Re: [rtcweb] Codec Draft

"Bran, Cary" <> Fri, 04 November 2011 17:07 UTC

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From: "Bran, Cary" <>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <>, Xavier Marjou <>
Thread-Topic: [rtcweb] Codec Draft
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Date: Fri, 4 Nov 2011 17:07:23 +0000
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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:

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.



From: Iñaki Baz Castillo []
Sent: Friday, November 04, 2011 9:31 AM
To: Xavier Marjou
Cc: Bran, Cary;
Subject: Re: [rtcweb] Codec Draft

El 04/11/2011 15:20, "Xavier Marjou" <<>> escribió:
>, 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.



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