Hi Paul!
If you use the nathelper module, you can use force_rtpproxy with the 'l' parameter. Check nathelper.c for documentation.
I don't know if there is an equivalent feature in the mediaproxy module.
regards, klaus
Java Rockx wrote:
Hi All.
I'm using ser-0.9.1.
Is there a way to determine if mediaproxy is in use for an existing SIP call so that re-INVITE messages can avoid losing audio when one or the other SIP UAs are NATed?
We do not proxy RTP streams unless one or more SIP UAs are NATed. But on re-INVITE messages I cannot figure out how to test the destination of the re-INVITE for the UAs NAT status.
I've scoured the archives and found a few related articles http://lists.iptel.org/pipermail/serdev/2004-March/001515.html http://lists.iptel.org/pipermail/serusers/2004-March/006806.html
But nothing has led me to a solution. I cannot just use lookup("location") and test the nat_flag since that won't always work on re-INVITEs. A mediaproxy function for something like is_existing_session() would be awesome to lookup the Call-ID in the existing mediaproxy sessions.
What am I missing?
Regards, Paul
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