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

[Terry Wilson]

----- Original Message -----
> From: "Saúl Ibarra Corretgé" <saghul at gmail.com>
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Sent: Wednesday, February 8, 2012 11:43:12 AM
> Subject: Re: [asterisk-dev] [Code Review]: Add a SIP nat=auto setting
> 
> Hi Terry,
> 
> >
> > Of course, since it is trunk, we could also just rip everything
> > apart and do something like:
> >
> > nat_force_rport=on/off/auto
> > nat_comedia=on/off/auto
> >
> > We could keep legacy settings around, but something like this would
> > be a lot cleaner.
> >
> 
> Personally I don't like this option very much. It requires 2 settings
> instead of one and the PBX administrator needs to know what "force
> rport" and "comedia" mean.

The thing is, it really *is* two different settings. We pretend like there is a "nat" setting when there are really several different things that we can set to affect devices that aren't dependent on each other at all. Eh, I don't care very much either way, to be honest. The underlying implementation will end up looking something like the above, and user-facing stuff can be whatever everyone thinks is best.

Terry