[asterisk-dev] Including port number on SDP Contact

[Sergey Tamkovich]

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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