Re: [rtcweb] Mapping between SIP and ROAP/JSEP

"Ravindran, Parthasarathi" <> Wed, 09 May 2012 07:15 UTC

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From: "Ravindran, Parthasarathi" <>
To: Harald Alvestrand <>, "" <>
Thread-Topic: [rtcweb] Mapping between SIP and ROAP/JSEP
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Date: Wed, 09 May 2012 07:15:18 +0000
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Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP
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My guess is that running code already exists in the form of SIP over websocket. The point is not about the running code. The running code will not cover all the offer/answer state machine issues which I'm talking about. The running code is just equivalent to proof of concept but will not cover all the deployable solution and does not cover all the existing offer/answer .

Even though JSEP is projected as API, offer/pr-answer/answer state MUST be traversed in some magic way between two web-browsers for establishing the session and these states are not possible to directly map with RFC 3264 or RFC 3261 (SIP) usage. So, there is a need of interworking document. The point to be noted is that the interworking between JSEP to SIP shall occur at browser or at WebRTC server.


From: [] On Behalf Of Harald Alvestrand
Sent: Wednesday, May 09, 2012 12:31 PM
Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP

On 05/09/2012 08:52 AM, Ravindran, Parthasarathi wrote:

I agree with you that Interworking document for JSEP to SIP is important for developing the interworking solution. I'll interested in contributing one such draft. Such a draft will help in identify the gap in JSEP to support SIP interworking like UPDATE during early dialog.
Since JSEP is an API, and has been implemented, I suggest that the most reasonable form of documentation is an opensource SIP implementation on top of JSEP, not an internet-draft.

If someone wants to contribute such code, I'm sure we can host it as part of webrtc-samples, if that helps - the other piece would be a recipe for setting up some opensource SIP server to prove that it actually works.

Running code has the advantage that it is easy to test it.