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<DIV><FONT face=Arial size=2>I wondering how FreeSwitch does codec negotiation
in the following call flow:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2> SIP Endpoint A ---->
FreeSwitch ----> SIP Endpoint B</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>For example, if the SIP Endpoint A offers Codecs:
G.711u, G.711a, G.729<BR>when FreeSwitch sets up the Call to SIP Endpoint B, how
does it determine<BR>what list of Codecs to offer to SIP Endpoint
B?</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>What I'm looking for is for Freeswitch to always
offer the same list of<BR>Codecs to SIP Endpoint B as it received from SIP
Endpoint A.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>In this application, having the RTP packets bypass
FreeSwitch and<BR>go directly between SIP Endpoint A and SIP Endpoint B is not
an option<BR>since the two SIP Endpoints cannot talk to each other
directly.<BR>SIP Endpoint A is in public IP space, and SIP Endpoint B is in
private IP<BR>space.<BR></FONT></DIV></BODY></HTML>