Re: [SR-Users] Caller Not Receiving RTP Feed

Hey all, **I apologize if this is a duplicate, I received a bounce back on my first email. I have kamailio set up behind a nat(port restricted, with firewall rules to allow sip transactions and allowing rtpproxy packet handling if needed) on Amazon EC2. I can register and calls complete, however, the Caller(the one initiating the transaction) receives no rtp media feed. I am running with NAT enabled on kamailio and have rtpproxy installed listening on the public IP. Kamailio and the rtpproxy are communicating(I have verified via the kamailio debug logs). If I make a call between the exact same voip machines directly via local IP on the same Nat(skipping kamailio), the calls complete and they both receive feeds. Both the Caller(party making the call) and the Callee(party receiving the call) are behind a Port Restricted Nat. This is a folder containing the debug output for one of these calls and the kamailio.cfg settings https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp… Quick FYI, the Caller Display Name and the Callee SIP UserName are the same string. However, in my understanding about sip, the display name means pretty much nothing and is just a human readable string that does not effect packet transport. If I am wrong and should test with a different display name, let me know. Thank you for the assistance, ben _______________________________________________ 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