I was at the GSoC mentor summit on the weekend

I put together a session to discuss WebRTC as a collaboration technology.  In that session I ran through the following:

a) the repro SIP proxy as a SIP over WebSocket solution, we used the web interface to set up a bunch of test accounts for people in the room

b) JsSIP as a solution for people to use on their web sites: the volunteers used the test SIP accounts to log in with http://tryit.jssip.net

Everything just worked and several pairs of volunteers were able to make calls between themselves.  Thanks to Google for providing virtually unlimited bandwidth on their campus.

Some other WebRTC solutions were discussed during the session (not all involve SIP):
  https://togetherjs.com/
  http://peerjs.com/
  phono - a client hard-coded to use the Tropo service, appears to offer SIP calling through Tropo

There was also a session about the general state of free social networking (not specifically RTC).  I emphasized the need for free software developers to try and integrate RTC with other social technologies (e.g. a single user ID) to maximise the convenience for users and increase chances of success.

In both sessions I suggested the Free RTC mailing list as a good meeting point for people to pursue further interoperability using free software in these areas:
https://lists.fsfe.org/mailman/listinfo/free-rtc

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