Hi, I have a sipphone behind a router doing NAT, an asterisk box in the middle and another asterisk box, which works as gateway to further destinations. The asterisk box in the middle should do all call setup and tear down, but no RTP. RTP should flow directly between the sipphone via the router to the other asterisk box. When calling _from_ the sipphone, everthing is fine: The asterisk box in the middle is reinviting, and the other asterisk box is finally exchanging RTP with the sipphone in both directions. When calling _to_ the sipphone, there is a problem: The asterisk box in the middle again is reinviting, and the RTP stream from the sipphone than goes directly to the other asterisk box. But the RTP stream from the other asterisk box is sent to the private IP address of the sipphone (192.168....) The NAT workaround is not effective in this case. Any hints? Just for understanding: Who is responsible in this case for the NAT workaround: The asterisk box in the middle who is reinviting or the originating asterisk box who is then sending the RTP traffic? Roger.