Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

Tim Panton <tim@phonefromhere.com> Mon, 14 May 2012 09:25 UTC

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From: Tim Panton <tim@phonefromhere.com>
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Date: Mon, 14 May 2012 10:25:45 +0100
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To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:

> Support for multiple registered contacts and (parallel) forking are pretty fundamental to SIP.  This has little to do with conferencing.  Some systems (not all) may support only serial forking or may serialize media interactions associated with parallel forking, but forking remains widely deployed and useful.


I have a feeling that it is mostly useful because SIP is managing both the signalling and the identity. 
In most WebRTC cases, the real identity of the user will be established with a web protocol (perhaps OAuth).
The identity used in any browser based legacy signalling (e.g. SIP) will be a second-order derived or temporary identity.

In that case there is no need for SIP to use the same  temporary identity for both my phones, which (I think)
renders the forking case significantly less useful.

Tim.