VBR, which means that
the
>> application specifies an average rate and the codec decides how to
>> vary the
>> rate to achieve that (subject to some rate control constraints).
Would
>> other such mechanisms be useful to deal with congestion?
>
> I'm not sure this would work, because congestion control will
basically
> change the VBR input parameter continuously.

Changing the VBR parameters continuously is not a problem. VBR just
means
that you'd like to have an average of N kb/s for now, but you're OK if
there are variations from frame to frame.

        Jean-Marc
_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec
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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Sorry, my link was stripped. As per sp=
eex
documentation and my experience, one of the features of speex is &#8220;</s=
pan></font>Integration
of narrowband<a name=3D104></a> and wideband<a name=3D105></a> using an emb=
edded
bit-stream&#8221;<o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Jozsef</span></font><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p></o:p></span><=
/font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Bo=
rilin
[mailto:Borilin@spiritdsp.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, 20=
09
2:14 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Stephan Wen=
ger;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Speex is not layered, isn&#8217;t it?<=
o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span lang=3D=
RU
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best regards,</span=
></font><font
color=3Dnavy><span lang=3DRU style=3D'color:navy'><o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span lang=3D=
RU
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava Borilin</span=
></font><span
lang=3DRU><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Jozsef V=
ass
[mailto:jovass@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, October 27, 2=
009
12:09 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Stephan Wenger; Slava Bo=
rilin;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span lang=3DR=
U
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>In rfc 3984 processing packets by mane=
 has
some clear benefits, e.g., discarding packets that the decoder would descri=
be
anyways (section 7.3), dropping packets in case of congestion, etc.<o:p></o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Here, we mostly talking about reducing
packet size, and I am not sure how beneficial it is regarding the overall
system. Consider speex. Quality level six, wideband and narrowband encoder
produces 52 bytes and 28 bytes of payload, respectively. Adding IP/UDP/RTP
header, you go from 92 bytes to 68 bytes. Since packet rate is the same, I =
do
not think it will affect congestion too much. Please also consider that
processing packets by intermediary, adds additional delay.<o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jozsef<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Stephan =
Wenger
[mailto:stewe@stewe.org] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, October 26, 20=
09
11:15 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jozsef Vass; Slava Boril=
in;
Henry Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 face=3DC=
alibri><span
style=3D'font-size:11.0pt;font-family:Calibri'>I think that we have a termi=
nology
problem here.<br>
<br>
Routers are AFAIK devices that route packets at ISO/OSI layer 3; that is th=
ey
don&#8217;t modify IP packets. &nbsp;Boxes that look deep into the IP packe=
ts, and
(potentially) modify them, are higher layer devices. &nbsp;After long
discussions in AVT at the time, RFC 3984 and its successors call those boxe=
s
&#8220;Media Aware Network Elements&#8221;, MANEs. &nbsp;MANEs modify packe=
ts at the
application layer, for example to remove layers, re-package H.264 NAL units=
 to
adapt to different MTU sizes, and do all kinds of other things that violate=
 the
end-to-end principle. &nbsp;A MANE has business in looking into, and
potentially modifying, IP/UDP/RTP packets, a router has not.<br>
<br>
When MANEs are in the picture, it&#8217;s IMO totally acceptable (from both=
 an
abstract layering theory viewpoint and in practice) that packets are being
opened and the content is repackaged. &nbsp;It&#8217;s actually common prac=
tice.<br>
<br>
Stephan<br>
<br>
On 10/26/09 10:55 AM, &quot;Jozsef Vass&quot; &lt;<a href=3D"jovass@adobe.c=
om">jovass@adobe.com</a>&gt;
wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do not like the idea of sending mult=
iple
layers in a packet either. I feel very uncomfortable of routers modifying
packets. It is OK not to deliver, but delivering semantically modified pack=
et
is not expected.<br>
&nbsp;<br>
Using layers for video totally makes sense where the rates are much higher =
and
layers can be transported in separate packets.<br>
&nbsp;<br>
Jozsef<br>
&nbsp;</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> Slava Borilin [<a
href=3D"mailto:Borilin@spiritdsp.com">mailto:Borilin@spiritdsp.com</a>] <br=
>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
12:23 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Henry
Sinnreich; Jean-Marc Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Again - my point is that router does NOT have to know tha=
t something
specifically is the codec, or specific codec- the only goal is to let route=
r
aware that the stream (and every packet) is LAYERED, and that it is legitim=
ate
to strip out some layers if necessary due to congestion.<br>
The content of such &#8220;layered&#8221; can be any &#8211; voice codec, v=
ideo codec, audio
codec, any type of content.<br>
&nbsp;<br>
And I agree with Gregory &#8211; that having current infrastructure using l=
ayers in
separate streams will require too much overhead (packets) which does not ma=
ke
sense as well &#8211; and I am thinking about one stream of packets comtain=
ing all
the layers, which, however, can be modified on the fly &#8220;in the middle=
&#8221;.<br>
So we may think about more generic concept of &#8220;declared layered conte=
nt&#8221; which
does NOT loose the information (well, most of it) by stripping out portions=
 of
the packet and reducing the traffic, so router can strip out any type of
layered content, without going deep into specific codec.<br>
&nbsp;<br>
Again- this is &#8220;na=EFve&#8221; suggestions from me (very limited engi=
neer) &#8211; and
definitely I am looking for objections even more then for agreement &#8211;=
 for the
sake of either finding the idea useful or making judged deny of it, and I a=
gree
that this concept is only one possible way of working with congestion.<br>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-fam=
ily:Tahoma;
font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> <a href=3D"codec-bounces@iet=
f.org">codec-bounces@ietf.org</a>
[<a href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a=
>] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Slava Borilin<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:13 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Henry Sinnreich; Jean-Ma=
rc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;fo=
nt-family:
Arial;color:navy'>Henry,<br>
With all of my respect (you know) &#8211; this is too high-level considerat=
ion and
too generic statement which doesn&#8217;t answer MPLS issue for example.<br=
>
&nbsp;<br>
</span></font><font size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt=
;
font-family:Calibri'><br>
</span></font><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-=
size:10.0pt;
font-family:Arial;color:navy'>best regards,<br>
Slava Borilin</span></font><o:p></o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p style=3D'margin-bottom:12.0pt'><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span>=
</font></b><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>=
 Henry
Sinnreich [<a href=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com<=
/a>] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, October 22, =
2009
11:10 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Slava Borilin; Jean-Marc
Valin; Lars Eggert<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <a href=3D"codec@ietf.or=
g">codec@ietf.org</a><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [codec] updated
charter proposal<br>
</span></font><br>
<font size=3D5 face=3DCalibri><span style=3D'font-size:18.0pt;font-family:C=
alibri'>Slava,<br>
<br>
&gt;I am not that sure that routers should not be aware of the codecs. Why?=
<br>
&gt;If we are building MPLS and other systems that is content-dependent, wh=
y<br>
&gt;not let router understand the codec?<br>
<br>
The essence of the Dumb Network (Internet) and its routers is that it knows
nothing about the applications.<br>
All applications that nobody planned for can run over it.<br>
The tragedy of the telephone network was is knew everything about its only =
one
application.<br>
<br>
Henry<br>
<br>
<br>
On 10/22/09 1:23 PM, &quot;Slava Borilin&quot; &lt;<a
href=3D"Borilin@spiritdsp.com">Borilin@spiritdsp.com</a>&gt; wrote:<br>
I am not that sure that routers should not be aware of the codecs. Why?<br>
If we are building MPLS and other systems that is content-dependent, why<br=
>
not let router understand the codec?<br>
<br>
Especially if we have hordes of voice channels going thru router, it can<br=
>
be too hard and too long to tell each sender to reduce the rate - why<br>
not to leave this freedom to router as this may be more efficient for<br>
congestion recovery?<br>
<br>
Same applies to H.264SVC for example - at the end of the day, router do<br>
not need to understand anything except that it is LAYERED streams (even<br>
don't know whether its voice, video or audio or anything else of layered<br=
>
nature, and router &quot;has the right&quot; to strip out layers in certain
order<br>
to prevent congestion?<br>
<br>
best regards,<br>
Slava Borilin<br>
<br>
-----Original Message-----<br>
From: <a href=3D"codec-bounces@ietf.org">codec-bounces@ietf.org</a> [<a
href=3D"mailto:codec-bounces@ietf.org">mailto:codec-bounces@ietf.org</a>] O=
n
Behalf<br>
Of Jean-Marc Valin<br>
Sent: Thursday, October 22, 2009 5:45 PM<br>
To: Lars Eggert<br>
Cc: <a href=3D"codec@ietf.org">codec@ietf.org</a><br>
Subject: Re: [codec] updated charter proposal<br>
<br>
Lars Eggert wrote:<br>
&gt; Also, what we've learned from media over DCCP is that if you don't<br>
&gt; involve the codec in some form, performance/quality will be bad. Some<=
br>
&gt; codecs need time to adjust the rate. Some can only adjust it in pretty=
<br>
<br>
&gt; coarse steps. Etc. If you let DCCP adjust the rate blindly, without<br=
>
&gt; involving the codec in some form, you might get very good TCP<br>
&gt; friendliness, but maybe quality will suffer.<br>
<br>
I'm totally in favour of &quot;involving the codec&quot; in here. I just wa=
nt
to<br>
be<br>
careful not to include in the codec things that don't belong there (e.g.<br=
>
a<br>
DCCP implementation or a whole jitter buffer).<br>
<br>
&gt; (And there is always the ramping up part - congestion control will<br>
never<br>
&gt; attempt to probe for more capacity unless the codec generates more<br>
data,<br>
&gt; and why would the codec do that if it's converged to transmit at some<=
br>
&gt; lower rate.)<br>
<br>
Well, the codec transmits at the rate you ask it to use. If you want to<br>
probe for more capacity, then you tell the codec that it can use a<br>
higher<br>
rate. Now, if there's a special way in which you want to ask the codec,<br>
then maybe that can become an additional feature for the codec.<br>
<br>
&gt; By no means should routers have to understand codec formats! Routers<b=
r>
&gt; drop or mark IP packets. It is up to the application sending those IP<=
br>
&gt; packets to react appropriately to the marks or losses. In other words,=
<br>
a<br>
&gt; layered codec will see losses/marks to packets sent on all layers, and=
<br>
<br>
&gt; will then need to figure out whether to strip off a layer or not in<br=
>
&gt; response to whatever congestion level those losses/marks indicate.<br>
<br>
I definitely agree that the routers shouldn't know about codecs and this<br=
>
is<br>
why I was saying that IMO layered codecs aren't too useful for<br>
congestion<br>
control (though they have other uses). If there's nothing in the middle<br>
of<br>
the network to strip off the layers, then you might as well not have<br>
layers<br>
at all. All you need is to ask the sender to reduce the rate. You don't<br>
need layers to have a multi-rate codec. Layering just means that you can<br=
>
<br>
strip off bits and still have the result sound OK.<br>
<br>
&gt;&gt; OK, I may be wrong here, but my understanding is that how much the=
<br>
rate<br>
&gt;&gt; should change depending on congestion is actually not that<br>
&gt;&gt; codec-specific.<br>
&gt;<br>
&gt; It *can* be non-codec-specific, see DCCP for example. But quality will=
<br>
<br>
&gt; suffer IMO.<br>
<br>
Well, we need to figure out the aspects that should be codec-specific<br>
and<br>
implement just those. Does anyone have a list of the issues codecs<br>
should<br>
be aware of?<br>
<br>
&gt; Most Internet congestion control is based on *creating* losses/marks<b=
r>
to<br>
&gt; determine capacity. People have tried to create congestion control<br>
&gt; schemes that avoid causing losses (delay-based schemed, for example),<=
br>
&gt; but they are perceived to be more fragile.<br>
<br>
Yes, I can imagine the problem not being trivial at all.<br>
<br>
&gt;&gt; Once you figure this out, you<br>
&gt;&gt; can just tell the codec to use that rate.<br>
&gt;<br>
&gt; It's not that easy, because rates in the Internet are never stable -<b=
r>
the<br>
&gt; rate computation is usually clocked by the RTT. You can obviously<br>
dampen<br>
&gt; this somehow, but you can never say &quot;now I have found a rate that=
 I<br>
can<br>
&gt; safely use forever.&quot;<br>
<br>
Oh, I never said we should set a rate forever. I was just saying that<br>
you<br>
can have one module that determines in real-time the best rate to use at<br=
>
<br>
that moment and then it tells the codec what to use. If it needs help<br>
from<br>
the codec, then we can define special functionality.<br>
<br>
&gt;&gt; On top of that, the codec may be<br>
&gt;&gt; able to do a bit better by providing a &quot;service&quot; to the
other layers.<br>
On<br>
&gt;&gt; example here (also included in the draft) is VBR, which means that=
<br>
the<br>
&gt;&gt; application specifies an average rate and the codec decides how to=
<br>
&gt;&gt; vary the<br>
&gt;&gt; rate to achieve that (subject to some rate control constraints).<b=
r>
Would<br>
&gt;&gt; other such mechanisms be useful to deal with congestion?<br>
&gt;<br>
&gt; I'm not sure this would work, because congestion control will<br>
basically<br>
&gt; change the VBR input parameter continuously.<br>
<br>
Changing the VBR parameters continuously is not a problem. VBR just<br>
means<br>
that you'd like to have an average of N kb/s for now, but you're OK if<br>
there are variations from frame to frame.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Jean-Marc<br>
_______________________________________________<br>
codec mailing list<br>
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g/mailman/listinfo/codec</a></span></font><font
size=3D2 face=3DCalibri><span style=3D'font-size:11.0pt;font-family:Calibri=
'><o:p></o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
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</span></font></div>

