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

Iñaki Baz Castillo <ibc@aliax.net> Fri, 30 September 2011 20:54 UTC

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Date: Fri, 30 Sep 2011 22:57:14 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Low Level Javascript API Proposal avail on the webrtc list
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2011/9/30 Tim Panton <tim@phonefromhere.com>om>:
>> 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.

-- 
Iñaki Baz Castillo
<ibc@aliax.net>