[asterisk-users] NAT solutions

[Tim Panton]

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/