<p class=3DMsoNormal><font size=3D2 face=3DConsolas><span style=3D'font-siz=
e:10.0pt;
font-family:Consolas'>_______________________________________________<br>
codec mailing list<br>
<a href=3D"codec@ietf.org">codec@ietf.org</a><br>
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From bmschwar@fas.harvard.edu  Mon Oct 26 14:48:42 2009
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Stephan Wenger wrote:
> When operating in low delay environments (as it is the assumption here), I
> agree that there is not much point in attempting to reduce the packet
> count---doing so would increase delay and would presumably be unacceptable
> to the application.

If decreased packet count is necessary in order for my VoIP session to
work on the connection I'm using, then of course I want my client to
reduce the packet count.  Increased delay is better than no connection.
As is usual on the internet, graceful degradation is almost always
preferable to terminating a session and throwing up a

"WARNING: Latency guarantee could not be satisfied."

> If you have thousands or millions of sessions
> on the same fiber, dropping layers could be argued to make a difference.

Again, you don't seem to be talking about the public internet.  Even on a
private network, I still don't see much of an argument for layering.  The
bitrate reduction could more easily be achieved at the encoder.  Moreover,
layered encoding is typically less efficient than "unlayered" encoding at
any particular bitrate.

The only sensible argument I'm aware of for layered streams is in
heterogeneous multicast networks.  We can revisit this issue if multicast
ever becomes widely available on the public internet
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From lars.eggert@nokia.com  Tue Oct 27 03:57:44 2009
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From: Lars Eggert <lars.eggert@nokia.com>
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Cc: draft-ietf-tsvwg-byte-pkt-congest@tools.ietf.org
Subject: Re: [codec] updated charter proposal
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Hi,

On 2009-10-26, at 23:28, Stephan Wenger wrote:
> I think that these arguments boil down to the discussion of =20
> measurement of congestion in the packet count domain or in the =20
> bitrate domain.  I don=92t think there=92s a verdict yet which of the =20=

> two are really important.  It may very well depend on your viewpoint =20=

> and on what your company makes money of.

FYI, http://tools.ietf.org/html/draft-ietf-tsvwg-byte-pkt-congest =20
discusses byte vs. packet congestion in more detail. It talks about =20
the generic data transport case and not specifically media traffic =20
though.

> When operating in low delay environments (as it is the assumption =20
> here), I agree that there is not much point in attempting to reduce =20=

> the packet count---doing so would increase delay and would =20
> presumably be unacceptable to the application.  Therefore, =20
> congestion friendliness, when congestion is measured in the packet =20
> count domain, seems to be independent from the codec =20
> characteristics; as long as the codec is optimized for low delay.
>
> However, if we start from the assumption that congestion is measured =20=

> in the bitrate domain, then the ballgame is somewhat different.  =20
> Yes, when we drop a layer from a hypothetical layered audio codec, =20
> we would not safe a lot of bits in absolute numbers, and yes, the =20
> percentage would be even more disappointing considering the =20
> packetization overhead.  But still, we would reduce the overall =20
> bitrate.  If you have thousands or millions of sessions on the same =20=

> fiber, dropping layers could be argued to make a difference.

Lars=

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From: "Roni Even" <ron.even.tlv@gmail.com>
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Subject: [codec] Review of the proposed charter
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Hi,

I read the proposed charter and in general is a good start I still have some
comments

 

1.  In the deliverable section "Guidelines that define the work process of
the group, using draft-valin-codec-guidelines as the basis for achieving
consensus. This document shall be Informational." Does this mean that once
the charter is approved we are also approving the mentioned ID as a WG
document for guideline. Otherwise I think that the text should be
"Guidelines that define the work process of the group. This document shall
be Informational."

2.  The same comment for the next section about requirements. I also think
that if the requirement will include some metrics used to test the codec
this draft should have Normative context and may be a standard track
document

3.  The last section of the deliverable "Specification of one or more codecs
that meet the agreed-upon requirements," I think that if the proposed
working group is going to define more than one codec, each codec should have
his own requirement draft giving motivation and the specific requirement for
having a different codec from the first proposed one that will be developed
based on the requirement document from the previous section. I do not think
that one set of requirement merit more than one codec.

Thanks

Roni Even

 

  _____  


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<div class=3DSection1>

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

<p class=3DMsoNormal>I read the proposed charter and in general is a =
good start I
still have some comments<o:p></o:p></p>

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

<p class=3DMsoPlainText =
style=3D'margin-left:.5in;text-indent:-.25in;mso-list:l2 level1 =
lfo3'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span><![endif]><span
dir=3DLTR></span>In the deliverable section &quot;Guidelines that define =
the work
process of the group, using draft-valin-codec-guidelines as the basis =
for
achieving consensus. This document shall be Informational.&quot; Does =
this mean
that once the charter is approved we are also approving the mentioned ID =
as a
WG document for guideline. Otherwise I think that the text should be =
&nbsp;&quot;Guidelines
that define the work process of the group. This document shall be
Informational.&quot;<o:p></o:p></p>

<p class=3DMsoPlainText =
style=3D'margin-left:.5in;text-indent:-.25in;mso-list:l2 level1 =
lfo3'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span><![endif]><span
dir=3DLTR></span>The same comment for the next section about =
requirements. I also
think that if the requirement will include some metrics used to test the =
codec
this draft should have Normative context and may be a standard track =
document<o:p></o:p></p>

<p class=3DMsoPlainText =
style=3D'margin-left:.5in;text-indent:-.25in;mso-list:l2 level1 =
lfo3'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span><![endif]><span
dir=3DLTR></span>The last section of the deliverable &quot;Specification =
of one
or more codecs that meet the agreed-upon requirements,&quot; I think =
that if the
proposed working group is going to define more than one codec, each =
codec
should have his own requirement draft giving motivation and the specific =
requirement
for having a different codec from the first proposed one that will be =
developed
based on the requirement document from the previous section. I do not =
think
that one set of requirement merit more than one codec.<o:p></o:p></p>

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

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

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

</div>

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------=_NextPart_000_01A1_01CA5728.FAA758A0--


From stpeter@stpeter.im  Tue Oct 27 13:40:22 2009
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From: Peter Saint-Andre <stpeter@stpeter.im>
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 10/27/09 9:14 AM, Roni Even wrote:
> Hi,
> 
> I read the proposed charter and in general is a good start I still have
> some comments
> 
> 1.  In the deliverable section "Guidelines that define the work process
> of the group, using draft-valin-codec-guidelines as the basis for
> achieving consensus. This document shall be Informational." Does this
> mean that once the charter is approved we are also approving the
> mentioned ID as a WG document for guideline. Otherwise I think that the
> text should be  "Guidelines that define the work process of the group.
> This document shall be Informational."
> 
> 2.  The same comment for the next section about requirements. 

The charter is the WG's contract with the IESG. I think it is much
better to show the IESG that the group has been actively working on both
the guidelines and the requirements by pointing to specific I-Ds,
especially because the major issue raised in Stockholm was the lack of
clear guidelines and requirements. IMHO draft-valin-codec-guidelines and
draft-valin-codec-requirements would form the basis of discussion by the
WG (if formed), so yes I think the WG would adopt them as WG items,
although I think that a final decision will be up to the WG if and when
it is formed.

> I also
> think that if the requirement will include some metrics used to test the
> codec this draft should have Normative context and may be a standard
> track document

Do you see that as a separate document or as part of the requirements
document?

> 3.  The last section of the deliverable "Specification of one or more
> codecs that meet the agreed-upon requirements," I think that if the
> proposed working group is going to define more than one codec, each
> codec should have his own requirement draft giving motivation and the
> specific requirement for having a different codec from the first
> proposed one that will be developed based on the requirement document
> from the previous section. I do not think that one set of requirement
> merit more than one codec.

That seems unnecessarily complex. IMHO it would be simpler for the
requirements document to define the requirements in as much detail as
possible, while leaving some requirements flexible by specifying ranges
of acceptable behavior or performance, dimensions that a particular
codec could legitimately optimize for, etc. Then each codec spec would
state the more particular requirements it has tried to meet, without
writing a separate requirements document for each codec spec. That
sounds like an awful lot of documents for very little benefit, but I
think that these editorial issues aren't something that needs to be in
the charter and can be worked out by the WG (if approved).

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Peter Saint-Andre'" <stpeter@stpeter.im>
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Hi Peter,
I think that you can mention the documents to the IESG but they cannot =
be part of the deliverable since they are ID that did not go yet through =
any consensus.

