I have been using openser/kamailio for some time now and have two ports 5060 and 5090
(both udp) in the configuration or .cfg file. I do use rtpproxy (and have used
mediaproxy)for NAT traversal.
What I have noticed is that a phone which uses the standard SIP port 5060 (UDP) and calls
a phone using 5090 (UDP), the call goes thru using "loose route" and audio also
is passed both ways. But when a call from a phone using port 5090 (UDP) is made to a phone
which is using port 5060 (UDP), the call goes thru using "loose route" and the
bell also rings but there is no audio.
Any suggestions on how to make this audio work in this situation?
Thanks in advance...
Raju