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Go the next step with VoIP - don't miss out. My S&TT Post for this week

By Andrew Pollack on 03/02/2006 at 11:28 AM EST

Forget Skype and Vonage for a minute. You may be just touching the surface if you’re using Vonage or one of its competitors, Skype, or any of a myriad other VoIP tools. There is so much more out there. My contribution to SNTT is to show you how to get started and take it one step further.

Introduction - Its not as complex as it sounds. All it really comes down to is this:

1. You have a standard that defines what a computer-telephone is. The standard lays out how one computer-phone talks to any other computer-phone, and how they send sound back and forth to each other. Just like with 'real' analog telephones. There are implementations of that standard (computer-phones) that are built to look and feel like telephones. There are implementations of that standard which are purely software -- many use a picture of a telephone for the interface. There are devices that hide the implementation in a little plastic case, then let you plug your own regular telephone into it. Still, all we're talking about is a phone.

2. There are products and services that let you connect to other people who have these phones. Although you can connect directly, its difficult and not very effective. There are FREE services which make it easier.

3. There are products and services you can buy which make it easier to connect with people who do not have these computer phones, but are still using old style "real" phones or cellular phones.

4. There are hardware and software products which let you be the service that connects the computer-phones together. Think of your office PBX as a hardware product that connects your office phones together. Asterisk is a software PBX that can do EVERYTHING your office PBX can do for both analog "real" phones as well as these "computer-phones". Asterisk is free. It runs on Linux. Linux is free. If you need to use them, there are hardware products that you plug into a machine running Asterisk that let it do the hardware stuff of talking to your existing office phone network. Thus, Asterisk can replace your existing office PBX.

5. The whole thing is recursive. If you are your own service for your office network (as in 4 above) you can (and should) still use the services in 2 and 3 above to connect to other people doing the same thing, and to the rest of the world which hasn't caught on.

Here's how to start

To talk to people you need a phone (or software), an address (or phone number), and an agreed way to connect and share sound with someone who also has a phone and an address.

Get a phone that is compatible with the rest of the VoIP world. There are many devices out there you can buy, but start with a software one. If you must buy hardware, do not buy any device which is going to use just one network (a Skype phone for example). Instead, get a device that communicates using SIP or IAX2. Generally, you’re going to find SIP much more common.

So, step one is to pick a phone. It can be a software based phone or a bit of hardware. It could even be a cheap device that connects with SIP to a provider on one end, and your analog phone on the other. That’s what your Vonage device does.

Now, you have a phone and can call out – in theory. You probably need to register with a SIP Proxy in order to do anything useful. Without that, hardware or software based, your SIP phone will be listening for calls on port 5060. There are a number of firewall issues with this, and not just related to that port. Data traffic often traverses other ports. This means that although your phone has an address already, its probably not going to be what you need right away. By registering with a SIP proxy service, your phone is making an outbound connection and registering, and data will then flow through that back to you avoiding some firewall issue. SIP is difficult with firewall traversal, but it can be done.

Since you’re just getting started, I’d recommend you try out “Free World Dialup” at http://www.freeworlddialup.com. They have a client side phone you can download called the “Pulver Client” but you don’t have to use that one. You may want to look at others.

Once you’ve registered with FWD, you have a very useable SIP address. First of all, anyone with FWD can call you just knowing your number. Users of SIP phones who do not have FWD can call your SIP phone via FWD as well. Your address will be “sip:#####@fwd.pulver.com” -- but try doing that on a numeric push button phone.

Well, you don’t have to. First, many other VoIP providers provide numeric sequences that allow their users to access FWD numbers. In addition, once you have a valid SIP address that works, you can sign up for a number at SIPBROKER ( http://www.sipbroker.com ) and that does two things. First, it provides even more direct paths to you from most providers; second, it provides a way for you to connect to the SIP phones of other people.

What about other people – with ‘real’ phones?

Direct Inward Dial – is what you think of as your telephone number. It’s actually an address that is given to you by your telephone provider and is part of a big international agreement that defines who gets what address and how the call should be routed. It’s a little like getting a public IP address. You can buy one in almost any country code or area code in the US and have it routed to your SIP phone. Service ranges in price and plans from as little as a few dollars a month but on a pay per minute basis for use, to full featured services like Vonage with unlimited use and voicemail features. You can also buy the ability to make OUTBOUND calls to PSTN numbers. Sometimes, you buy this together in a package. You don’t have to.

For my office lines, I’m moving to a phone number provided by connect.voicepulse.com in my area for $11/mo that includes unlimited inbound calls with up to four calls at once. It has no frills at all. If my SIP phone (or in my case my Asterisk server) isn’t there to take the call, the call doesn’t go through. If more than 4 people call at once, they get a fast busy signal. Of course, that only counts for inbound calls.

There are MANY other ways to buy this. For example, I can (and may) buy an 800 number for $2.00 per month plus $0.02 per minute (2 cents) for the calls that come in. I could get a number with an unlimited number of concurrent calls in at the same time, and pay as little as 1.1 cents per minute for the time used by those inbound callers. I can even buy unlimited inbound calls to a pre-determined number of concurrent use lines (called channels) at a cost of about $20 per channel – and then buy DID numbers to route to that pool of channels for $2 each in whatever country or US area code I want.

