[rtcweb] SIP on the Web: presentation and video

José Luis Millán <jmillan@aliax.net> Tue, 11 October 2011 06:28 UTC

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Date: Tue, 11 Oct 2011 08:28:48 +0200
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From: José Luis Millán <jmillan@aliax.net>
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Subject: [rtcweb] SIP on the Web: presentation and video
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Some weeks ago we made a concrete proposal for signalling between Web
browser and server in RTCweb. Such a definition was described in

We don't advocate for a default signaling protocol in RTCweb (not at all).
Current proposal is just an example of a working signaling for RTCweb based
on SIP and carried over WebSocket.

We have just uploaded a video demo in which there can be seen a real
implementation of the mentioned draft.

The components of the signalling architecture are a WebSocket capable
SIP proxy and a SIP stack written in JavaScript. These are needed to
make the browser became a real SIP node, capable to interact with real
SIP World.

We would like to emphasize that the aim of this video is nothing but
showing an implementation of the proposed signalling.

We wish you enjoy.



Iñaki Baz <ibc@aliax.net>
José Luis Millán <jmillan@aliax.net>