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</o:shapelayout></xml><![endif]--></head><body lang=EN-AU link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Hello,<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I have opensips 2.3 beta, along with a local phone both running inside and behind the same router.<o:p></o:p></p><p class=MsoNormal>I have one port forwarded for opensips to listen on and a port range forwarded for the local phone.<o:p></o:p></p><p class=MsoNormal>There is a remote phone in another domain behind another router, also with a port range forwarded.<o:p></o:p></p><p class=MsoNormal>I am using stun for both phones and this resolves the correct IP domains for each phone.<o:p></o:p></p><p class=MsoNormal>With stun implemented and saying it is full cone on both phones. The phones can now ring each other.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The local phone can call the remote phone and there is two way audio.<o:p></o:p></p><p class=MsoNormal>When the remote phone calls the local phone, there is neither way audio.<o:p></o:p></p><p class=MsoNormal>Invites from the remote phone always appear with the correct expected provisioned sip and rtp ports.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I would expect that the local router is changing the local phones sip contact port when it registers.<o:p></o:p></p><p class=MsoNormal>When I look at a sipgrep capture of an outgoing invite both the sip and the rtp ports are changed.<o:p></o:p></p><p class=MsoNormal>I am not all that sure where in the process or even why the rtp port for the invite has been changed.<o:p></o:p></p><p class=MsoNormal>Inbound calls then of course end up sending and returning rtp through non forwarded port ranges.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>What I would like to understand is how to make an inclusion, when any local phones register,<o:p></o:p></p><p class=MsoNormal>that will allow the outgoing contact details to show the phones actual provisioned sip ports.<o:p></o:p></p><p class=MsoNormal>With that correct, in the outgoing invite, the rtp streams would then normally be within the<o:p></o:p></p><p class=MsoNormal>expected range of ports opened and forwarded to the phones and that would solve the audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I am looking for working examples, but I have not turned up enough specific information about just this.<o:p></o:p></p><p class=MsoNormal>Knowing better where and how to start and the names of what I am looking for would be most helpful.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thankyou<o:p></o:p></p><p class=MsoNormal>Alex<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>