Hi,
we are building a setup where we use an rtpproxy in all cases. This works
fine except for one scenario.
Caller -> SIP(+rtpproxy) -> B2BUA -> SIP(+rtpproxy) -> Called
In this case, the B2BUA implements forwarding and sends the call back
through our setup. The B2BUA does not send out a 183 reponse by itself.
Now, when the caller sends the INVITE, the rtpproxy gets enabled in both
cases. The caller sends his RTP to the rtpproxy, after a 183 or 200 OK
response, the called sends RTP to the rtpproxy, too, But since the B2BUA
doesn't send any audio, both rtpproxies don't know where to pass on the
RTP.
Does anybody know how to circumvent this issue? I searched for an option to
tell rtpproxy to send the RTP to the address advertised in the SDP as long
as it hasn't received any packets on the port, but couldn't find it.
Any hints?
Thanks in advance,
Sebastian