Re: [rtcweb] Codec Draft

Xavier Marjou <xavier.marjou@orange.com> Fri, 04 November 2011 17:26 UTC

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Date: Fri, 4 Nov 2011 18:26:04 +0100
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From: Xavier Marjou <xavier.marjou@orange.com>
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Subject: Re: [rtcweb] Codec Draft
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On Fri, Nov 4, 2011 at 5:31 PM, Iñaki Baz Castillo <ibc@aliax.net> wrote:

> El 04/11/2011 15:20, "Xavier Marjou" <xavier.marjou@orange.com> escribió:
>
> >
> >
> http://tools.ietf.org/html/draft-kaplan-rtcweb-sip-interworking-requirements-00, which
> I fully support by the way.
>
> Xavier, such draft does not propose that Webrtc must implement all the
> requirements in the draft. It just lists all the requirements needed in
> order to fully interoperate with current SIP deployments and opens the door
> for discussion about it.
>
You rephrasing is correct : however the draft does indicate that many
codecs would be desirable from an interworking perspective

So if you "fully support" this draft it means that you are just interested
> in making Webrtc to work with current SIP, regardless security requirements
> in the Web.
>
I am not "just" interested : I am "also" interested in making WebRTC work
with existing networks using SIP and RTP.


So let me know: do you support that browsers must implement g729?
>
yes (for some calls, not for all calls)



> Do you support that webrtc requires not security at all in the media plane
> (like legacy SIP)?
>
out-of-thread : no, of course. However in some circumstances some features
may be done at another level or not needed at all (eg: for RTP vs SRTP)


If so, I dont think you care about Webrtc for the Web, but just for telcos.
> Behaviors like this one makes this WG to seem a telco party rather than a
> WG working for the Web. WebRTC means RTC for the Web, rather than Web for
> telcos, or that is what I hope.
>
> Regards.
>
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