Hi Daniel, we've developed a Javascript SIP stack which supports WebRTC and we use Debian as base OS and Kamailio as SIP proxy. Some notes about the enviroment in case it could help you:

- Kamailio stable (4.0) version included in official repo works fine.
  - Site: http://www.kamailio.org/wiki/packages/debs#latest_kamailio_40_release
  - Howto: https://quobis.atlassian.net/wiki/display/QoffeeSIP/Server+configurations
  - I've tested it with QoffeeSIP and JsSIP some days ago and there is no problem.

- I've been playing with "resiprocate-turn-server" package but I had problems. It could be related with our client but we should take a look. I didn't give a try to Google TURN server but I'm going to do it, I'll keep you updated.


PS: I'm co-mentor with you in GSoC, so we can speak about in in IRC channel.


2013/5/13 Daniel Pocock <daniel@pocock.com.au>

I'd like to write a brief blog about the status of WebRTC in Debian,
with a focus on SIP

I understand Kamailio 4.0.1 is already in unstable, is that recommended
for potential WebSocket users?  Has anybody else written any quickstart
blog about WebRTC with that particular version, possibly with examples
that are consistent with the Debian usage?

For client side, SIPml5 is packaged, and I've had discussions with the
JsSIP guys about packaging.

For TURN, has anybody tried the TURN server project from Google code?
It appears more advanced than the existing two TURN servers in Debian
(e.g. it has database-backed authentication)
http://code.google.com/p/rfc5766-turn-server/

and there is a package in progress:
http://mentors.debian.net/package/rfc5766-turn-server




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Jesús Pérez
VoIP Engineer at Quobis

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