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I was at the GSoC mentor summit on the weekend<br>
<br>
I put together a session to discuss WebRTC as a collaboration
technology. In that session I ran through the following:<br>
<br>
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<br>
<br>
b) JsSIP as a solution for people to use on their web sites: the
volunteers used the test SIP accounts to log in with
<a class="moz-txt-link-freetext" href="http://tryit.jssip.net">http://tryit.jssip.net</a><br>
<br>
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.<br>
<br>
Some other WebRTC solutions were discussed during the session (not
all involve SIP):<br>
<a class="moz-txt-link-freetext" href="https://togetherjs.com/">https://togetherjs.com/</a><br>
<a class="moz-txt-link-freetext" href="http://peerjs.com/">http://peerjs.com/</a><br>
phono - a client hard-coded to use the Tropo service, appears to
offer SIP calling through Tropo<br>
<br>
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.<br>
<br>
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:<br>
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<a href="https://lists.fsfe.org/mailman/listinfo/free-rtc">https://lists.fsfe.org/mailman/listinfo/free-rtc</a><br>
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