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