Hi Everyone, Does anyone have a example of the config where I can get the following to work
I want Kamailio to process websocket converting wss to tcp and srtp to rtp and forward to asterisk as tcp and rtp incoming call on websocket (wss rtp/savpf) --> kamailio/rtpproxy (protocol - convert wss to tcp) and (sdp - convert rtp/savpf to rtp/avp) and (audio - convert srtp to rtp) --> asterisk (recieve protocol: tcp, sdp: rtp/avp and audio:rtp) This application would only have incoming call from websocket to forward on to asterisk. _______________________________________________ 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
