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
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,