[SR-Users] KAMAILIO RTP PROXY

Klaus Darilion klaus.mailinglists at pernau.at
Tue Sep 6 11:16:20 CEST 2011


Make sure force_rtpproxy is called also in reply route, and make sure
the reply route is activated (t_on_reply).

See the default config file for how it works.

For debugging use (or Wireshark ...)
  ngrep -i any -W byline port 5060
and analyze the SDP before and after Kamailio.

Also, add xlogl() statements to the config file before and after
activation of rtpproxy and verify that you see them in the log file.

regards
Klaus

Am 06.09.2011 09:19, schrieb Phillman25 Kyriacou:
> Hi
> 
> I want to use Kamailio to handle both signalling and media always.
> My scenario is the following PSTN==>KAMAILIO==>ASTERISK. I have a
> dialplan that points to ASTERISK when a call comes into Kamailio.
> Signalling is fine, however i cant seem to get the Media to work, i have
> installed the RTPPROXY as suggested by RTPPROXY module and have
> implemented it into the routing logic, however i have one way audio. The
> PSTN side cannot hear anything.
> 
> I dont use the Kamailio for sip registrations or for users behind NAT. I
> just use a simple dialplan for forwarding SIP messages to our certain
> VOIP carriers, however some of our carriers request that we use both
> signalling and media from the same IP thats why i need to implement this
> scenario.
> Is this possible?
> 
> Thanking you in advance for your help!
> Phillip
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



More information about the sr-users mailing list