[asterisk-users] IAX vs SIP trunks between Asterisk boxes

[Brad Templeton]

On Sun, Jan 07, 2007 at 04:12:27PM +0000, Thomas Kenyon wrote:
> Brad Templeton wrote:
> 
> >
> >For SIP phone calling * box, relay to other * box and out to SIP
> >phone, you definitely want SIP all the way.
> >
> Unless bandwidth between the * servers is a concern, then you're better 
> off keeping the link between servers as IAX. (preferably trunked)

The bandwidth of the audio stream dwarfs the bandwidth of signalling
traffic by orders of mangitude.   So in fact, I think this is exactly
wrong.  If bandwidth to or between the servers is a concern, that's
where you most want to not be in the audio path.
> 
> It is worth remembering in this sort of setup, often the phones at one 
> site will not have a route to the phons on the other site, so the calls 
> wont be re-invited off to the handsets anyway.
> 

If it's phone-on-NAT to phone-on-different-NAT, it typically will
not work.

That doesn't mean it can't work if bandwidth is important.

I think the complete solution, not yet in Asterisk as I understand it
is for Asterisk to be aware of both the internal and external addresses
of a phone, and to connect internal phones with their internal addresses,
but to connect internal phones to external endpoints through their
external addresses.   Ideally audio never flows through asterisk unless
it's doing an IVR dialogue or otherwise explicitly wants it to.
(In fact, ideally DTMF goes via SIP INFO or its successors so that
Asterisk can listen to the DTMF without being in on the audio.)

Flowing audio through your box costs not just bandwidth, it adds
latency, and very slight extra risks of packet loss.  Latency is the bane
of voip calls, it also worsens echo.