[SR-Users] bypass rtp traffic.

[Klaus Darilion]

Am 13.07.2011 10:07, schrieb MingHon:
> Hi,
> 
> i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still
> trying to send rtp traffic to asterisk.

That should not happen. You have to investigate why. You have to take a
look at the SIP signaling during and after call setup.

You should see reINVITE messages from Asterisk to the clients. Take a
look at the SDPs in those requests and their responses to find out if
they are malformed.

regards
Klaus