I wondering how FreeSwitch does codec negotiation in the following call flow: SIP Endpoint A ----> FreeSwitch ----> SIP Endpoint B For example, if the SIP Endpoint A offers Codecs: G.711u, G.711a, G.729 when FreeSwitch sets up the Call to SIP Endpoint B, how does it determine what list of Codecs to offer to SIP Endpoint B? What I'm looking for is for Freeswitch to always offer the same list of Codecs to SIP Endpoint B as it received from SIP Endpoint A. In this application, having the RTP packets bypass FreeSwitch and go directly between SIP Endpoint A and SIP Endpoint B is not an option since the two SIP Endpoints cannot talk to each other directly. SIP Endpoint A is in public IP space, and SIP Endpoint B is in private IP space.