For outbound calling, I’ll be using Voxee and paying about 1.1 cents per minute for any call I make in the U.S. and about 1.4 cents for most calls to Europe. I don’t have to use just Voxee though. I could decide to use them for US calls, and sign up with a European based provider for those calls. I could sign up for 20 different providers and use a complex table to determine which one to use for each call. It all gets configured in the “dial plan” for my phone (or in my case my Asterisk PBX). There are even services that you can sign up for to do this for you – called “Least Cost Routing” automatically.

Stepping out even further….

It gets even better – There is a standard called “ENUM” which works like DNS. You can take any telephone number for which you are responsible, and assign it an ENUM entry at http://www.e164.org/. You basically say “Here’s my home phone, but if someone wants to call me with VoIP, they could just use sip:#####@fwd.pulver.com instead.” For those of us with Asterisk VoIP servers and for any provider who wants to use the system (many VoIP providers automatically do, not many PSTN of Cell Phone companies do) they will route the call directly to your VOIP phone.

For example, if I use my SIP phone in the office which is connected to my own sip proxy (an Asterisk telephone switch running on Linux) to call your office number in another state, I just punch in your telephone number. I can do that, because I’ve signed up with an outbound dialing service to the PSTN. Now, even before my Asterisk box picks which of the PSTN service providers to place the call on (as I discussed above), my server does an ENUM lookup. If that results in me seeing that you (or your provider) has listed your PSTN number with a SIP alternative, I’ll get connected DIRECTLY to your SIP address and not incur and charges for per-minute dialing because I didn’t use any PSTN resources.

That means if you’ve got a working SIP phone address and you’ve registered your phone number – given to you by old fashioned telco, a VoIP telco like Vonage, or one you purchased from a DID provider like I did – my system will place the call directly to that SIP address and bypass the public phone system entirely, saving us both money.

To make this easy, I’ll be using http://www.voxalot.com to do this form me – and you can to. You can sign up with www.voxalot.com with just your regular SIP soft phone or hardware phone and make them part of your dial plan.

This is just a TASTE of what you can do with VoIP. Give it a try.


There are  - loading -  comments....

My own thoughts on this are...By Scott Good on 03/02/2006 at 04:14 PM EST
Wow, Andrew. That is really interesting. I may need to slip you a few bucks at
some point to help us get something like yours set up here.

Thanks for posting it!

Scott Good
My own thoughts on this are...By Jerry Glover on 03/03/2006 at 12:30 AM EST
Great post Andrew; you definitely have pretty deep in the VoIP/Asterisk world.
How resource intensive is Asterisk? Is it something you can combine with other
functions on the same Linux box?
Depends alot on what you're doing - it can run small enough to be added to a linksys wifi router if you're using linux based firmware.....By Andrew Pollack on 03/03/2006 at 02:27 AM EST
The biggest factor is going to be processor use, and second to that network
bandwidth.

It comes down to the choice of codec. If all your inbound calls and outbout
calls are using the same codec, your processor is barely doing anything. If
some calls are on g.729 and others are on ulaw or gsm and the software has to
transcode live, it can be much more harsh. Some codecs are inherently more
processor intesive than others as well.

The box you're using which is my site is an Athlon 2600 single processor with
1gb of RAM and a pair of ide drives. Its at Serverbeach so its on a hellacious
network. Its running 2 Domino partitions, ASSP (perl based multithreaded
antispam proxy), Asterisk, SQUID, and a host of other linux based utility level
applications.

I start Asterisk with the 'nice' command in Linux and give it a higher priority
by 20% than the Domino processes so when things get tight, Asterisk gets time
it needs. It seems smooth enough for now, though when I go to production for
my fire department application and have a dozen Asterisk calls at once going
on, I'll probably need a little more box. A dual processor machine would go a
LONG way toward best in class operation.
My own thoughts on this are...By Jerry Glover on 03/06/2006 at 10:54 PM EST
Gotcha, thanks for the detail. One more question (which I suppose I could find
on own, but hey! you're an expert now). Does Asterisk have a voicemail
component?
Maybe I haven't made clear what Asterisk is...By Andrew Pollack on 03/06/2006 at 11:26 PM EST
..."Does it have one" -- its got, built in, a best in class voicemail
application. It also has all the scripting commands, menu commands, api's,
external interfaces, database hooks, reporting tools, recording tools, queue
management tools and any other thing you can think of.

It can be a top notch IVR system. It can easily be an ACD system for a help
desk. Music on hold, call routing, etc. is all pretty trivial with it.

You can create and be your own Vonage with it. Many companies do. I don't
know what Vonage itself uses, but you COULD do that with Asterisk.
I assumed so, but had to askBy Jerry Glover on 03/08/2006 at 12:26 AM EST
Sounds like quite a nice package. I assume you've already got it routing vmail
messages to your "universal inbox" in Notes. :-)


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