Re: [rtcweb] Codec Draft

Iñaki Baz Castillo <ibc@aliax.net> Fri, 04 November 2011 17:16 UTC

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Date: Fri, 4 Nov 2011 18:16:52 +0100
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: "Bran, Cary" <Cary.Bran@plantronics.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Xavier Marjou <xavier.marjou@orange.com>
Subject: Re: [rtcweb] Codec Draft
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Hi Bran. I meant
http://tools.ietf.org/html/draft-kaplan-rtcweb-sip-interworking-requirements-00
.

That draft describes all the requirements for out of the box
interoperabiliy between current SIP and WebRTC. It does not try to mandate
those requirements, but just to list them. So if somebody "fully supports"
this draft it means that browsers MUST implement g729 and allow not
security at all, which is annoying. I just replied to a comment "fully
supporting" this draft.

Regards.
El 04/11/2011 18:07, "Bran, Cary" <Cary.Bran@plantronics.com> escribió:

>  Iñaki,
>
>
>
> I agree with you with regards to WebRTC needs to embrace a web paradigm
> and I am a little confused how the latest version of our document implies a
> telco paradigm.
>
>
>
> The 01 version can be found here:
> http://tools.ietf.org/id/draft-cbran-rtcweb-codec-01.txt
>
>
>
> Currently the documents lists two video codec candidates (VP8/H.264) , and three audio codec candidates, PCMA/PCMU, Telephone Event and Opus.
>
>
>
> If you have anything codecs to add that would be considered more web centric, please let us know and we can add it to the open issues list.
>
>
>
> Cheers,
>
>
>
> -Cary
>
>
>
> *From:* Iñaki Baz Castillo [mailto:ibc@aliax.net]
> *Sent:* Friday, November 04, 2011 9:31 AM
> *To:* Xavier Marjou
> *Cc:* Bran, Cary; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Codec Draft
>
>
>
> 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.
>
> 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.
>
> So let me know: do you support that browsers must implement g729? Do you
> support that webrtc requires not security at all in the media plane (like
> legacy SIP)?
>
> 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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