[SR-Users] Kamailio Failed to have audo converstaion with RTP

[Karsten Horsmann]

Hi Ryan,


if you didnt use the nating Route - rtpproxy_manage() would
never called and so rtpproxy didnt work.

Try to use rtpproxy_manage and use xlog to show that is fired up.

2012/4/11 Ryan Gholam <ryangholam at gmail.com>:
> Dear Daniel ,
>
> I thank you for your reply , I have a server  having the Astersisk  ip
> address (192.168.10.15) , and rtp + kamailio is installed on an
> another pc have the  following ip (192.168.10.17) which is linked to
> the Astersik , and on the same pc , another network card exists having
> the ip address (192.168.20.3 ) which is linked to a client pc having
> the ip address ( 192.168.20.4) .
>
> I tracked the call and i can see  SIP ACK nd BYE between 20.3 and 20.4
> but there is no audio conversation   this is my configuration file for
> kamailio attached above .
>
> P.S :  testing without  NATING as described in the above setup  .
>
> I thank you alot again for all your help .
>
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-- 
Mit freundlichen Grüßen
*Karsten Horsmann*