Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine.
Kamailio installed as frontend (registration, auth, proxy ) of asterisksk servers. WEBRTC users registred at kamailio and asterisk works as media server. When I try to call from Jssip from Firefox to chrome to way audio is fine. When I call from chrome - I see rtp packets only from firefox. Not from chrome. At kamailio log when I call from chrome log the same as whe i try to call from firefox (I can not see anithing wrong) So somebody else have same issue and may be somebody know reasons of this. Thanks for advice
_______________________________________________ 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
