asterisk users - Sep 2004 - SIP rtp port forcing

When a SIP call starts (INVITE / 200 OK), asterisk seems to create a 
random port number for voice (rtp) packets. Is it possible to force this 
port value (without using reinvite since i am trying to use SIP against 
something else than sip)

thanks a lot in advance
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check   rtp.conf

-Kannaiyan
  ----- Original Message ----- 
  From: boris.vincent@mindspeed.com 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, September 06, 2004 6:15 PM
  Subject: [Asterisk-Users] SIP rtp port forcing



  When a SIP call starts (INVITE / 200 OK), asterisk seems to create a random
port number for voice (rtp) packets. Is it possible to force this port value
(without using reinvite since i am trying to use SIP against something else than
sip)

  thanks a lot in advance 



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You can only restrict the range of ports used, in rtp.conf.
I suppose restricting it to 2 ports starting on even number might do it,
but if you're not using SIP on one end, how are you going to start a call?
You need to have at least rudimentary call control for SIP invite and SDP
exchange, and given that you now have SDP exchange you should be able
to accept any port presented by asterisk.

boris.vincent@mindspeed.com wrote:
>
> When a SIP call starts (INVITE / 200 OK), asterisk seems to create a 
> random port number for voice (rtp) packets. Is it possible to force 
> this port value (without using reinvite since i am trying to use SIP 
> against something else than sip)
>
> thanks a lot in advance
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

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.
There can be 2 solutions :
        - 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)
        - configure asterisk to force the udp port to match the server 
one.

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).

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.

thanks again




Karl Brose wrote :
> You can only restrict the range of ports used, in rtp.conf.
> I suppose restricting it to 2 ports starting on even number might do it,
> but if you're not using SIP on one end, how are you going to start a 
call?
> You need to have at least rudimentary call control for SIP invite and 
SDP
> exchange, and given that you now have SDP exchange you should be able
> to accept any port presented by asterisk.
boris.vincent at mindspeed.com wrote:
>
> When a SIP call starts (INVITE / 200 OK), asterisk seems to create a 
> random port number for voice (rtp) packets. Is it possible to force 
> this port value (without using reinvite since i am trying to use SIP 
> against something else than sip)
>
> thanks a lot in advance
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