[asterisk-users] NAT solutions

[Brad Templeton]

On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote:
> > On Fri, Jan 26, 2007 at 12:34:30PM +0000, Tim Panton wrote:
> >>
> >> Unless you are monitoring calls, want full CDR  etc,
> >> then that's what you want anyway.
> >
> > CDR are not affected by how the audio flows.
> 
> While technically true, I believe (it may have changed in 1.4) that if you
> allow reinvites, the signalling path follows the audio path, and you end
> up with reported calls lasting 3 seconds.
> 
> So, if you want full (ie accurate as to the length of time) CDR, then I
> think asterisk has to remain in the call path.

That would have been an odd bug.  Signalling in SIP only moves
when you do a REFER or similar.   Reinvites can't change it. 

Having the signalling flow differently from the audio is a feature,
not a bug, a very important one.   With SIP INFO (or its planned
successor) you can get the DTMF without having to get the audio,
which is highly valuable.   Right now Asterisk needs to stay in
the audio stream to get DTMF, and that is one of the prime reasons
it does.  (The others are NAT, recording and meetme, the latter 2
of which should be a small minority of calls.)

This is an important thing.  Done properly, audio should almost
never flow through the switching machine, or only flow for a portion
of the call.  The result can be orders of mangitude difference
in bandwith requirements.