VoIPowering Your Office: Swatting NATs with IAX, Part 2

we began an investigation into ways to deal with the many problems
inherent in getting SIP (Session Initiation Protocol) through NAT (Network Address
Translation) firewalls. There are two basic strategies: spend your life fighting
with SIP—with its attendant hair loss and unhealthy blood pressure levels—or
sidestep the problem entirely by using the IAX (Inter-Asterisk Exchange) protocol.
So we’re looking at the pros and cons of using IAX. We’ve already looked at
some good IAX softphones; now let’s see what the hardphone world has to offer.

That’s right, nothing. Okay, there is one that I found: the Atcom
. It gets decent user reviews, and it is priced under $100. It supports
both SIP and IAX. If you can find a local vendor it is worth a try. Or you can
order it from the manufacturer
in China

Bah, humbug IAX
I contacted a number of commercial vendors of both soft and hardphones, and
asked about their plans to include IAX support in their products. IP phones
already support rafts of codecs and features; some even support multiple protocols
(mainly SIP and H.323), so what’s the problem with adding one more? For some
reason this is a touchy subject; the ones who actually deigned to speak to me
wouldn’t say anything other than “We have no plans to support IAX.” Ordinarily
they’re all happy when I call. So naturally I’m suspecting all kinds of crazy
things: bribery from a shadowy anti-IAX cabal, Mark
got caught igniting dog doo on their doorsteps, or maybe everyone
is having a bad day all at the exact same time, and I shouldn’t take it personally.
Or it’s a cunning plot to prop up the lucrative SIP proxy market.

I know there is substantial customer demand for this—customer forums
are full of requests for IAX support, and they get the same curt “you go away
now” responses. So if any of you fine readers have inside knowledge as to why
this silly state of affairs exists—imagine, customers requesting useful
and valuable features! Such audacity!—I would love to hear about it.

IAX service providers
There are quite a few of these; visit VoIP
Service Providers
for starters. These come in a variety of flavors. Some
are intended for individuals, so you only need an IP phone that supports IAX.
Some support BYOD, or Bring Your Own Device. If you’re running your own Asterisk
server and don’t want to hassle with doing your own PSTN termination, a service
provider that does this for you might be a nice cost-effective way to go. Phone
number portability is still iffy; but you may be able to use your existing phone
numbers, so always ask.

Be sure to visit a few relevant forums for advice and feedback on service providers before signing up, such as the Digium forums, Trixbox forums, and DSL Reports.

Analog IAX adapters
These are worth considering for remote clients that connect to your Asterisk server over the Internet, such as road warriors, small branch offices where you don’t want to hassle with running an Asterisk server, and for setting up your far-flung relatives to talk to you for free. Yes, you’re stuck with using gormy old-fashioned analog telephones, instead of shiny new IP phones, but is that so bad? Well, yes it is, except for your tech-inexperienced users it’s probably better.

Single-line IAX adapters like Digium’s IAXy are popular with travelers.

The Atcom AG188 ATA adapter supports both SIP and IAX, and it even includes a PSTN pass-through.

The GlobetelX IAD-200X is a nice 2-port box cram-full of useful features: NAT router, DHCP server, VLAN, QoS, and more. It’s rather a bear to configure, though, even with a Web configuration interface. But you get a lot of features for the money.

Putting it all together
So let’s review where/how all these pieces fit together. In an ideal world,
you would have nice efficient IAX trunks everywhere, serving SIP and IAX endpoints.
There would be no cursing at the party-pooping NAT, merely happy people talking
and carrying on without a care. So anyone who is running an Asterisk-based server
is a potential IAX peer: your branch offices, friends, partners, and customers.
You can run all the SIP endpoints behind your Asterisk server that your heart
desires, as long as your SIP traffic is escorted through NAT via IAX.

It is true that doing this foils the SIP architecture, which by design separates
signaling from the media stream, and allows the media stream to find its own
best routes. Unfortunately, in a NAT-dominated Internet, the advantages gained
from this flexible routing are lost to all the workarounds required for reliable
NAT traversal. Moral: VoIP is still in its infancy, and is therefore messy,
as babies always are.

Reams of good information on SIP and NAT

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