As for you response on multiple codec, it look to me that the group will =
be a codec factory where anyone can define a codec based on part of the =
requirements and still be inline with the charter. I find it strange =
that you suggest that the group will develop requirements that do not =
need to be fully answered by one codec and let everybody select part of =
them and develop a codec. I am not aware that the IETF does two =
competing solutions for the same requirement document.

When the discussion on doing a codec in the IETF was started it was =
about defining a codec for wideband audio that will be easily available =
in order to help with deployment of wideband in Internet applications. =
Doing multiple codecs will not answer this basic need of =
interoperability.

Regards
Roni Even

> -----Original Message-----
> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]
> Sent: Tuesday, October 27, 2009 10:41 PM
> To: Roni Even
> Cc: codec@ietf.org
> Subject: Re: [codec] Review of the proposed charter
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 10/27/09 9:14 AM, Roni Even wrote:
> > Hi,
> >
> > I read the proposed charter and in general is a good start I still
> have
> > some comments
> >
> > 1.  In the deliverable section "Guidelines that define the work
> process
> > of the group, using draft-valin-codec-guidelines as the basis for
> > achieving consensus. This document shall be Informational." Does =
this
> > mean that once the charter is approved we are also approving the
> > mentioned ID as a WG document for guideline. Otherwise I think that
> the
> > text should be  "Guidelines that define the work process of the
> group.
> > This document shall be Informational."
> >
> > 2.  The same comment for the next section about requirements.
>=20
> The charter is the WG's contract with the IESG. I think it is much
> better to show the IESG that the group has been actively working on
> both
> the guidelines and the requirements by pointing to specific I-Ds,
> especially because the major issue raised in Stockholm was the lack of
> clear guidelines and requirements. IMHO draft-valin-codec-guidelines
> and
> draft-valin-codec-requirements would form the basis of discussion by
> the
> WG (if formed), so yes I think the WG would adopt them as WG items,
> although I think that a final decision will be up to the WG if and =
when
> it is formed.
>=20
> > I also
> > think that if the requirement will include some metrics used to test
> the
> > codec this draft should have Normative context and may be a standard
> > track document
>=20
> Do you see that as a separate document or as part of the requirements
> document?
>=20
> > 3.  The last section of the deliverable "Specification of one or =
more
> > codecs that meet the agreed-upon requirements," I think that if the
> > proposed working group is going to define more than one codec, each
> > codec should have his own requirement draft giving motivation and =
the
> > specific requirement for having a different codec from the first
> > proposed one that will be developed based on the requirement =
document
> > from the previous section. I do not think that one set of =
requirement
> > merit more than one codec.
>=20
> That seems unnecessarily complex. IMHO it would be simpler for the
> requirements document to define the requirements in as much detail as
> possible, while leaving some requirements flexible by specifying =
ranges
> of acceptable behavior or performance, dimensions that a particular
> codec could legitimately optimize for, etc. Then each codec spec would
> state the more particular requirements it has tried to meet, without
> writing a separate requirements document for each codec spec. That
> sounds like an awful lot of documents for very little benefit, but I
> think that these editorial issues aren't something that needs to be in
> the charter and can be worked out by the WG (if approved).
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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> =3Dc5PQ
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On 10/27/09 3:16 PM, Roni Even wrote:
> Hi Peter, I think that you can mention the documents to the IESG but
> they cannot be part of the deliverable since they are ID that did not
> go yet through any consensus.

I freely grant that consensus has not yet been reached regarding the
guidelines and requirements I-Ds, but to expect consensus before a WG
has even been formed is a novel interpretation of the Internet Standards
Process! The proposed charter states that these I-Ds will form the basis
for discussion within the WG, hopefully leading to eventual consensus,
WGLC, and IETF Last Call. That is the normal process for discussion
within the IETF. Do you suggest that consensus should be reached (and by
whom?) before a charter can even mention an I-D as the basis for further
discussion and refinement? And on what basis could such consensus be
reached, given that a WG has not even been formed yet? This is a
Catch-22 and would prevent the formation of any WG, unless you suggest a
process by which IETF LC would be necessary for WG formation.

> As for you response on multiple codec, it look to me that the group
> will be a codec factory where anyone can define a codec based on part
> of the requirements and still be inline with the charter. 

The proposed charter states:

###

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor to produce as few codecs as possible, and in any case shall not
function as a "codec factory" that produces a large number of codecs.

###

That clearly states the preference of the group to NOT be a codec factory.

> I find it
> strange that you suggest that the group will develop requirements
> that do not need to be fully answered by one codec and let everybody
> select part of them and develop a codec. 

What I am suggesting is a conceptual approach. Let me provide an
example. The requirements document as it stands today states that the
operating space includes medium delay (20-30 ms) applications from
narrowband (8 kb/s to 16 kb/s) to full band (32 to 80 kb/s) and very low
delay (< 10 ms) applications from super wideband (32 to 80 kb/s) to full
band (48 to 128 kb/s). The requirements document does not say that we
must produce "one codec to bind them all" -- one codec that will work
optimally for everything from medium delay + narrowband to very low
delay + full band. Yes, that would be great, but perhaps it is not
possible. IMHO we might produce two codecs, one that is optimized for
certain parts of the operating space and another that is optimized for
oher parts of the operating space. To say that the WG must produce one
and only codec for all parts of the operating space implies that the WG
would need to ignore facts the group might discover as it tries to solve
the problem.

> I am not aware that the IETF
> does two competing solutions for the same requirement document.

RFC 2779 defined requirements that have been met by both the XMPP WG and
the SIMPLE WG.

However, in the codec space the solutions would not be competing, they
would be complementary because they would be optimized for different
aspects of the operating space.

Would it help to add some text to the effect that, if the WG finds that
it must produce multiple codecs, it would try to produce codecs that are
complementary instead of competing?

> When the discussion on doing a codec in the IETF was started it was
> about defining a codec for wideband audio that will be easily
> available in order to help with deployment of wideband in Internet
> applications. Doing multiple codecs will not answer this basic need
> of interoperability.

The world seems to contain dozens or hundreds of codecs that are
optimized for different aspects of media encoding and communication. In
fact other SDOs have produced many, many codecs without, it seems, doing
permanent damage to the Internet. Yes, ideally an IETF Codec WG would
produce one codec to bind them all, but to require that in the charter
seems quite unhelpful to me because it would unnecessarily restrict the
WG from finding optimal solutions.

Peter
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On 10/27/09 4:11 PM, Peter Saint-Andre wrote:
> On 10/27/09 3:16 PM, Roni Even wrote:
>> Hi Peter, I think that you can mention the documents to the IESG but
>> they cannot be part of the deliverable since they are ID that did not
>> go yet through any consensus.
> 
> I freely grant that consensus has not yet been reached regarding the
> guidelines and requirements I-Ds, but to expect consensus before a WG
> has even been formed is a novel interpretation of the Internet Standards
> Process! The proposed charter states that these I-Ds will form the basis
> for discussion within the WG, hopefully leading to eventual consensus,
> WGLC, and IETF Last Call. That is the normal process for discussion
> within the IETF. Do you suggest that consensus should be reached (and by
> whom?) before a charter can even mention an I-D as the basis for further
> discussion and refinement? And on what basis could such consensus be
> reached, given that a WG has not even been formed yet? This is a
> Catch-22 and would prevent the formation of any WG, unless you suggest a
> process by which IETF LC would be necessary for WG formation.

Hi Roni,

I'm sorry, I misunderstood your point. We mention the existing I-Ds in
two places. In the "Objectives" section we say:

###

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

###

And in the "Deliverables" section we say:

###

1. Guidelines that define the work process of the group, using
draft-valin-codec-guidelines as the starting point for working toward
consensus. This document shall be Informational.

2. Detailed technical requirements, using draft-valin-codec-requirements
as the starting point for working toward consensus.  This document
shall be Informational.

###

I thought you wanted to remove these mentions in both places. I think
you're right that the "Deliverables" section would be better without
mentioning specific drafts, because the WG needs to officially accept
I-Ds as WG items. So I think the "Deliverables" section needs to say:

###

1. Guidelines that define the work process of the group. This
document shall be Informational.

2. Detailed technical requirements. This document shall be
Informational.

###

My apologies for the misunderstanding.

Do you think it is acceptable to mention these I-Ds earlier in the
charter text?

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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The following text incorporates all feedback received to date.

###

Proposed Charter for Codec WG
Last Updated: 2009-10-28

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.

Objectives

The goal of this group is to develop one or more high-quality audio
codecs that are optimized for use over the Internet and that can be
widely implemented and easily distributed among application developers,
service operators, and end users. Core technical considerations include
but are not necessarily limited to the following:

1. Use in interactive applications (examples include but are not
limited to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music
performance)

2. Addressing the real transport conditions of the Internet as
identified and prioritized by the group

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
the result should not depend on the details of any particular signalling
technology

Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor to produce as few codecs as possible, and in any case shall not
function as a "codec factory" that produces a large number of codecs.

In completing its work, the group should collaborate with other IETF
working groups to complete particular tasks.  These might include but
would not be limited to the following:

- - Within the AVT WG, define the codec's payload format for use with
  the Real-time Transport Protocol (RTP).

- - Collaborate with working groups in the Transport area to identify
  important aspects of packet transmission over the Internet.

- - Collaborate with working groups in the Transport area to understand
  the degree of rate adaptation desirable, and to reflect that
  understanding in the design of a codec that can adjust its
  transmission in a way that minimizes disruption to the audio.

- - Collaborate with working groups in the RAI area to ensure that
  information about and negotiation of the codec can be easily
  represented at the signalling layer.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

The group would prefer to produce a codec that can be widely
implemented and easily distributed among application developers,
service operators, and end users.  This is important because many
existing codecs that might fulfill some or most of the technical
attributes listed above are encumbered in various ways.  For example,
patent holders might require that those wishing to implement the codec
in software, deploy the codec in a service, or distribute the codec
in software or hardware need to request a license, enter into a
business agreement, pay licensing fees or royalties, or adhere to
other special conditions or restrictions.

Because such encumbrances have made it difficult to widely implement
and easily distribute high-quality audio codecs across the entire
Internet community, the group shall prefer unencumbered technologies in
a way that is consistent with BCP 78 and BCP 79.  In particular, the
group shall heed the preference stated in BCP 79: "In general, IETF
working groups prefer technologies with no known IPR claims or, for
technologies with claims against them, an offer of royalty-free
licensing."  Although this preference cannot guarantee that the group
will produce an unencumbered codec, the group shall attempt to adhere
to the spirit of BCP 79.  This preference does not explicitly rule out
the possibility of adapting encumbered technologies; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group. This
document shall be Informational.

