[asterisk-users] NAT solutions

[Brad Templeton]

On Mon, Jan 22, 2007 at 09:59:06AM +0000, Tim Panton wrote:
> In the meanwhile, use IAX, which understands about NAT pretty well.
> If you have multiple SIP phones on a LAN behind a NATing router, just
> put a small asterisk box on the LAN. It can manage your hairpin
> calls internally, save you bandwidth by trunking the IAX traffic
> to the central asterisk and avoid all the NAT hassle by using
> a single port (outgoing) and refreshing it often enough for the
> router to hold it open.
> 
> 
> Tim Panton
> 
> www.mexuar.net
> www.westhawk.co.uk/

IAX is a fine protocol as far as it goes, however this answer
is really not a workable one.   There are only a few IAX phones,
and they are not nearly as solid and full featured as the many
SIP phones.   There are some IAX termination and origination
providers, but there are far more SIP providers.

For a remote phone, not on the same network as the Asterisk
box (in which event the NAT worries are different) you definitely
want to use the same protocol for the phone as for your
term/orig provider.   Otherwise you will be forced to hairpin
your audio through your asterisk server, adding latency and
wasting bandwidth and cpu for little reason.

In addition, many people just want to do things like give
family or employees a phone they can take home, or take to
a remote location and use on the PBX.   They probably can't
"just" put up an Asterisk server to make this happen, and
nor should they want to.

An additional server is not only more work and requires an
always-on server computer, it's another thing that can go
wrong.

No thanks.  Even if you can run Asterisk on a WRT54G, and
thus don't have the $200/year power expense of a server,
it's still not what you really want.

IAX is great but SIP is also a reality, and putting
Asterisk into the "just works" category is a really
important milestone.  One I think that is intended
to be improved a lot for 1.6.