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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users