Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

"Bogineni, Kalyani" <> Wed, 13 March 2013 14:21 UTC

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From: "Bogineni, Kalyani" <>
To: 'Harald Alvestrand' <>, "" <>
Date: Wed, 13 Mar 2013 10:20:48 -0400
Thread-Topic: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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Section 4.3.1 of  draft-ietf-rtcweb-use-cases-and-requirements-10.txt
has usecases for mobile devices.

Here is the exact wording from the WG charter:

"..The working group cannot explicitly rule out the possibility of adopting encumbered
technologies; however, the working group will try to avoid encumbered
technologies that require royalties or other encumbrances that would
prevent such technologies from being easy to use in web browsers."
So the proposal to include support for AMR/AMR-WB is within the scope of the
current work.

Kalyani Bogineni

From: [] On Behalf Of Harald Alvestrand
Sent: Wednesday, March 13, 2013 10:06 AM
Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

High level thoughts from a browser implementor:

- The first target of WebRTC is the browser to browser case. These codecs do not add any capability to the browser to browser case, because OPUS is available.

- In the most common browser distribution models, it is an advantage to include at least a fallback version of all available features in the binary, so that the set of features available to the user is constant. This means that the browser either incurs licensing costs or support costs (to support the variability of user scenarios).

- The inclusion of royalty-required codecs severely crimps the distribution models available for browsers, which makes it even harder for new browsers to enter the market than it already is, and the WG charter we agreed on says that we "prefer non-encumbered codecs".

We can discuss the cost of transcoding separately, but the cost to the browsers of extra codecs, especially royalty-encumbered ones, should not be underestimated.

On 03/13/2013 02:14 PM, Xavier Marjou wrote:

Here is a summary of the draft-marjou-rtcweb-audio-codecs-for-interop-00 presentation that I had prepared for yesterday's session:

- The co-authors want to underline that non-WebRTC voice endpoints usually use one of the following codecs: AMR, AMR-WB or G.722, which will result in massive transcoding when there will be communications between WebRTC endpoints and non-WebRTC endpoints.

- On one side, transcoding is bad for many reasons discussed in the draft (cost issues, intrinsic quality degradation, degraded interactivity, fallback from HD to G.711...);

- On the other side, it is recognized that implementing additional codecs in the browsers can generate additional costs.

- In order to reach a compromise, we would like to add some text in the WG draft draft-ietf-rtcweb-audio providing incentives for the browser to use these three codecs: make them mandatory to implement when there is no cost impact on the browser (e.g. if codec already installed, paid by the device vendor...).

Any opinion on that?



PS: I will be ready to present the slides on Thursday if time permits it.

(c.f. )

On Thu, Mar 7, 2013 at 11:18 AM, Ted Hardie <<>> wrote:
Magnus and I discussed this this morning, and we encourage you to
prepare something.  If the discussion of working group last call items
runs short, we may be able to fit this in at that time or at the end
of day one if its full agenda his finished.  This is not a commitment,
however, so please try and get discussion on the list on the points
from the draft you feel need resolution.



On Mon, Mar 4, 2013 at 2:31 PM, Espen Berger (espeberg)
<<>> wrote:
> Hello,
> I would like to request agenda time for:
> draft-marjou-rtcweb-audio-codecs-for-interop-01
> The document  presents use-cases underlining why WebRTC needs AMR-WB,  AMR
> and G.722 as additional relevant voice codecs to satisfactorily ensure
> interoperability with existing systems.
> A 10-minute time slot should be sufficient for presentation and discussion.
> Regards
> -Espen
> _______________________________________________
> rtcweb mailing list
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