[asterisk-users] Odd one way RTP on SIP to SIP calls

[Örn Arnarson]

Hi everyone,

I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP server.

When a call comes to the Nortel server through the PSTN and is routed
to the Asterisk, audio is fine. Two way RTP and no problems. When a
SIP client registered on the Nortel server calls the Asterisk, the
Asterisk doesn't seem to send any RTP.

As far as I can tell, there isn't anything wrong with the call setup.

show core version shows:
Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
2007-05-17 06:39:34 UTC

SIP and RTP debugging on Asterisk shows this:
http://www.arnarson.net/~orn/calldebug.txt

On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
root @ build.trixbox.org on a i686 running Linux on 2007-04-25
19:59:21 UTC) on the same network (same subnet and physical location)
as the 1.4.4 this problem does not exist. There is no RTP problem when
SIP clients registered on Nortel call.

If anyone could help or suggest anything it would be greatly appreciated.

Best regards,
Örn