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

[Kevin P. Fleming]

On 01/27/2012 02:53 PM, Luke Hamburg wrote:
> 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.

This is what I suspected would be the case, and I thank you for 
responding so quickly.

The reason I'm being so picky about this is that *any* code change of 
this type has the potential to cause regressions, and so we need to be 
pretty sure that we need the change to justify putting it in. With the 
recent change to the default setting for 'nat' in chan_sip, I think that 
a large percentage of the situations that didn't use to work 'out of the 
box' (and required tweaking/changing the 'nat' setting) now *will* work 
(as you've pointed out above). We'll see if we get more feedback from 
other users about cases where 'nat=auto' would be demonstrably helpful.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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