<br><font size=3 face="Courier New">To make it simple, asterisk is running behind a kind of server. They both have the same ip address. The server has its own udp port opened. If an incoming rtp packet is on one of these ports, the server swallows it, else it forwards it to asterisk. The goal is to get the server to swallow the rtp flow coming from the SIP phone.</font>
<br><font size=3 face="Courier New">There can be 2 solutions :</font>
<br><font size=3 face="Courier New"> - configure the server to use the udp port asterisk created (this requires to know the udp port create, i don;t know if that's possible)</font>
<br><font size=3 face="Courier New"> - configure asterisk to force the udp port to match the server one.</font>
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<br><font size=3 face="Courier New">Since multiple SIP calls must be possible, the rtp.conf solution won't work as it can only force 1 udp. The solution would be to change the rtp.conf and reload it each time a sip call comes in (don't know eather is that's possible).</font>
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<br><font size=3 face="Courier New">anyway, thanks a lot for the tip, i didn't know about the rtp.conf file. If 1 of the above solution is possible, please let me know. And if you know of any other possible solution, i would be very happy to hear about it.</font>
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<br><font size=3 face="Courier New">thanks again</font>
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<br><font size=3 face="Times New Roman"><b>Karl Brose</b></font><font size=3 face="Courier New"> wrote :</font>
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<br><font size=3 face="Courier New">> You can only restrict the range of ports used, in rtp.conf.<br>
> I suppose restricting it to 2 ports starting on even number might do it,<br>
> but if you're not using SIP on one end, how are you going to start a call?<br>
> You need to have at least rudimentary call control for SIP invite and SDP<br>
> exchange, and given that you now have SDP exchange you should be able<br>
> to accept any port presented by asterisk.<br>
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</font><a href="http://lists.digium.com/mailman/listinfo/asterisk-users"><font size=3 color=blue face="Courier New"><u>boris.vincent at mindspeed.com</u></font></a><font size=3 face="Courier New"> wrote:<br>
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><i><br>
</i>><i> When a SIP call starts (INVITE / 200 OK), asterisk seems to create a <br>
</i>><i> random port number for voice (rtp) packets. Is it possible to force <br>
</i>><i> this port value (without using reinvite since i am trying to use SIP <br>
</i>><i> against something else than sip)<br>
</i>><i><br>
</i>><i> thanks a lot in advance<br>
</i></font>