On 25 Jan 2007, at 06:57, Brad Templeton wrote:
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.
I've never had a problem finding an IAX provider indeed
they seem to be more clue'd :-) than the SIP only ones.
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.
Unless you are monitoring calls, want full CDR etc,
then that's what you want anyway.
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.
I agree. Single SIP phones can usually be got to work behind
a reasonable NAT router.
An additional server is not only more work and requires an
always-on server computer, it's another thing that can go
wrong.
For a single phone - you are quite right. For multiple phones,
I'm not sure I agree - multiple SIP phones behind a NAT router
is going to require some extensive config , or a SIP proxy in the
router.
If you are going to be maintaining a proxy, why not use asterisk
on an NSlu2 or an WRT ?
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.
Ah, but it isn't just asterisk you have to change - it is
all the SIP implementations and all the routers :-)
It will happen, SIP will move such that it uses fewer
ports in a more predictable way (thus becoming more IAX ish)
routers will come with sane SIP proxies etc, but (as I said)
in the meanwhile IAX is a useful tool to have to solve some
of these problems now.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
