[asterisk-dev] [Code Review] Add a SIP nat=auto setting

[Luke Hamburg]

I'll admit I don't have concrete examples- it's been months since I've
experienced NAT issues myself.  However, earlier last year when Pedro first
released this patch it DID resolve some 1-way audio issues with a few
(buggy?) softphone clients that were connecting over 3G to a 1.8.3(?) server
which was itself behind a NAT.  I think that current 1.8 trunk with nat=yes
set everywhere no longer exhibits that specific issue.

In any case this is a "feel good" patch because it takes the guesswork out
of the nat= settings which, I believe to this day are still
oft-misunderstood and sometimes result in needless frustration.

Luke


-----Original Message-----
From: Kevin P. Fleming
Sent: Friday, January 27, 2012 11:12 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [asterisk-dev] [Code Review] Add a SIP nat=auto setting

On 01/27/2012 10:05 AM, Luke Hamburg wrote:
> This is a really useful patch!

I'm curious as to what this patch resolves for you; do you have endpoints
that do not operate properly with 'nat=yes' (or 'force_rport' or 'comedia'
separately)? If so, what are they, and do you know why this is the case?


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Kevin P. Fleming
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