Just in case someone is interested, I created a sample script that could help new comers having the same problem.
I will write a blog entry explaining how this works, but in a nutshell: - this script is configured to run behind NAT, port TCP 10080 and TCP/UDP 5090 are exposed to the Internet - you have to create valid users using, preferably, "kamctl add ..." - RTP ports should be open in range 30k-35k, inclusive - I used jssip as WEBRTC SIP UA: http://tryit.jssip.net/ - Always disable video before placing a call from jssip UA - I tested calls between: - jssip to csipsimple - csipsimple to jssip - csipsimple to csipsimple Link to the scripts: https://github.com/caruizdiaz/kamailio-ws Regards, On Sat, Feb 22, 2014 at 9:31 AM, Richard Fuchs <rfu...@sipwise.com> wrote: > On 02/22/14 07:07, Mihai Marin wrote: > > Hello Sirs, Sir Richard, > > Thank you for your detailed explication. > > I'm still thinking on that but I would say to act as the caller and keep > > caller decision. If caller makes an offer with rtcp-mux , > > include separate ICE candidates for RTCP for media proxy too and forward > > as it is to alice. If callee accept it (or not) you will receive the OK > > with alice sdp, modify it (depending on her choices) and forward to bob. > > In this way, we cover all the cases. Eventually we can add another > > parameter to always ignore rtcp-mux offers. > > > > What are the disadvantages on doing that? Is there any possibility that > > some SIP clients not to respond properly to an SDP with rtcp-mux and > > that's why you are removing it - or for '+' case where delay will be > added? > > Compatibility is exactly the reason. I don't have any exact numbers, but > I'm sure that there's a large number of SIP/RTP clients out there (I'd > say the vast majority) which don't support rtcp-mux at all. Some of them > might start misbehaving if they receive an rtcp-mux offer (even though > as per RFC, they shouldn't, but experience shows that RFC compliance is > often just wishful thinking). Since from our point of view (always > either '+' or '-') there's no disadvantage in always demuxing RTCP, this > was what was implemented. > > In any case, I'll see if I can get a solution implemented in the near > future. > > cheers > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
