Hello, kamailio + rtpengine can be used for webrtc calls between browsers as well as browser to classic sip phones. You can fine on github some config examples, published by Carlos Ruiz Diaz.
Using this combination you can place an instance in front of asterisk and let asterisk behave as a classic sip/rtp media server. Cheers, Daniel On 02/05/15 00:44, Ivan Vujisic wrote: > I made successful audio calls from browser to browser using Asterisk > 13.1 and SIPML5 browser phone. > > Asterisk can't manage WebRTC video calls due to lack of codec > negotiation module, but I also faced RTP ports NAT traversal issue. To > my understanding Kamailio is capable to resolve this. > > Can anybody confirm that he made successful browser to browser video > calls with Kamailio sip proxy / registrar in front of Asterisk PBX. > > Also, any link to good tutorial or doc pages will be appreciated. > > Best Regards, > Ivan Vujisic > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ 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
