Alex,
The media ports are configured in rtp.conf. Also, note that Asterisk sends RTP packets out the same ports it expects them to return on. This has the effect of creating a NAT mapping for that 5-tuple, as well as opening a hole in your firewall (naturally, YMMV depending on exactly what you're running for a firewall).
One interesting consequence of the way Asterisk works is that if you don't have anything behind the NAT/Firewall that's generating RTP packets (ie, no audio) no hole gets made and incoming packets will get rejected. I recently ran into an interesting problem with two SIP phones trying to talk through Asterisk behind a (non-NAT) firewall.
The problem was both phones were sending RTP to the Asterisk box but the firewall was blocking both RTP streams because Asterisk never sent any RTP out those ports. And the reason Asterisk hadn't sent RTP out those ports was because it was waiting for RTP from each of the two SIP phones. This was the classic chicken-and-egg scenario.
I resolved it by opening up the firewall for the range of ports I had configured Asterisk to use for RTP. A better solution would be fore Asterisk to always send a "starter" RTP packet so that it can ensure that the firewall opens up.
-brian
Alexander Simeonidis wrote:
Hello everybody,
I'm new to Asterisk and I'm trying to configure the SIP side.
I use Asterisk under the following configuration:
SIP Proxy ---- INTERNET ---- | NAT FIREWALL | ---- Asterisk ---- SIP Phone A
I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution?
Regards,
Alex.
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