Re: [rtcweb] Low Level Javascript API Proposal avail on the webrtc list

Harald Alvestrand <harald@alvestrand.no> Sat, 01 October 2011 07:25 UTC

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Date: Sat, 01 Oct 2011 09:28:36 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: Iñaki Baz Castillo <ibc@aliax.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Low Level Javascript API Proposal avail on the webrtc list
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On 09/30/2011 10:57 PM, Iñaki Baz Castillo wrote:
> 2011/9/30 Tim Panton<tim@phonefromhere.com>:
>>> Not in case you implement pure SIP on top of JavaScript/WebSocket :)
>> So the question is - do you think that you _could_ implement SIP on top of JavaScript/WebSocket
>> and use the RTP/media api as described ? We tried to make them orthogonal, but I think Neil feels that
>> to support forking you may need additional features or flags to the calls described in the proposal.
> I don't know (yet) if a WebRTC client can establish different media
> sessions with different peers at the same time (it seems difficult as
> there is supposed to be a single microphone). If not, then at
> javascript level I can decide which session to enable and which one to
> pause (I hope). Also, if I make an outgoing call from the webbrowser
> and receive multiple 180/183 with different SDP, I could choose which
> one to render so I just establish the media session with one of them.
The scenarios-and-requirements document requires that this be supported.