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
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
check rtp.conf
-Kannaiyan
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
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
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
