12 Nov
2008
12 Nov
'08
11:01 a.m.
2008/11/12 Raju Abhyankar <kf6rzt(a)yahoo.com>om>:
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.
Sorry, but that is not enough information to detect the issue. What I
suggest to you is:
- Do a capture of the SIP flow.
- Examine values in SDP (in INVITE and 200 OK) before and after
passing by the proxy.
- You should see that some SDP media address has not been modified by
rtpproxy so you could examinate why in your Kamailio script.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>