2. Detailed technical requirements. This document shall be
Informational.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation, as well as
any specific technical requirements for which the codec has been
optimized. The text description of each codec shall indicate which
components of the encoder and decoder are mandatory, recommended, and
optional.  It is envisioned that each such specification shall be a
Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: WGLC on Requirements
Jul-2010: Requirements to IESG (Infomational)
Mar-2011: WGLC on codec specification(s)
Jun-2011: Submit codec specification(s) to IESG (Standards Track)

###

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Peter,
Thanks, I am trying to be helpful and this was exactly my meaning on the =
first two items in the deliverables section.
As for the objective I think it is acceptable to mention documents, it =
just need to be clear that they are not yet based on full consensus as =
can be seen on the mailing list and that there may be other requirement =
documents.

The only point I still feel is open is about how to decide which =
proposed codecs are within the charter of the group. Having it too open =
as it is now, will not help you with IESG, but this is my personal view =
based on my experience with my WG charter.


Roni Even

> -----Original Message-----
> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]
> Sent: Wednesday, October 28, 2009 6:34 PM
> To: Roni Even
> Cc: codec@ietf.org
> Subject: Re: [codec] Review of the proposed charter
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 10/27/09 4:11 PM, Peter Saint-Andre wrote:
> > On 10/27/09 3:16 PM, Roni Even wrote:
> >> Hi Peter, I think that you can mention the documents to the IESG =
but
> >> they cannot be part of the deliverable since they are ID that did
> not
> >> go yet through any consensus.
> >
> > I freely grant that consensus has not yet been reached regarding the
> > guidelines and requirements I-Ds, but to expect consensus before a =
WG
> > has even been formed is a novel interpretation of the Internet
> Standards
> > Process! The proposed charter states that these I-Ds will form the
> basis
> > for discussion within the WG, hopefully leading to eventual
> consensus,
> > WGLC, and IETF Last Call. That is the normal process for discussion
> > within the IETF. Do you suggest that consensus should be reached =
(and
> by
> > whom?) before a charter can even mention an I-D as the basis for
> further
> > discussion and refinement? And on what basis could such consensus be
> > reached, given that a WG has not even been formed yet? This is a
> > Catch-22 and would prevent the formation of any WG, unless you
> suggest a
> > process by which IETF LC would be necessary for WG formation.
>=20
> Hi Roni,
>=20
> I'm sorry, I misunderstood your point. We mention the existing I-Ds in
> two places. In the "Objectives" section we say:
>=20
> ###
>=20
> Work processes and technical requirements for achieving the foregoing
> objectives are outlined in draft-valin-codec-guidelines and
> draft-valin-codec-requirements respectively; these documents will form
> the starting point for working toward consensus and will be refined by
> the group in accordance with the usual IETF procedures.
>=20
> ###
>=20
> And in the "Deliverables" section we say:
>=20
> ###
>=20
> 1. Guidelines that define the work process of the group, using
> draft-valin-codec-guidelines as the starting point for working toward
> consensus. This document shall be Informational.
>=20
> 2. Detailed technical requirements, using draft-valin-codec-
> requirements
> as the starting point for working toward consensus.  This document
> shall be Informational.
>=20
> ###
>=20
> I thought you wanted to remove these mentions in both places. I think
> you're right that the "Deliverables" section would be better without
> mentioning specific drafts, because the WG needs to officially accept
> I-Ds as WG items. So I think the "Deliverables" section needs to say:
>=20
> ###
>=20
> 1. Guidelines that define the work process of the group. This
> document shall be Informational.
>=20
> 2. Detailed technical requirements. This document shall be
> Informational.
>=20
> ###
>=20
> My apologies for the misunderstanding.
>=20
> Do you think it is acceptable to mention these I-Ds earlier in the
> charter text?
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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> GaEAnikpDFIYfYudGyvAPOkLwW5vpICd
> =3DowS6
> -----END PGP SIGNATURE-----


From mknappe@juniper.net  Thu Oct 29 11:09:06 2009
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Date: Thu, 29 Oct 2009 11:06:58 -0700
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Catching up on the charter discussion, please disregard if content below ha=
s already been covered...



Scoping for potentially more than one candidate codec is a real strength of=
 the proposed charter, and will help avoid many potential process pitfalls,=
 but it does open up challenges.

Given the often differing or differently prioritized requirements for each =
class of interactive audio application, a one-size-fits-all set of criteria=
 is most likely intractable. As we move further with this selection process=
, it may be helpful to set up selection 'gates' that begin with meeting a c=
ommon set of basic and relatively high level criteria, followed by subseque=
nt more narrow gates with stricter selection criteria dependent on the clas=
s(es) of application targeted by the codec. These subsequent candidate sele=
ction paths and gates would run in parallel as resources permit. This shoul=
d allow concentration on the true merits of a particular codec, and not on =
the relative importance of the application(s) it is best suited for.

Thoughts?

Mike


On 10/28/09 1:11 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

The following text incorporates all feedback received to date.

###

Proposed Charter for Codec WG
Last Updated: 2009-10-28

Problem Statement

There are no standardized, high-quality audio codecs that are optimized
for use in interactive Internet applications and that can be widely
implemented and easily distributed among application developers,
service operators, and end users.  This gap in the Internet protocol
stack has hindered protocol designers from being able to specify
mandatory-to-implement codecs in their protocols for the sake of
interoperability, developers from building innovative, interactive
applications, service operators from deploying affordable, high-quality
services, and end users from enjoying an optimal user experience.

Objectives

The goal of this group is to develop one or more high-quality audio
codecs that are optimized for use over the Internet and that can be
widely implemented and easily distributed among application developers,
service operators, and end users. Core technical considerations include
but are not necessarily limited to the following:

1. Use in interactive applications (examples include but are not
limited to point-to-point voice calls, multi-party voice conferencing,
telepresence, teleoperation, in-game voice chat, and live music
performance)

2. Addressing the real transport conditions of the Internet as
identified and prioritized by the group

3. Ensuring interoperability with the Real-time Transport Protocol
(RTP), including secure transport via SRTP

4. Ensuring interoperability with Internet signalling technologies such
as Session Initiation Protocol (SIP), Session Description Protocol
(SDP), and Extensible Messaging and Presence Protocol (XMPP); however,
the result should not depend on the details of any particular signalling
technology

Codecs optimized for very low bit rates (typically below 2.4 kbps) and
codecs for non-interactive audio are out of scope because they might
necessitate specialized optimizations.

Although the codec(s) produced by the group might be used as
mandatory-to-implement technologies by designers of particular Internet
protocols, it is explicitly not a goal of the group to produce a codec
that will be mandated for use across the entire IETF or Internet community.

The group might produce only one codec or it might produce multiple
codecs; however, to reduce confusion in the marketplace it shall
endeavor to produce as few codecs as possible, and in any case shall not
function as a "codec factory" that produces a large number of codecs.

In completing its work, the group should collaborate with other IETF
working groups to complete particular tasks.  These might include but
would not be limited to the following:

- - Within the AVT WG, define the codec's payload format for use with
  the Real-time Transport Protocol (RTP).

- - Collaborate with working groups in the Transport area to identify
  important aspects of packet transmission over the Internet.

- - Collaborate with working groups in the Transport area to understand
  the degree of rate adaptation desirable, and to reflect that
  understanding in the design of a codec that can adjust its
  transmission in a way that minimizes disruption to the audio.

- - Collaborate with working groups in the RAI area to ensure that
  information about and negotiation of the codec can be easily
  represented at the signalling layer.

Work processes and technical requirements for achieving the foregoing
objectives are outlined in draft-valin-codec-guidelines and
draft-valin-codec-requirements respectively; these documents will form
the starting point for working toward consensus and will be refined by
the group in accordance with the usual IETF procedures.

The group would prefer to produce a codec that can be widely
implemented and easily distributed among application developers,
service operators, and end users.  This is important because many
existing codecs that might fulfill some or most of the technical
attributes listed above are encumbered in various ways.  For example,
patent holders might require that those wishing to implement the codec
in software, deploy the codec in a service, or distribute the codec
in software or hardware need to request a license, enter into a
business agreement, pay licensing fees or royalties, or adhere to
other special conditions or restrictions.

Because such encumbrances have made it difficult to widely implement
and easily distribute high-quality audio codecs across the entire
Internet community, the group shall prefer unencumbered technologies in
a way that is consistent with BCP 78 and BCP 79.  In particular, the
group shall heed the preference stated in BCP 79: "In general, IETF
working groups prefer technologies with no known IPR claims or, for
technologies with claims against them, an offer of royalty-free
licensing."  Although this preference cannot guarantee that the group
will produce an unencumbered codec, the group shall attempt to adhere
to the spirit of BCP 79.  This preference does not explicitly rule out
the possibility of adapting encumbered technologies; such decisions
will be made in accordance with the rough consensus of the group.

Deliverables

1. Guidelines that define the work process of the group. This
document shall be Informational.

2. Detailed technical requirements. This document shall be
Informational.

3. Specification of one or more codecs that meet the agreed-upon
requirements, in the form of Internet-Drafts that define the codec
algorithm and source code for a reference implementation, as well as
any specific technical requirements for which the codec has been
optimized. The text description of each codec shall indicate which
components of the encoder and decoder are mandatory, recommended, and
optional.  It is envisioned that each such specification shall be a
Proposed Standard document.

Milestones

Mar-2010: WGLC on Codec Standardization Guidelines
May-2010: Codec Standardization Guidelines to IESG (Informational)
May-2010: WGLC on Requirements
Jul-2010: Requirements to IESG (Infomational)
Mar-2011: WGLC on codec specification(s)
Jun-2011: Submit codec specification(s) to IESG (Standards Track)

###

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Date: Thu, 29 Oct 2009 18:49:43 -0700
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Chairs,
I believe that adding a few items to the BoF page at 
<http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76>http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76 
would help tremendously for those looking for information on the 
upcoming CODEC BoF. Namely:

  - Add a paragraph of description. See as an example the KMART/KARP 
BoF listed just below the CODEC BoF
  - add links to the SILK and CELT drafts, and any other helpful 
reading material, be they drafts or not, that would show the work 
being done, for those not following the mail list
  - add a link to the mail list archive message from Cullen and me 
which summarized the first CODEC BoF at IETF75 - 
<http://www.ietf.org/mail-archive/web/codec/current/msg00540.html>http://www.ietf.org/mail-archive/web/codec/current/msg00540.html 


I think that would help new-comers a lot, particularly knowing what 
already occurred, so they know where we will be picking up in Hiroshima.

Hope it helps,
Gregory.

