[asterisk-dev] Including port number on SDP Contact
serg at maxnet.ru
Fri Jan 25 04:42:47 CST 2008
Robert Moskowitz ?????:
> Johansson Olle E wrote:
>> 13 jan 2008 kl. 23.09 skrev Robert Moskowitz:
>>> I have discovered a challenge with Asterisk running with a public IP
>>> address and thus not needing NAT, when working with a service that of
>>> course knows there has to be a NAT there....
>>> I come here to the developer's list, as this requires some inside
>>> understanding of SIP/SDPs.
>> At this point, you have to change the source code.
>> If there's no port number mentioned in a SIP uri, it means that we
>> default to the default in the RFC, which is 5060. So there's should be no need to add
>> it to the URI and it should not make any difference in an ideal world ;-)
> Being one of the co-authors of RFC1918, I know the difference between
> and ideal world and the Internet :)
> So if the SIP trunking definition for Broadvoice had, say, a port=6050,
> then :6050 would be in all the 'appropriate' URIs. Then I would need an
> IPTABLES packet mangler to attack all SIP packets, changing the port
> from 6050 to 5060 and all imbedded URIs as well.
i don't think that your sip port has anything to do with audio in your
call. Check your rtp.conf to find out which ports are used for audio.
Most likely UDP 10000..20000 - open that range in your firewall and try
to make call.
> Scary thought. Guess I will just have to threaten Broadvoice that they
> fix this or I am going elsewhere. As their behaviour is what is broken...
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