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

[Saúl Ibarra Corretgé]

Hi Kevin,

>>
>> +1, I think nat=auto is a really good thing and it should be the default.
>
>
> Can you explain *why* you think it is valuable? (and before doing so, please
> the comment I just made on reviewboard)
>

I might be missing something, but this is my reasoning:

Currently the default setting for nat is force_rport, which means that
SIP will work, but RTP will not, since comedia mode is no enabled.
Personally, I would set nat=yes by default, but I guess there are
devices that don't like this too much. I've been lucky not to run into
them yet :-)

This new setting (if I got it correctly) would toggle the same
machinery as nat=yes if the source is behind nat, which I believe it's
a good idea, an SIP account should work on any condition.

Also, FWIW, a free SIP service we offer at work forces rport always
(if source is behind NAT) and we use MediaProxy for media relaying, so
comedia is always used.

I'm all in for flexible settings, but I believe configuration defaults
should try to cover most use cases.


Regards,

-- 
/Saúl
http://saghul.net | http://sipdoc.net