+++++++++++++++++++++++
IETF-related email from
Gregory M. Lebovitz
Juniper Networks
g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m 
--=====================_253476812==.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<body>
Chairs,<br>
I believe that adding a few items to the BoF page at
<a href="http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76">
http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76</a>&nbsp; would help
tremendously for those looking for information on the upcoming CODEC BoF.
Namely:<br><br>
&nbsp;- Add a paragraph of description. See as an example the KMART/KARP
BoF listed just below the CODEC BoF<br>
&nbsp;- add links to the SILK and CELT drafts, and any other helpful
reading material, be they drafts or not, that would show the work being
done, for those not following the mail list<br>
&nbsp;- add a link to the mail list archive message from Cullen and me
which summarized the first CODEC BoF at IETF75 -
<a href="http://www.ietf.org/mail-archive/web/codec/current/msg00540.html">
http://www.ietf.org/mail-archive/web/codec/current/msg00540.html</a>
<br><br>
I think that would help new-comers a lot, particularly knowing what
already occurred, so they know where we will be picking up in
Hiroshima.<br><br>
Hope it helps,<br>
Gregory.<br>
<x-sigsep><p></x-sigsep>
+++++++++++++++++++++++<br>
IETF-related email from<br>
Gregory M. Lebovitz<br>
Juniper Networks<br>
g r e go r&nbsp; y d o t&nbsp; i e tf a t&nbsp; g m a i l&nbsp; do t c
o&nbsp; m</body>
</html>

--=====================_253476812==.ALT--


From stpeter@stpeter.im  Thu Oct 29 20:31:02 2009
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On 10/29/09 7:49 PM, Gregory M. Lebovitz wrote:
> Chairs,
> I believe that adding a few items to the BoF page at
> http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76  would help
> tremendously for those looking for information on the upcoming CODEC
> BoF. Namely:
> 
>  - Add a paragraph of description. See as an example the KMART/KARP BoF
> listed just below the CODEC BoF
>  - add links to the SILK and CELT drafts, and any other helpful reading
> material, be they drafts or not, that would show the work being done,
> for those not following the mail list
>  - add a link to the mail list archive message from Cullen and me which
> summarized the first CODEC BoF at IETF75 -
> http://www.ietf.org/mail-archive/web/codec/current/msg00540.html
> 
> I think that would help new-comers a lot, particularly knowing what
> already occurred, so they know where we will be picking up in Hiroshima.

Those are good points. I'll update the page tomorrow.

Peter

- --
Peter Saint-Andre
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From Borilin@spiritdsp.com  Fri Oct 30 03:41:36 2009
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From: "Slava Borilin" <Borilin@spiritdsp.com>
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Please add link to IPMR draft as well - =
http://tools.ietf.org/search/draft-spiritdsp-ipmr-00 =9A

=20

best regards,

Slava Borilin

________________________________

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf =
Of Gregory M. Lebovitz
Sent: Friday, October 30, 2009 4:50 AM
To: Peter Saint-Andre; Eric Burger
Cc: codec@ietf.org
Subject: [codec] Items to add to BoF page

=20

Chairs,
I believe that adding a few items to the BoF page at =
http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76  would help =
tremendously for those looking for information on the upcoming CODEC =
BoF. Namely:

 - Add a paragraph of description. See as an example the KMART/KARP BoF =
listed just below the CODEC BoF
 - add links to the SILK and CELT drafts, and any other helpful reading =
material, be they drafts or not, that would show the work being done, =
for those not following the mail list
 - add a link to the mail list archive message from Cullen and me which =
summarized the first CODEC BoF at IETF75 - =
http://www.ietf.org/mail-archive/web/codec/current/msg00540.html=20

I think that would help new-comers a lot, particularly knowing what =
already occurred, so they know where we will be picking up in Hiroshima.

Hope it helps,
Gregory.



+++++++++++++++++++++++
IETF-related email from
Gregory M. Lebovitz
Juniper Networks
g r e go r  y d o t  i e tf a t  g m a i l  do t c o  m


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Please add link =
to IPMR
draft as well - <a =
href=3D"http://tools.ietf.org/search/draft-spiritdsp-ipmr-00">http://tool=
s.ietf.org/search/draft-spiritdsp-ipmr-00</a>
=9A<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p>=
</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>best =
regards,</span></font><font
color=3Dnavy><span lang=3DEN-US =
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Slava =
Borilin</span></font><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Tahoma'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Gregory M. =
Lebovitz<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Friday, October 30, =
2009
4:50 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Peter Saint-Andre; =
Eric Burger<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> codec@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [codec] Items to =
add to
BoF page</span></font><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Chairs,<br>
I believe that adding a few items to the BoF page at <a
href=3D"http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76">http://trac.t=
ools.ietf.org/bof/trac/wiki/BifIETF76</a>&nbsp;
would help tremendously for those looking for information on the =
upcoming CODEC
BoF. Namely:<br>
<br>
&nbsp;- Add a paragraph of description. See as an example the KMART/KARP =
BoF
listed just below the CODEC BoF<br>
&nbsp;- add links to the SILK and CELT drafts, and any other helpful =
reading
material, be they drafts or not, that would show the work being done, =
for those
not following the mail list<br>
&nbsp;- add a link to the mail list archive message from Cullen and me =
which
summarized the first CODEC BoF at IETF75 - <a
href=3D"http://www.ietf.org/mail-archive/web/codec/current/msg00540.html"=
>http://www.ietf.org/mail-archive/web/codec/current/msg00540.html</a>
<br>
<br>
I think that would help new-comers a lot, particularly knowing what =
already
occurred, so they know where we will be picking up in Hiroshima.<br>
<br>
Hope it helps,<br>
Gregory.<br>
<br>
<o:p></o:p></span></font></p>

<x-sigsep>

<p></x-sigsep><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>+++++++++++++++++++++++<br>
IETF-related email from<br>
Gregory M. Lebovitz<br>
Juniper Networks<br>
g r e go r&nbsp; y d o t&nbsp; i e tf a t&nbsp; g m a i l&nbsp; do t c =
o&nbsp;
m<o:p></o:p></span></font></p>

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On 10/30/09 4:41 AM, Slava Borilin wrote:
> Please add link to IPMR draft as well -
> http://tools.ietf.org/search/draft-spiritdsp-ipmr-00  

Will do.

Peter

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On 10/29/09 7:49 PM, Gregory M. Lebovitz wrote:
> Chairs,
> I believe that adding a few items to the BoF page at
> http://trac.tools.ietf.org/bof/trac/wiki/BifIETF76  would help
> tremendously for those looking for information on the upcoming CODEC
> BoF. Namely:
> 
>  - Add a paragraph of description. See as an example the KMART/KARP BoF
> listed just below the CODEC BoF
>  - add links to the SILK and CELT drafts, and any other helpful reading
> material, be they drafts or not, that would show the work being done,
> for those not following the mail list
>  - add a link to the mail list archive message from Cullen and me which
> summarized the first CODEC BoF at IETF75 -
> http://www.ietf.org/mail-archive/web/codec/current/msg00540.html
> 
> I think that would help new-comers a lot, particularly knowing what
> already occurred, so they know where we will be picking up in Hiroshima.

Done.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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Subject: [codec] *draft* agenda
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Eric Burger and I have created a *draft* agenda for the Codec BoF.
Feedback is welcome!

###

1. Note Well & Agenda Bashing (5 minutes)

2. Progress to Date (15 minutes)
   a. BoF at IETF 75 established constituency and interest
   b. Open issues: work process, requirements, ITU-T coordination
   c. Several individual drafts published and discussed on the list
   d. Coordination has occurred with ITU-T SG 16
   e. Consensus converging around draft charter

3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)
   a. Informational presentation about ITU-T Study Group 16
   b. Informational presentation about IETF Codec WG
   c. Discussion about alternatives
   d. Determination of consensus

4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 minutes)
   a. Problem statement
   b. Objectives
   c. Deliverables
   d. Milestones

5. Key WG Formation Hums (10 minutes)
   a. Is IETF the best place to do this work?
   b. If a WG is formed, will you write drafts?
   c. If a WG is formed, will you review drafts?
   d. If a WG is formed, will you otherwise participate?


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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
Date: Fri, 30 Oct 2009 11:11:47 -0700
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This looks like a very reasonable agenda, except the humming process elimin=
ates voting for those who cannot travel to the meeting.

Is there some better mechanism possible? Online voting by using a suitable =
conferencing package?
Or counting the votes from the Jabber IM  online participants?
If not, I suggest NOT use the "hum consensus" but some other criteria, such=
 as having a credible team to produce the planned deliverables.

Incidentally, I believe having the IETF process dominated by folks who have=
 a travel budget to be a generic problem, since it eliminates many creative=
 individual experts to the detriment of those companies who (still) can aff=
ord the travel budget. From this perspective, humming is just a broken proc=
ess.

My personal two cents (I plan to attend).
Thanks, Henry



On 10/30/09 12:43 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Eric Burger and I have created a *draft* agenda for the Codec BoF.
Feedback is welcome!

###

1. Note Well & Agenda Bashing (5 minutes)

2. Progress to Date (15 minutes)
   a. BoF at IETF 75 established constituency and interest
   b. Open issues: work process, requirements, ITU-T coordination
   c. Several individual drafts published and discussed on the list
   d. Coordination has occurred with ITU-T SG 16
   e. Consensus converging around draft charter

3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)
   a. Informational presentation about ITU-T Study Group 16
   b. Informational presentation about IETF Codec WG
   c. Discussion about alternatives
   d. Determination of consensus

4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 minutes)
   a. Problem statement
   b. Objectives
   c. Deliverables
   d. Milestones

5. Key WG Formation Hums (10 minutes)
   a. Is IETF the best place to do this work?
   b. If a WG is formed, will you write drafts?
   c. If a WG is formed, will you review drafts?
   d. If a WG is formed, will you otherwise participate?


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_______________________________________________
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<HTML>
<HEAD>
<TITLE>Re: [codec] *draft* agenda</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
13pt'>This looks like a very reasonable agenda, except the humming process =
eliminates voting for those who cannot travel to the meeting.<BR>
<BR>
Is there some better mechanism possible? Online voting by using a suitable =
conferencing package?<BR>
Or counting the votes from the Jabber IM &nbsp;online participants?<BR>
If not, I suggest NOT use the &#8220;hum consensus&#8221; but some other cr=
iteria, such as having a credible team to produce the planned deliverables.=
<BR>
<BR>
Incidentally, I believe having the IETF process dominated by folks who have=
 a travel budget to be a generic problem, since it eliminates many creative=
 individual experts to the detriment of those companies who (still) can aff=
ord the travel budget. From this perspective, humming is just a broken proc=
ess.<BR>
<BR>
My personal two cents (I plan to attend).<BR>
Thanks, Henry<BR>
<BR>
<BR>
<BR>
On 10/30/09 12:43 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@=
stpeter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:13pt'>-----BEGIN PGP SIGNED MESSAGE-----<BR>
Hash: SHA1<BR>
<BR>
Eric Burger and I have created a *draft* agenda for the Codec BoF.<BR>
Feedback is welcome!<BR>
<BR>
###<BR>
<BR>
1. Note Well &amp; Agenda Bashing (5 minutes)<BR>
<BR>
2. Progress to Date (15 minutes)<BR>
&nbsp;&nbsp;&nbsp;a. BoF at IETF 75 established constituency and interest<B=
R>
&nbsp;&nbsp;&nbsp;b. Open issues: work process, requirements, ITU-T coordin=
ation<BR>
&nbsp;&nbsp;&nbsp;c. Several individual drafts published and discussed on t=
he list<BR>
&nbsp;&nbsp;&nbsp;d. Coordination has occurred with ITU-T SG 16<BR>
&nbsp;&nbsp;&nbsp;e. Consensus converging around draft charter<BR>
<BR>
3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)<BR>
&nbsp;&nbsp;&nbsp;a. Informational presentation about ITU-T Study Group 16<=
BR>
&nbsp;&nbsp;&nbsp;b. Informational presentation about IETF Codec WG<BR>
&nbsp;&nbsp;&nbsp;c. Discussion about alternatives<BR>
&nbsp;&nbsp;&nbsp;d. Determination of consensus<BR>
<BR>
4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 minutes)<BR>
&nbsp;&nbsp;&nbsp;a. Problem statement<BR>
&nbsp;&nbsp;&nbsp;b. Objectives<BR>
&nbsp;&nbsp;&nbsp;c. Deliverables<BR>
&nbsp;&nbsp;&nbsp;d. Milestones<BR>
<BR>
5. Key WG Formation Hums (10 minutes)<BR>
&nbsp;&nbsp;&nbsp;a. Is IETF the best place to do this work?<BR>
&nbsp;&nbsp;&nbsp;b. If a WG is formed, will you write drafts?<BR>
&nbsp;&nbsp;&nbsp;c. If a WG is formed, will you review drafts?<BR>
&nbsp;&nbsp;&nbsp;d. If a WG is formed, will you otherwise participate?<BR>
<BR>
<BR>
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_______________________________________________<BR>
codec mailing list<BR>
<a href=3D"codec@ietf.org">codec@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.or=
g/mailman/listinfo/codec</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
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From stpeter@stpeter.im  Fri Oct 30 11:44:48 2009
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Subject: Re: [codec] *draft* agenda
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Hi Henry, comments inline.

