[asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

[Joshua Colp]

Matthew J. Roth wrote:
> Joshua Colp wrote:
>> If you set nat=no for that specific peer it should work as you need.
>> 'rport' is forced on these days which works for most situations, except
>> with some platforms and Cisco phones.>_>
>
> Joshua,
>
> That sounds much easier than what I came up with, so I'd recommend to Markus
> that he try your suggestion first.
>
> If you have a moment, please take a look at my response and let me know if my
> understanding of the Contact and Via headers was correct.  If it was, is the
> 'nat=no' solution just a way to workaround the provider's RFC-noncompliant
> platform?

Most of your response is correct except it doesn't take into account the 
rport RFC. Lack of implementation of an RFC doesn't make it 
non-compliant, so their stuff really is fine for this scenario. It all 
comes down to us forcing rport to be on by default.

This is now the second known platform on my list that uses a random 
source port for the IP header instead of the actual one. The other 
being, like I mentioned, older Cisco phones.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org