One of the problems often encountered with VoIP is the use of the SIP protocol, specifically when used between a VoIP adapter such as the venerable Linksys PAP2 (or something similar) and an Asterisk server. Everything works great until you try to deploy an adapter at a remote location and the user has some kind of funky NAT firewall that just doesn’t play well with SIP. If you have any doubts that this is a real issue, Google the phrase “one-way audio.”
The crux of the problem is that SIP uses a wide number of ports, and unless you get everything set up exactly so at both ends, you may find that although the VoiP adapter will register with the server, when you actually attempt to place a call it will go through but neither party will be able to hear the other, or more commonly, only one party will be able to hear the other.
It’s an especially frustrating problem for VoIP providers, because a certain number of customers — we don’t know how many, because it’s not the sort of statistic providers like to report — cancel their VoIP service because they just can’t get it working in the first place, and the one-way audio probably certainly must account for a not-insignificant percentage of those cancellations.
There is another protocol that Asterisk supports, called IAX (actually it’s now up to IAX2, but we’ll just call it IAX to keep things simple). It works quite well in “difficult” situations because it only uses ONE port for all information. Therefore, if you can see the device register on your Asterisk system, you know that you’re also going to get audio. Assuming you don’t have problems with excessive packet loss or something of that nature, an IAX connection will almost always work, even if the end user is behind some sort of funky router (or even a chain of routers). The main reason that IAX hasn’t been more widely used is that there has been a real lack of decent ATA’s that support IAX (also, I’ve heard some comments that it doesn’t “scale” well in very large installations, but unless you plan on competing with one of the large commercial VoIP providers, I wouldn’t worry about that). For a while, Digium (the company behind Asterisk) made a unit called the IAXy, but like much of the hardware that Digium sells it was priced just a tad on the high side, and everyone was looking for a less expensive solution, so SIP-based ATA’s have become the norm.
But that doesn’t mean that the need for a decent IAX-based adapter has gone away. Just about every Asterisk system administrator has run into one or two “tough cases” where SIP just wasn’t cutting it, and if they did manage to resolve the issue they probably acquired a few grey hairs in the process. If you are the administrator of an Asterisk system, perhaps you’d just like to be able to pre-configure an ATA before sending it to someone that’s doing some off-site work for you, and know that when they receive it they can plug it into whatever crappy old router they happen to have and it’s probably going to work — right out of the box, with no frustrating hours of “… try changing this setting …” as you burn up cell phone minutes.
Recently I became aware of the fact that there’s an adapter available that can communicate using both IAX and SIP, and it even has both a WAN and a LAN port so that if the end-user doesn’t even have a router, they can plug this unit into the cable or DSL modem, and then plug their computer into the LAN port on the adapter and both will work. I made arrangements to obtain a unit for review purposes, and in the coming days I plan to tell you more about this device. It’s called the Atcom AG-188N and it’s sold in North America by CIGear. Just to give you a small preview, here are a few “unboxing” pictures. The small box below came inside a larger carton with all the requisite packing peanuts, but I’m not a “pro” at making these photos (a couple came out way too blurry to use) so I decided to just show you the interesting part:
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Good things come in small packages
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Out of the carton
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Contents of the box, without the bubble wrap
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Closeup of the AG-188N, sorry it's a bit blurry
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The back side, showing the connections, from CIGear's web site.
EDIT: Since I first posted this review I’ve since found out that the particular unit I received had been previously used on a demo and therefore had been repackaged. Normally the power supply comes in a small white box, not wrapped in bubble wrap. Since I had specifically asked for a unit for review purposes, it doesn’t bother me at all that I received a unit that is new for all intents and purposes, but can’t be sold as new – that’s the perfect unit to send to a reviewer.
Note that the yellow ruler doesn’t come with the unit, it’s just a 7″ ruler I had lying around that I put in the pictures to give you an idea of relative sizes. Actually, if you set this unit next to a Linksys PAP2 you’d be very hard-pressed to tell any difference in the size. One thing notably missing from the contents of the box: A network cable. If you were going to buy these in quantity to send out to users, you’d probably also want to obtain some some network cables in bulk, so you could throw a cable in with each shipment, because for every guy like me who’s somehow obtained a small surplus of them, there will always be the folks that only have ONE network cable to their name.
One thing you may notice in the last image above is that there is a phone port and a PSTN (telephone company line) port. Yes, that does mean that this is a single line unit. But, don’t get the idea that because there is a PSTN port, that this is the full equivalent of something like a Sipura SPA-3000 or Linksys SPA-3102, because it isn’t — the PSTN port is a “passthru” port, allowing you to send some calls to the PSTN and others to your VoIP provider based on the pattern dialed, and letting you receive calls from both the PSTN and your VoIP provider without the need use two separate telephones. The difference between this and one of the aforementioned Linksys/Sipura boxes is that you can configure the Linksys/Sipura’s PSTN port to be a trunk on the Asterisk server (essentially an FXO port), and that particular functionality is not included with this unit.
EDIT: The above paragraph was edited slightly after receiving some clarification from CIGear — they also offered this additional information: With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu (this gets a bit into configuration, which will be discused much more thoroughly in upcoming posts).
Of course, you don’t HAVE to connect a PSTN line to this unit, and you don’t HAVE to connect a computer to the LAN port — I have used it successfully tonight without doing either. Ah, but I’m getting ahead of myself a bit – more on that in the next installment, after I’ve had some time to spend working with the unit and reading the documentation (yes, I did get it working mostly without referring to the documentation, so for those who hate wading through documentation, fear not — it’s really not difficult to set up)!
Disclosure: CIGear provided me with an Atcom AG-188N for review purposes, and allowed me to keep it after I was finished writing this series, and for that I am most grateful.
Articles in the series: Review of Atcom AG-188N IAX+SIP ATA (VoIP adapter)
Part 1 – The unboxing
Part 2 – Initial setup using IAX
Part 3 – Setting the time and configuring outbound dialing
Part 4 – Setting up SIP, and securing the adapter
Part 5 – Networking and Internal Router
Part 6 – Final Thoughts and Summary Review
Part 7 – Addendum
Image may be NSFW.
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Image may be NSFW.Clik here to view.