On 10/30/09 12:11 PM, Henry Sinnreich wrote:
> This looks like a very reasonable agenda, except the humming process
> eliminates voting for those who cannot travel to the meeting.
> 
> Is there some better mechanism possible? Online voting by using a
> suitable conferencing package?
> Or counting the votes from the Jabber IM  online participants?

I think that method has been used in the past, yes.

> If not, I suggest NOT use the “hum consensus” but some other criteria,
> such as having a credible team to produce the planned deliverables.

Well, hums are traditional.

That said, I do agree with something Adam Roach said at the BoF in
Stockholm. Quoting from the minutes...

###

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.

###

In Stockholm we established that we have people who want to do the work,
and who seem to be capable of doing the work.

Coming out of the Stockholm meeting, we also had several open issues,
which Cullen Jennings mentioned in his summary of the BoF at IETF 75
<http://www.ietf.org/mail-archive/web/codec/current/msg00540.html>:

1. The scope and processes for a Codec WG were not clear. The work on
draft-valin-codec-guidelines has been meant to address those concerns,
and we've had a fairly comprehensive discussion about them.

2. The technical and business requirements were not clear. The technical
requirements have been covered by draft-valin-codec-requirements as
discussed on this list, and the business requirements (mainly IPR to
help ensure that the resulting codec(s) can be widely implemented and
easily distributed) are covered by draft-valin-codec-guidelines.

I think those open issues have been addressed about as completely as
possible to this point, as reflected in the I-Ds mentioned above, in
list discussions, and in the draft charter. Naturally if a WG is formed
then further discussion would happen within the group to reach consensus
on which I-Ds to accept as WG items, on exact requirements, etc.

An additional open issue was coordination with other SDOs in accordance
with liaison agreements in place. I know that discussions have occurred
with ITU-T regarding Study Group 16, but it is not my place to talk
about those conversations because I was not directly involved. I will
leave that for others to talk about, either on this list or at the BoF.
As far as I know, discussions have not occurred with other SDOs.

In any case, the foregoing text provides a bit of what I see as the
context for the BoF in Hiroshima. Other people might disagree, but I
think we've made good progress since Stockholm but need to reach
consensus on a few more topics as input to the IESG's deliberations
about whether it is appropriate to form a WG.

> Incidentally, I believe having the IETF process dominated by folks who
> have a travel budget to be a generic problem, since it eliminates many
> creative individual experts to the detriment of those companies who
> (still) can afford the travel budget. From this perspective, humming is
> just a broken process.

Let's take that up on ietf@ietf.org. :)

/psa
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From ron.even.tlv@gmail.com  Fri Oct 30 11:49:05 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Henry Sinnreich'" <hsinnrei@adobe.com>, "'Peter Saint-Andre'" <stpeter@stpeter.im>, <codec@ietf.org>
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Henry,

I see travel budget and willingness to travel as part of the commitment to
participate in the work a working group. 

 

Roni Even

 

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
Henry Sinnreich
Sent: Friday, October 30, 2009 8:12 PM
To: Peter Saint-Andre; codec@ietf.org
Subject: Re: [codec] *draft* agenda

 

This looks like a very reasonable agenda, except the humming process
eliminates voting for those who cannot travel to the meeting.

Is there some better mechanism possible? Online voting by using a suitable
conferencing package?
Or counting the votes from the Jabber IM  online participants?
If not, I suggest NOT use the "hum consensus" but some other criteria, such
as having a credible team to produce the planned deliverables.

Incidentally, I believe having the IETF process dominated by folks who have
a travel budget to be a generic problem, since it eliminates many creative
individual experts to the detriment of those companies who (still) can
afford the travel budget. From this perspective, humming is just a broken
process.

My personal two cents (I plan to attend).
Thanks, Henry



On 10/30/09 12:43 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Eric Burger and I have created a *draft* agenda for the Codec BoF.
Feedback is welcome!

###

1. Note Well & Agenda Bashing (5 minutes)

2. Progress to Date (15 minutes)
   a. BoF at IETF 75 established constituency and interest
   b. Open issues: work process, requirements, ITU-T coordination
   c. Several individual drafts published and discussed on the list
   d. Coordination has occurred with ITU-T SG 16
   e. Consensus converging around draft charter

3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)
   a. Informational presentation about ITU-T Study Group 16
   b. Informational presentation about IETF Codec WG
   c. Discussion about alternatives
   d. Determination of consensus

4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 minutes)
   a. Problem statement
   b. Objectives
   c. Deliverables
   d. Milestones

5. Key WG Formation Hums (10 minutes)
   a. Is IETF the best place to do this work?
   b. If a WG is formed, will you write drafts?
   c. If a WG is formed, will you review drafts?
   d. If a WG is formed, will you otherwise participate?


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


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<p class=3DMsoNormal><span =
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color:#1F497D'>Henry,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I see travel budget and willingness to travel as part of =
the commitment
to participate in the work a working group. <o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
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color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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<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>Henry
Sinnreich<br>
<b>Sent:</b> Friday, October 30, 2009 8:12 PM<br>
<b>To:</b> Peter Saint-Andre; codec@ietf.org<br>
<b>Subject:</b> Re: [codec] *draft* agenda<o:p></o:p></span></p>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:13.0pt;
font-family:"Calibri","sans-serif"'>This looks like a very reasonable =
agenda,
except the humming process eliminates voting for those who cannot travel =
to the
meeting.<br>
<br>
Is there some better mechanism possible? Online voting by using a =
suitable
conferencing package?<br>
Or counting the votes from the Jabber IM &nbsp;online participants?<br>
If not, I suggest NOT use the &#8220;hum consensus&#8221; but some other =
criteria, such as
having a credible team to produce the planned deliverables.<br>
<br>
Incidentally, I believe having the IETF process dominated by folks who =
have a
travel budget to be a generic problem, since it eliminates many creative
individual experts to the detriment of those companies who (still) can =
afford
the travel budget. From this perspective, humming is just a broken =
process.<br>
<br>
My personal two cents (I plan to attend).<br>
Thanks, Henry<br>
<br>
<br>
<br>
On 10/30/09 12:43 PM, &quot;Peter Saint-Andre&quot; &lt;<a
href=3D"stpeter@stpeter.im">stpeter@stpeter.im</a>&gt; =
wrote:</span><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:13.0pt;
font-family:"Calibri","sans-serif"'>-----BEGIN PGP SIGNED =
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<br>
Eric Burger and I have created a *draft* agenda for the Codec BoF.<br>
Feedback is welcome!<br>
<br>
###<br>
<br>
1. Note Well &amp; Agenda Bashing (5 minutes)<br>
<br>
2. Progress to Date (15 minutes)<br>
&nbsp;&nbsp;&nbsp;a. BoF at IETF 75 established constituency and =
interest<br>
&nbsp;&nbsp;&nbsp;b. Open issues: work process, requirements, ITU-T
coordination<br>
&nbsp;&nbsp;&nbsp;c. Several individual drafts published and discussed =
on the
list<br>
&nbsp;&nbsp;&nbsp;d. Coordination has occurred with ITU-T SG 16<br>
&nbsp;&nbsp;&nbsp;e. Consensus converging around draft charter<br>
<br>
3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)<br>
&nbsp;&nbsp;&nbsp;a. Informational presentation about ITU-T Study Group =
16<br>
&nbsp;&nbsp;&nbsp;b. Informational presentation about IETF Codec WG<br>
&nbsp;&nbsp;&nbsp;c. Discussion about alternatives<br>
&nbsp;&nbsp;&nbsp;d. Determination of consensus<br>
<br>
4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 =
minutes)<br>
&nbsp;&nbsp;&nbsp;a. Problem statement<br>
&nbsp;&nbsp;&nbsp;b. Objectives<br>
&nbsp;&nbsp;&nbsp;c. Deliverables<br>
&nbsp;&nbsp;&nbsp;d. Milestones<br>
<br>
5. Key WG Formation Hums (10 minutes)<br>
&nbsp;&nbsp;&nbsp;a. Is IETF the best place to do this work?<br>
&nbsp;&nbsp;&nbsp;b. If a WG is formed, will you write drafts?<br>
&nbsp;&nbsp;&nbsp;c. If a WG is formed, will you review drafts?<br>
&nbsp;&nbsp;&nbsp;d. If a WG is formed, will you otherwise =
participate?<br>
<br>
<br>
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_______________________________________________<br>
codec mailing list<br>
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<a =
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From petithug@acm.org  Fri Oct 30 13:01:16 2009
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From: Marc Petit-Huguenin <petithug@acm.org>
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To: Roni Even <ron.even.tlv@gmail.com>
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Cc: codec@ietf.org, 'Drew Dvorshak' <Dvorshak@ISOC.org>
Subject: [codec] Sponsorship for FOSS developers [was Re:  *draft* agenda]
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Roni Even wrote:
> 
> 
> Henry,
> 
> I see travel budget and willingness to travel as part of the commitment
> to participate in the work a working group.
> 

I will not comment on this, but I wonder if ISOC could not have a Sponsorship
program for FOSS developers, similar to the existing sponsorship.  After all,
FOSS developers are the one writing the really good code for the Internet this
days, and the less likely to be able to secure a travel budget to attend the
IETF meetings.

