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

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?

Cheers,

Xavier

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

(c.f. http://tools.ietf.org/agenda/86/slides/slides-86-rtcweb-6.pdf )

On Thu, Mar 7, 2013 at 11:18 AM, Ted Hardie <ted.ietf@gmail.com> 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.
>
> regards,
>
> Ted
>
>
> On Mon, Mar 4, 2013 at 2:31 PM, Espen Berger (espeberg)
> <espeberg@cisco.com> 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
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> _______________________________________________
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> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>