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.