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Re: [rtcweb] Media forking solution for SIP interoperability (without a media gateway)

Harald Alvestrand <harald@alvestrand.no> Mon, 31 October 2011 01:08 UTC

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Date: Sun, 30 Oct 2011 18:08:33 -0700
From: Harald Alvestrand <harald@alvestrand.no>
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To: Cullen Jennings <fluffy@cisco.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Media forking solution for SIP interoperability (without a media gateway)
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On 10/30/2011 11:41 AM, Cullen Jennings wrote:
> On Oct 30, 2011, at 9:58 , Christer Holmberg wrote:
>
>> In my opinion a better solution is to create a new PeerConnection for every new early dialog, which uses the *same* address/candidate parameters as the "parent" PeerConnection.
> Hmm we should talk about this - I have a hard time seeing how that is going to work. I was working on the assumption that  PeerConnection could deal with more than one offer/answer pair at a time. Given the current WebRTC API and something like ROAP - it seems like that should just work.
I think having a single PeerConnection object send media to multiple 
peers at the same time breaks the model we've been working from.

It may be reasonable to give a PeerConnection the ability to switch from 
one remote peer to another remote peer (to support 180-like things), but 
I don't think it's reasonable to have it connected to multiple remote peers.

So if we support real forking, we'll have to create multiple 
PeerConnections to do that.

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