I'm a SIP/SER newbie, so please forgive me if the answer to this is obvious to someone experienced: I currently have SER running as a registration and outbound proxy server, and I am able to forward calls to a PSTN gateway. Right now SIP-to-PSTN calls are establishing sessions, being neither side is able to hear the other talking. The softphone user is behind a restricted-cone NAT, for which I'm using a STUN server to traverse. I have surmised that this is an issue of the RTP packets not being able to reach the client through the firewall, but I would expect the PSTN phone to be able to hear the audio from the client. Is there a simple cause of and solution to this problem that I'm overlooking? John