Hi,

I currently have an issue where asterisk is not forwarding the RTP traffic 
between 2 endpoints. The SIP session gets set up correctly, and both parties 
get connected. The RTP audio is being sent by both endpoints to the correct 
ports on the Asterisk server as per the session description in the SIP 
conversation. However, Asterisk is not forwarding either endpoint's RTP traffic 
to the other.

When using 'rtp debug' on the asterisk console, it shows that it is receiving 
traffic from one endpoint, but not the other. A wireshark trace reveals it is 
actually receiving traffic from both ends.

It doesn't seem to be complaining or generating any errors that I can see, any 
suggestions on what I can do or where to look to find out what is going on?

Thanks in advance,

Ryan

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to