>  
> 
> Roni Even
> 
>  
> 
> *From:* codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] *On
> Behalf Of *Henry Sinnreich
> *Sent:* Friday, October 30, 2009 8:12 PM
> *To:* Peter Saint-Andre; codec@ietf.org
> *Subject:* Re: [codec] *draft* agenda
> 
>  
> 
> This looks like a very reasonable agenda, except the humming process
> eliminates voting for those who cannot travel to the meeting.
> 
> Is there some better mechanism possible? Online voting by using a
> suitable conferencing package?
> Or counting the votes from the Jabber IM  online participants?
> If not, I suggest NOT use the “hum consensus” but some other criteria,
> such as having a credible team to produce the planned deliverables.
> 
> Incidentally, I believe having the IETF process dominated by folks who
> have a travel budget to be a generic problem, since it eliminates many
> creative individual experts to the detriment of those companies who
> (still) can afford the travel budget. From this perspective, humming is
> just a broken process.
> 
> My personal two cents (I plan to attend).
> Thanks, Henry

-- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org

From stpeter@stpeter.im  Fri Oct 30 13:11:31 2009
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Subject: [codec] questions to be answered
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RFC 5434 has a list of questions that need to be answered before a WG is
formed, and I think we have fairly clear answers to these questions:

1. Is there a problem that needs solving?

(Yes, see problem statement in charter.)

2. Is the scope of the problem well defined and understood?

(Yes, see objectives in charter, guidelines I-D, and requirements I-D.)

3. Is there agreement on what the WG needs to deliver?

(Yes, see deliverables in charter, guidelines I-D, and requirements I-D.)

4. Are there enough IETF participants to do the work?

(Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)

5. Does the WG have a reasonable probability of success?

(Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
emphasis on "reasonable" because we won't know until we try.)

6. Is the IETF is the right place to do the work?

(Maybe. I think this is the major item to clarify at the BoF.)

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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Cc: codec@ietf.org, IETF discussion list <ietf@ietf.org>, 'Drew Dvorshak' <Dvorshak@ISOC.org>
Subject: Re: [codec] Sponsorship for FOSS developers (was: Re: *draft* agenda)
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On 10/30/09 2:01 PM, Marc Petit-Huguenin wrote:
> Roni Even wrote:
>>
>> Henry,
>>
>> I see travel budget and willingness to travel as part of the commitment
>> to participate in the work a working group.
>>
> 
> I will not comment on this, but I wonder if ISOC could not have a Sponsorship
> program for FOSS developers, similar to the existing sponsorship.  After all,
> FOSS developers are the one writing the really good code for the Internet this
> days, and the less likely to be able to secure a travel budget to attend the
> IETF meetings.

That's a great idea. It might help us stay closer to running code, too.

Peter

- --
Peter Saint-Andre
https://stpeter.im/


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Peter Saint-Andre'" <stpeter@stpeter.im>, <codec@ietf.org>
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Subject: Re: [codec] questions to be answered
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Hi Peter,
I am not sure that there is a consensus about what the group need to
deliver. I do not feel that the issue of the number of codecs and how to
decide which codecs are in scope for this working group is agreed. My view
that this issue is not addressed since I assume that all the parties that
submitted candidate codecs would like their codecs to be approved at the
IETF which will make this group a rubber stamping WG.
I think that the if the charter describes a list of applications it need to
have a separate requirement for each application on which this proposed
group is going to develop a separate codec. If there is one requirement
document for all apllications there should be one codec to address it.

Thanks
Roni Even



> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Peter Saint-Andre
> Sent: Friday, October 30, 2009 10:12 PM
> To: codec@ietf.org
> Subject: [codec] questions to be answered
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> RFC 5434 has a list of questions that need to be answered before a WG
> is
> formed, and I think we have fairly clear answers to these questions:
> 
> 1. Is there a problem that needs solving?
> 
> (Yes, see problem statement in charter.)
> 
> 2. Is the scope of the problem well defined and understood?
> 
> (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
> 
> 3. Is there agreement on what the WG needs to deliver?
> 
> (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> D.)
> 
> 4. Are there enough IETF participants to do the work?
> 
> (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
> 
> 5. Does the WG have a reasonable probability of success?
> 
> (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> emphasis on "reasonable" because we won't know until we try.)
> 
> 6. Is the IETF is the right place to do the work?
> 
> (Maybe. I think this is the major item to clarify at the BoF.)
> 
> Peter
> 
> - --
> Peter Saint-Andre
> https://stpeter.im/
> 
> 
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From mknappe@juniper.net  Fri Oct 30 16:16:32 2009
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From: Michael Knappe <mknappe@juniper.net>
To: Roni Even <ron.even.tlv@gmail.com>, Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
Date: Fri, 30 Oct 2009 16:15:48 -0700
Thread-Topic: [codec] questions to be answered
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Subject: Re: [codec] questions to be answered
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Roni,

On your latter points, how about a set of general benchmark requirements th=
at all codecs must meet, followed by more stringent or more specific criter=
ia to meet narrower requirements of a target application? This could be all=
 coupled together in a single requirements document. A single doc would mak=
e it easier to lay out common test methodologies just once, followed by spe=
cific application requirements broken out into separate subsequent sections=
.

Mike


On 10/30/09 3:34 PM, "Roni Even" <ron.even.tlv@gmail.com> wrote:

Hi Peter,
I am not sure that there is a consensus about what the group need to
deliver. I do not feel that the issue of the number of codecs and how to
decide which codecs are in scope for this working group is agreed. My view
that this issue is not addressed since I assume that all the parties that
submitted candidate codecs would like their codecs to be approved at the
IETF which will make this group a rubber stamping WG.
I think that the if the charter describes a list of applications it need to
have a separate requirement for each application on which this proposed
group is going to develop a separate codec. If there is one requirement
document for all apllications there should be one codec to address it.

Thanks
Roni Even



> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Peter Saint-Andre
> Sent: Friday, October 30, 2009 10:12 PM
> To: codec@ietf.org
> Subject: [codec] questions to be answered
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> RFC 5434 has a list of questions that need to be answered before a WG
> is
> formed, and I think we have fairly clear answers to these questions:
>
> 1. Is there a problem that needs solving?
>
> (Yes, see problem statement in charter.)
>
> 2. Is the scope of the problem well defined and understood?
>
> (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
>
> 3. Is there agreement on what the WG needs to deliver?
>
> (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> D.)
>
> 4. Are there enough IETF participants to do the work?
>
> (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
>
> 5. Does the WG have a reasonable probability of success?
>
> (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> emphasis on "reasonable" because we won't know until we try.)
>
> 6. Is the IETF is the right place to do the work?
>
> (Maybe. I think this is the major item to clarify at the BoF.)
>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.8 (Darwin)
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> =3D9vPK
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> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec

_______________________________________________
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https://www.ietf.org/mailman/listinfo/codec


From mknappe@juniper.net  Fri Oct 30 16:32:23 2009
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From: Michael Knappe <mknappe@juniper.net>
To: Michael Knappe <mknappe@juniper.net>, Roni Even <ron.even.tlv@gmail.com>,  Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
Date: Fri, 30 Oct 2009 16:31:38 -0700
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Also, I would suggest before Hiroshima we take a stab on email at writing o=
ut a short list of both critical and non-critical technical criteria for ea=
ch application (gaming, interactive music, voice communication, etc). By go=
ing through this exercise it will help us all mentally group requirements a=
nd should help simplify the overall problem definition (e.g. a) Which appli=
cation's requirements are supersets of another, b) which application's requ=
irements would have fundamental technical tradeoffs vs another, etc.). If w=
e can narrow down the problem to a couple of fundamental codec 'description=
s', that will be a big help, and if a one-size-fits-all codec is at all pos=
sible, it should become easier to recognize.

Mike




On 10/30/09 4:15 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

Roni,

On your latter points, how about a set of general benchmark requirements th=
at all codecs must meet, followed by more stringent or more specific criter=
ia to meet narrower requirements of a target application? This could be all=
 coupled together in a single requirements document. A single doc would mak=
e it easier to lay out common test methodologies just once, followed by spe=
cific application requirements broken out into separate subsequent sections=
.

Mike


On 10/30/09 3:34 PM, "Roni Even" <ron.even.tlv@gmail.com> wrote:

Hi Peter,
I am not sure that there is a consensus about what the group need to
deliver. I do not feel that the issue of the number of codecs and how to
decide which codecs are in scope for this working group is agreed. My view
that this issue is not addressed since I assume that all the parties that
submitted candidate codecs would like their codecs to be approved at the
IETF which will make this group a rubber stamping WG.
I think that the if the charter describes a list of applications it need to
have a separate requirement for each application on which this proposed
group is going to develop a separate codec. If there is one requirement
document for all apllications there should be one codec to address it.

Thanks
Roni Even



> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Peter Saint-Andre
> Sent: Friday, October 30, 2009 10:12 PM
> To: codec@ietf.org
> Subject: [codec] questions to be answered
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> RFC 5434 has a list of questions that need to be answered before a WG
> is
> formed, and I think we have fairly clear answers to these questions:
>
> 1. Is there a problem that needs solving?
>
> (Yes, see problem statement in charter.)
>
> 2. Is the scope of the problem well defined and understood?
>
> (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
>
> 3. Is there agreement on what the WG needs to deliver?
>
> (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> D.)
>
> 4. Are there enough IETF participants to do the work?
>
> (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
>
> 5. Does the WG have a reasonable probability of success?
>
> (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> emphasis on "reasonable" because we won't know until we try.)
>
> 6. Is the IETF is the right place to do the work?
>
> (Maybe. I think this is the major item to clarify at the BoF.)
>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
>
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> _______________________________________________
> 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

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


From jean-marc.valin@usherbrooke.ca  Fri Oct 30 17:42:34 2009
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Date: Fri, 30 Oct 2009 20:42:36 -0400
From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
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Roni,

I don't believe anyone here intends the WG to produce more than one
codec unless it turns out that doing so is the only way to achieve good
coverage of the requirements. Would you like to see such a statement in
the charter? Personally, I tend to see this question mostly as a
technical issue that will be settled once the technical work gets
started. Believe me, there's nothing I would like more than being able
to address all the requirements we have with a single, unified codec. I
also believe that's what you would also like so see, so we agree on
this. Now technically, I am not yet sure that it is feasible to have a
single codec that does everything we want to do. So I believe it is a
possibility that we would end up with one codec (A) addressing one part
of the requirements and another codec (B) addressing a second part of
the requirements. Sure, we could decide to take those two codecs, put
them in the same proposal and call that one big codec with two
"profiles", but I don't think we would be gaining anything. At this
point in time, saying that we will have a single codec would be assuming
the output of the technical work before it has begun.

It is the same issue with separating requirements. I assume that if we
end up going with two codecs (which is not my first choice), we may need
to split the requirements into two parts. However, at this point not
only is it not clear that we will have more than one codec, but it is
not clear where the split would be if there were to be more than one
codec. I have a few ideas on how the split could be done, but I do not
see how we could be choosing that split before actually working on the code.

