[SR-Users] RTPproxy issue in forwarding scenario

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