Re: [rtcweb] RTCWeb default signaling protocol [was RE: Aboutdefining a signaling protocol for WebRTC (or not)]

Randell Jesup <randell-ietf@jesup.org> Thu, 22 September 2011 20:05 UTC

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Date: Thu, 22 Sep 2011 16:04:12 -0400
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Subject: Re: [rtcweb] RTCWeb default signaling protocol [was RE: Aboutdefining a signaling protocol for WebRTC (or not)]
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On 9/22/2011 2:22 PM, Ravindran Parthasarathi wrote:

> OK, let's take the game example... only 2-player games would be able to
> use a simple rtcweb-SIP agent.  Anything more than 2-player would want
> to use the multi-party "conferencing" model of rtcweb, which can't even
> be signaled with SIP today as far as I can tell. (not that I've thought
> about it too much, but I can't see how it would without some changes to
> SIP)
>
> <partha>  In fact rtcweb client shall acts as conference un-aware client
> in the conference model and game server acts as conference server. I
> have worked in 3-party conference using SIP in the endpoint but I have
> never seen endpoint acts as conference server. So, I may be missing
> something here</partha>

I've built videophones that do in-phone 3 or 4 way conferencing (bridging
and mixing the audio and video).  It can also be done without mixing by
forwarding.  I'm unclear on if that would be possible in webrtc; is there
any way to loop around or mix audio/video from one PeerConnection to another.
(You can always turn off AEC and loop the audio... ;-)


-- 
Randell Jesup
randell-ietf@jesup.org