Cheers,

	Jean-Marc


Roni Even a crit :
> Hi Peter,
> I am not sure that there is a consensus about what the group need to
> deliver. I do not feel that the issue of the number of codecs and how to
> decide which codecs are in scope for this working group is agreed. My view
> that this issue is not addressed since I assume that all the parties that
> submitted candidate codecs would like their codecs to be approved at the
> IETF which will make this group a rubber stamping WG.
> I think that the if the charter describes a list of applications it need to
> have a separate requirement for each application on which this proposed
> group is going to develop a separate codec. If there is one requirement
> document for all apllications there should be one codec to address it.
> 
> Thanks
> Roni Even
> 
> 
> 
>> -----Original Message-----
>> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
>> Of Peter Saint-Andre
>> Sent: Friday, October 30, 2009 10:12 PM
>> To: codec@ietf.org
>> Subject: [codec] questions to be answered
>>
> RFC 5434 has a list of questions that need to be answered before a WG
> is
> formed, and I think we have fairly clear answers to these questions:
> 
> 1. Is there a problem that needs solving?
> 
> (Yes, see problem statement in charter.)
> 
> 2. Is the scope of the problem well defined and understood?
> 
> (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
> 
> 3. Is there agreement on what the WG needs to deliver?
> 
> (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> D.)
> 
> 4. Are there enough IETF participants to do the work?
> 
> (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
> 
> 5. Does the WG have a reasonable probability of success?
> 
> (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> emphasis on "reasonable" because we won't know until we try.)
> 
> 6. Is the IETF is the right place to do the work?
> 
> (Maybe. I think this is the major item to clarify at the BoF.)
> 
> Peter
> 
_______________________________________________
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 gregory.ietf@gmail.com  Sat Oct 31 08:26:07 2009
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Subject: Re: [codec] *draft* agenda
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At 10:43 AM 10/30/2009, Peter Saint-Andre wrote:
>-----BEGIN PGP SIGNED MESSAGE-----
>Hash: SHA1
>
>Eric Burger and I have created a *draft* agenda for the Codec BoF.
>Feedback is welcome!
>
>###
>
>1. Note Well & Agenda Bashing (5 minutes)
>
>2. Progress to Date (15 minutes)
>    a. BoF at IETF 75 established constituency and interest
>    b. Open issues: work process, requirements, ITU-T coordination
>    c. Several individual drafts published and discussed on the list
>    d. Coordination has occurred with ITU-T SG 16

s/has occurred/is occurring

Justification:  We have had a good call between IETF leaders and 
ITU-T SG 16 and other ITU-T leaders to ensure good coordination 
around this topic. I and Cullen are drafting a formal Liaison to the 
ITU-T, which is being reviewed by the IAB and will be sent this 
weekend. But this effort will entail continued and on-going 
coordination with ITU-T, as they have much to offer in this process.

Gregory.

>    e. Consensus converging around draft charter
>
>3. Key Question to Answer: Is IETF the Right Forum? (45 minutes)

Possible restatement:

"Does IETF have a reasonable chance of success compared to other forums?"

Justification:  The term "Right" is overloaded. Right based on what 
criteria? It's not that black and white. But we may or may not have a 
more reasonable chance of success _meeting the stated goals_ compared 
to other forums. That's not right or wrong, its more objective.


>    a. Informational presentation about ITU-T Study Group 16
>    b. Informational presentation about IETF Codec WG
>    c. Discussion about alternatives
>    d. Determination of consensus
>
>4. If IETF Is or Might Be the Right Forum: Charter Bashing (45 minutes)
>    a. Problem statement
>    b. Objectives
>    c. Deliverables
>    d. Milestones
>
>5. Key WG Formation Hums (10 minutes)
>    a. Is IETF the best place to do this work?
>    b. If a WG is formed, will you write drafts?
>    c. If a WG is formed, will you review drafts?

Do we need to add:

"If a WG is formed, will you test and characterize candidate code?"
?

I think we've made a lot of progress on this agenda, and I look 
forward to a constructive and interesting BoF.

Gregory

>    d. If a WG is formed, will you otherwise participate?
>
>
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From gregory.ietf@gmail.com  Sat Oct 31 08:36:46 2009
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To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>, Roni Even <ron.even.tlv@gmail.com>
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Subject: Re: [codec] questions to be answered
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I would be comfortable with chartering for 1 or 2=20
codecs, under the assumption that one-size does=20
not fit all in the world of codecs, and believing=20
that we could define one or two application=20
spaces that cover a very large need and market=20
demand for a application or group of=20
applications. Sure, there are probably a handful=20
or so different applications that in order to=20
cover excellently would require different and=20
specific codec technologies, but I think we need=20
to focus, deploy, evaluate, and then determine next steps here.

I would oppose a charter that tries to do more=20
than one or two applications with corresponding codecs.

I'm most comfortable with one, but could see the=20
need for two, and would be willing to live with a charter that included two.

Gregory


At 05:42 PM 10/30/2009, Jean-Marc Valin wrote:
>Roni,
>
>I don't believe anyone here intends the WG to produce more than one
>codec unless it turns out that doing so is the only way to achieve good
>coverage of the requirements. Would you like to see such a statement in
>the charter? Personally, I tend to see this question mostly as a
>technical issue that will be settled once the technical work gets
>started. Believe me, there's nothing I would like more than being able
>to address all the requirements we have with a single, unified codec. I
>also believe that's what you would also like so see, so we agree on
>this. Now technically, I am not yet sure that it is feasible to have a
>single codec that does everything we want to do. So I believe it is a
>possibility that we would end up with one codec (A) addressing one part
>of the requirements and another codec (B) addressing a second part of
>the requirements. Sure, we could decide to take those two codecs, put
>them in the same proposal and call that one big codec with two
>"profiles", but I don't think we would be gaining anything. At this
>point in time, saying that we will have a single codec would be assuming
>the output of the technical work before it has begun.
>
>It is the same issue with separating requirements. I assume that if we
>end up going with two codecs (which is not my first choice), we may need
>to split the requirements into two parts. However, at this point not
>only is it not clear that we will have more than one codec, but it is
>not clear where the split would be if there were to be more than one
>codec. I have a few ideas on how the split could be done, but I do not
>see how we could be choosing that split before actually working on the=
 code.
>
>Cheers,
>
>         Jean-Marc
>
>
>Roni Even a =E9crit :
> > Hi Peter,
> > I am not sure that there is a consensus about what the group need to
> > deliver. I do not feel that the issue of the number of codecs and how to
> > decide which codecs are in scope for this working group is agreed. My=
 view
> > that this issue is not addressed since I assume that all the parties=
 that
> > submitted candidate codecs would like their codecs to be approved at the
> > IETF which will make this group a rubber stamping WG.
> > I think that the if the charter describes a list of applications it need=
 to
> > have a separate requirement for each application on which this proposed
> > group is going to develop a separate codec. If there is one requirement
> > document for all apllications there should be one codec to address it.
> >
> > Thanks
> > Roni Even
> >
> >
> >
> >> -----Original Message-----
> >> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> >> Of Peter Saint-Andre
> >> Sent: Friday, October 30, 2009 10:12 PM
> >> To: codec@ietf.org
> >> Subject: [codec] questions to be answered
> >>
> > RFC 5434 has a list of questions that need to be answered before a WG
> > is
> > formed, and I think we have fairly clear answers to these questions:
> >
> > 1. Is there a problem that needs solving?
> >
> > (Yes, see problem statement in charter.)
> >
> > 2. Is the scope of the problem well defined and understood?
> >
> > (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
> >
> > 3. Is there agreement on what the WG needs to deliver?
> >
> > (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> > D.)
> >
> > 4. Are there enough IETF participants to do the work?
> >
> > (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
> >
> > 5. Does the WG have a reasonable probability of success?
> >
> > (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> > emphasis on "reasonable" because we won't know until we try.)
> >
> > 6. Is the IETF is the right place to do the work?
> >
> > (Maybe. I think this is the major item to clarify at the BoF.)
> >
> > Peter
> >
>_______________________________________________
>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
>
>
>_______________________________________________
>codec mailing list
>codec@ietf.org
>https://www.ietf.org/mailman/listinfo/codec


From mknappe@juniper.net  Sat Oct 31 08:44:36 2009
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From: Michael Knappe <mknappe@juniper.net>
To: Roni Even <ron.even.tlv@gmail.com>, Peter Saint-Andre <stpeter@stpeter.im>, "codec@ietf.org" <codec@ietf.org>
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I will take a rough first crack at a codec application vs criteria table an=
d have it out to the list for discussion on Monday Nov 2nd.

Cheers,

Mike


On 10/30/09 4:31 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

Also, I would suggest before Hiroshima we take a stab on email at writing o=
ut a short list of both critical and non-critical technical criteria for ea=
ch application (gaming, interactive music, voice communication, etc). By go=
ing through this exercise it will help us all mentally group requirements a=
nd should help simplify the overall problem definition (e.g. a) Which appli=
cation's requirements are supersets of another, b) which application's requ=
irements would have fundamental technical tradeoffs vs another, etc.). If w=
e can narrow down the problem to a couple of fundamental codec 'description=
s', that will be a big help, and if a one-size-fits-all codec is at all pos=
sible, it should become easier to recognize.

Mike




On 10/30/09 4:15 PM, "Michael Knappe" <mknappe@juniper.net> wrote:

Roni,

On your latter points, how about a set of general benchmark requirements th=
at all codecs must meet, followed by more stringent or more specific criter=
ia to meet narrower requirements of a target application? This could be all=
 coupled together in a single requirements document. A single doc would mak=
e it easier to lay out common test methodologies just once, followed by spe=
cific application requirements broken out into separate subsequent sections=
.

Mike


On 10/30/09 3:34 PM, "Roni Even" <ron.even.tlv@gmail.com> wrote:

Hi Peter,
I am not sure that there is a consensus about what the group need to
deliver. I do not feel that the issue of the number of codecs and how to
decide which codecs are in scope for this working group is agreed. My view
that this issue is not addressed since I assume that all the parties that
submitted candidate codecs would like their codecs to be approved at the
IETF which will make this group a rubber stamping WG.
I think that the if the charter describes a list of applications it need to
have a separate requirement for each application on which this proposed
group is going to develop a separate codec. If there is one requirement
document for all apllications there should be one codec to address it.

Thanks
Roni Even



> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Peter Saint-Andre
> Sent: Friday, October 30, 2009 10:12 PM
> To: codec@ietf.org
> Subject: [codec] questions to be answered
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> RFC 5434 has a list of questions that need to be answered before a WG
> is
> formed, and I think we have fairly clear answers to these questions:
>
> 1. Is there a problem that needs solving?
>
> (Yes, see problem statement in charter.)
>
> 2. Is the scope of the problem well defined and understood?
>
> (Yes, see objectives in charter, guidelines I-D, and requirements I-D.)
>
> 3. Is there agreement on what the WG needs to deliver?
>
> (Yes, see deliverables in charter, guidelines I-D, and requirements I-
> D.)
>
> 4. Are there enough IETF participants to do the work?
>
> (Yes, see codec I-Ds, poll at Stockholm BoF, and list discussion.)
>
> 5. Does the WG have a reasonable probability of success?
>
> (Yes, see codec I-Ds, guidelines I-D, and list discussion, with an
> emphasis on "reasonable" because we won't know until we try.)
>
> 6. Is the IETF is the right place to do the work?
>
> (Maybe. I think this is the major item to clarify at the BoF.)
>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
>
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https://www.ietf.org/mailman/listinfo/codec

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