[Free-RTC] [JsSIP] Free RTC and WebRTC (repro+JsSIP) demo at GSoC mentor summit

[José Luis Millán]

Thanks a lot Daniel,

I've just subscribed to the list.

Best regards.


2013/10/22 Daniel Pocock <daniel at pocock.com.au>

>
>
> 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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-- 
José Luis Millán
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