Hi, I have the following architecture - SIP provider <-> Kamailio <-> Asterisk servers Currently I have everything setup and incoming calls from Sip are routed to my asterisk server. The issue is however that when I answer the call, there is no media in the call. I have tried connecting with a normal local extension(not SIP,eg 1001) and there is a normal flow of media. When i try to sniff my connection via Wireshark on the asterisk server, there is an outflow of RTP packets but the same RTP traffic does not appear on the Wireshark of my Kamailio server connection.  I am not sure if this is an RTP engine issue and how to resolve this. I have - modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038 <http://127.0.0.1:45038>") this in my kamailio cfg but I don;t know which port to use here. Any suggestions? _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev