A critical piece of the VoIP puzzle is your connection to the outside world.

Before VoIP, the phone company controlled how your telecom services got to your site. They owned what is commonly termed The Last Mile—all the infrastructure right to your door. If you wanted to connect to the Public Switched Telephone Network (PSTN), you had to go through them.

Traditional connection to PSTN


While you can still order traditional PSTN circuits, there’s hardly any point these days. The theoretical benefits of legacy circuits are seldom realized in practice. In fact, even old-school PSTN circuits are as often as not delivered across a VoIP connection.

These days, VoIP is almost always the way to go. You connect to a VoIP carrier across your internet connection, and they in turn have switching equipment that they tie into the PSTN at strategic points (this is normally called termination, since they are connecting your VoIP connection to the PSTN at the termination point). VoIP providers make their money because they can negotiate volume discounts, as well as carry toll traffic on their own network.

VoIP Topology in the WAN


Your VoIP carrier will provide some or all of your telephone connectivity to the PSTN. This could be in the form of a DID and termination provider who will provide connectivity to your existing PBX, or it could be a fully-hosted system that handles all of the call control for external and internal calls, with only VoIP telephones  at your site. Regardless of the VoIP service you select, the quality of the internet connection between you and your VoIP provider will be critical.

Fortunately, since most ISPs these days also offer some sort of VoIP service, they generally do a good job of ensuring all voice traffic is carried properly on their network. In addition to this, carriers generally do a good job of interconnecting between each other as well. This is unfortunately not always the case, so be sure you use google to get a feel for how well your ISP handles VoIP quality issues. There’s lots of competition, so you hold the cards so long as you’re willing to do the research.

In some cases, you may require a premium network connection to your carrier. Generally this will be required when there is poor internet service in your area, or where your ISP does not have strong competition and they have allowed their network to degrade. In this case you will need to add the cost of an enhanced network connection to the cost of your VoIP service, and perhaps compare that to the cost of a more traditional telephone service such as PRI. (note that many PRI providers actually deliver the PRI across a VoIP connection, so make sure you get it in writing what they are commiting to deliver in terms of quality).

Your VoIP provider should be able to work with you, your network team and your ISP to ensure the internet connection at all your facilities is able to handle the job of carrying your VoIP traffic. In many cases a test or trial can be set up to validate the quality prior to committing all of your telephone infrastructure to the new solution.

If you are not sure whether your internet connection is up to the task, get in touch with us. We have tools that can help to evaluate the suitability of your connection.


You probably have an internal network, but if it’s going to be used for voice, a few requirements need to be considered.

Voice actually doesn’t require a lot of network bandwidth (we’ve successfully implemented VoIP phones on CAT3 cabling). A telephone call uses less than 100kbps of bandwidth, therefore a typical network connection will provide one thousand times or more bandwidth than the voice requires. What makes voice tricky on an internal network is that it has no tolerance for delayed or lost packets. Since a basic network treats all traffic as equal, a massive data transfer on the network can push the voice traffic into a situation where it’s waiting its turn. This, for users, will appear to be a system fault, as parts of what are being said end up in a buffer, or perhaps even dropped completely. The bottom line is that while voice doesn’t need a lot of bandwidth, it sends and recieves an uninterrupted stream of little packets, and pretty much every single one must be delivered, on time and in order.

Another consideration that typically comes up when evaluating a network for voice is how to provide power to the phones. You can buy a little wall plug and see if there’s a spare spot on a power bar (under every single desk in the office!), or you can deploy Power over Ethernet, which will deliver the power right through the network plug. PoE allows you to power the phones from the network switch, which will usually save you money, through not having to buy dozens of little power supplies and power bars. PoE also allows you to centralize a single source of redundant power for your phones. Connect the PoE switch to your UPS and your phones can stay on when the lights go out.

There are several strategies you can follow to address the networking needs of your VoIP project, but I would say the details all boil down to one of two approaches: 1) Build a converged network including QoS and VLANs, or 2) Install a completely separate physical network, dedicated to just the phones with simple, small-business-grade unmanaged PoE switches.

Converged Network

In a converged network, all devices on the network are managed as part of a unified, managed whole. Intelligence within the network itself determines how various devices are handled.

This method is generally sold as the technically superior way to deploy a network, but it also will tend to cost more to implement and maintain.

Larger enterprises will tend to have converged, managed networks. Smaller organizations will tend towards a simpler design.

Pros and Cons of a Converged Network

Software-based management More expensive hardware
Enterprise-class architecture Higher level of technical skill required
Single cable can serve PC and phone Higher administrative overhead
Theoretically less need to move plugs More difficult for office admin staff to understand
Managed network more likely to stay neat

Generally-speaking, you’re going to look to implement a converged network if you have a highly-skilled (and factory-certified) network team, and a budget that allows for the purchase of high-end network hardware.

Running A Completely Separate Voice LAN

A simpler way to deliver a voice network is to build an entirely separate network to handle the phones. Computer traffic stays off the phone network, and voice traffic never enters the data network.

While at first this might seem like an expensive method, the reality is that in many cases it proves to be a far less expensive approach. Another often overlooked advantage of this strategy is that it is usually easier to explain to non-technical administrative staff. In my experience, this factor alone can be the deciding factor in favour of a stand-alone network.

Pros and Cons of a Stand-Alone Network

Less expensive hardware More cabling required (two runs per station)
Easier to understand Separate voice and data plugs
Expensive technician not always required Phones cannot run on any port
PoE hardware can be less expensive Hodge-podge of network gear over time
Existing data equipment does not need to be touched or upgraded


The decision as to what your LAN needs in order to deploy VoIP is dependent on many factors. In general, we have found that smaller companies tend to prefer a dedicated LAN for voice, whereas larger organizations tend towards a converged network.

When evaluating the network for your IP Telephony deployment, you need to be aware of some of the differences between a hosted PBX and an onsite-PBX.

In order to simplify this, we will break the network down into two separate concepts: The Local, onsite network (or LAN), and the wide-area, external network (or WAN, typically provided across the Internet).


Hosted PBX

In the case of a hosted PBX, all of the PBX intelligence will be located offsite. The only things located at the site will be the telephone sets, and the network components required to connect those sets through the internet to the hosted system.

Advantages of a Hosted PBX

  • No server onsite to maintain
  • Higher-quality server environment (redundant power, HVAC, internet, etc)
  • Software updates usually included
  • OPEX generally more popular with C-level execs
  • Onsite requirements can usually be handled by network team

Disadvantages of a Hosted PBX

  • Recurring costs – more expensive in the long term
  • Bandwidth requirements – desk-to-desk calls still have to pass through data center


On-site PBX

An on-site PBX provides all the services to the phone sets from the premises. Sets do not even require internet access, as all their requirements (provisioning files, time and date, firmware updates, security) can be handled through the PBX system.

Advantages of an On-site PBX

  • Once it’s paid for, it only needs to be maintained
  • Physical control of hardware
  • Easier termination of legacy PSTN circuits
  • Typically lower long-term costs
  • Lower bandwidth requirements

Disadvantages of an On-site PBX

  • Requires available technical team to maintain system
  • Server environment may require construction and other costs
  • CAPEX not always an easy sell to C-level execs


In both cases—Hosted or On-site—the underlying technologies are essentially the same. We use the LAN to replace the traditional telephone wiring, and the WAN to replace the traditional carrier circuit (PRI, POTS lines, etc). For hosted, we additionally handle PBX connections from the sets in the WAN as well.

If you have any questions about making sense of IP Telephony, please feel free to reach out and speak to us.

The next blog in this series will look at the onsite network environment, and the considerations for a Converged versus Dedicated LAN.

At its simplest, IP Telephony is just a way to carry voice from one place to another. What makes it so revolutionary is not so much due to any technical advantages, but simply that it is based on inexpensive and popular networking standards. You can buy the hardware at your local computer store. The details of the SIP protocol can be downloaded for free from the IETF, and IP networks can be built by anyone, at very low cost.

IP Telephony began as an experiment in carrying voice traffic across a packet network. The concept was simple: Run some voice traffic across a fixed-cost connection (piggy back it onto the data traffic), and avoid paying long distance charges between offices. Early equipment was expensive, but then again, so were long distance charges.

As the internet grew in popularity in the late 90s, the powerful (and free) Internet Protocols (IP) began to rapidly replace the almost ubiquitous Novell NetWare technology then popular in office networks (LANs), and it also became popular to layer IP on top of the various carrier circuits in use. The rise of the ISP began at this point as well. IP rapidly found use all over the place as a standardized all-purpose network protocol. Every network was becoming an IP network, so the awkward term “packetized voice” naturally became “IP Telephony” or “Voice over IP (VoIP)”.

A massive disruption has ocurred in the telecommunications industry. Old, complex, proprietary and obscure technologies have given way to well-documented, open-standards and popular protocols.

The challenge with IP Telephony is that the internet was not designed to carry voice traffic. The neat little packets being delivered all in a neat row, can too often end up more like this:

How Packets Sometimes Get Delivered

Human beings are not computers. When we talk, our brain expects to hear speech in a very specific way. It must be contiguous (no gaps), it must arrive in order (or sense make won’t it else), and any delay of more than a few hundred milliseconds is going to feel rude (“are you even paying attention?”). Computers can forgive all sorts of problems with communication, but the human ear is a bit more fussy.

In the early days of the internet, it was all but impossible to expect voice to pass reliably. Today, the opposite is true. IP Telephony is reliable, and more importantly perhaps, it is understood in the industry that everyone maintaining networks have a shared responsibility to ensure voice traffic passes without problems. In other words, current-generation networks are engineered to ensure voice quality.

Today, rather than purchasing expensive hardware to connect to limited physical circuits, IP telephony allows simple and reliable connections of multiple service types, all on a standard network server, running standard network hardware.

IP Telephony saves money, and opens the door to the full power of the internet. In fact, I think it would be safe to say the best is yet to come!

Next up we’ll dive into the details of what is required in an office network to support IP Telephony. In other words, how to VoIP enable your network.

The technologies that run the internet were first developed in the 60s.

The network could be built as a mesh of nodes, and information could be sent in more than one direction—routing. Also, connections between nodes could be shared with all the types of traffic passing through. This is analogous to the highway networks of today. Many vehicles can be on the same stretch of road at the same time, and can be of different sizes and speeds, and can be coming from different places and going to different destinations, starting and ending at different times. It’s a bit more messy than a neatly-arranged and carefully-scheduled train, but it’s far less expensive, and the infrastructure is not sitting doing nothing in between trains.

A key difference between circuit-switched and packet-switched networks is that packet-switched data must be encapsulated in a wrapper containing address information. The network doesn’t really care what’s in the package, so long as the addressing information is intact. In the train-to-road analogy, each car needs a driver, but an entire train follows one locomotive.

Circuit-Switched (Public Telephone Network)

Packet Switched (Internet)

Your computer encapsulates the data you want to send in a packet, and if there is more data to send than can fit in that packet (there are limits on how large a packet can be), it will send another, and another … as many packets as needed. At the other end, all the packages are opened, the data inside is recombined, and the other end then has a photo, or document, or video, or snippet of sound, or whatever it was you sent. Often, missing packets can be retransmitted, and packets can take multiple routes to reach their destination.

Circuit-switched networks have many benefits over packet-switched networks, however it is difficult to justify their cost, as they tend to be limited in their use. An analogy would be the high-speed passenger rail networks of Europe. You cannot carry freight, and you will never be able to justify the cost to build a station next to your house. Roads, on the other hand, are far more limited in terms of speed and efficiency, but they can be built everywhere, and can carry everything from bicycles to trucks to buses.

Humans love to talk. Any discussion of IP Telephony needs to always keep first and foremost the understanding that the primary purpose for telephones is to allow us to talk to each other. It seems obvious, but this simple fact sometimes gets lost in technical details.

Our vocal cords, tongue, lips and teeth allow us to create complex vibrations in the air around us, which when recieved by another person’s ear (and interpreted by their brain) allows the transfer of information from us to them.

The essential parts of the telephone were figured out through experimenting with ways to convert sound vibrations in the air into electrical signals, and then convert those electrical signals back into sound vibrations at the other end. Electrical signals have an advantage in that they can travel over vast distances, and do so far more quickly than sound waves can.

The first telephones were fascinating, but not very useful. You had to string a wire between you and the other end, hook up a battery, and talk really loudly.

Then, somebody got clever and realized if you wire everybody’s line into a central office and hire a bunch of people to sit at that central location and answer calls, you could power everything from there, and have those operators move plugs around to connect people together. This system was incredibly labour-intensive, and also prone to all sorts of errors and ‘human’ factors, but without it the telephone might never have been anything more than an interesting science experiment.

The first automatic telephone switch was invented by an undertaker, who believed that one of his town’s switchboard operators was sending all calls for ‘the undertaker’ exclusively to her husband, his competitor. The Step-by-step Switch revolutionized the industry by automating the switching of connections between the two ends of a circuit, with you doing the work of selecting the circuit you wanted using the dial on your phone.

For a hundred years telecommunications took place across this circuit-switched network. The circuit created every time you made a phone call was dedicated to your call, and nothing else. The closest analogy I can think of is that of a railroad network. All the switches and signals that have to line up just to support one train, and no other trains are permitted on those sections of rail until that train has passed. Running two trains at the same time requires two sets of tracks.

In the next article in the series we’ll talk about The Emergence Of Data Networks, which are essential to IP Telephony, as they are based on packet-switching rather than circuit switching.

IP Telephony—also referred to as Voice over IP (or simply VoIP)—is packet-switched voice, rather than the circuit-switched voice of decades past.

Stay tuned for the next article in the series: IP Telephony – The Emergence Of Data Networks

The best phone systems for small business operations are generally those systems which offer lots of configuration flexibility (i.e. lots of useful features to choose from), but can be delivered as a simple solution.

Traditionally, phone systems for smaller businesses lacked the features available to larger businesses (which of course also had larger budgets). This was typically the case because system features were often tied to hardware components (for example voicemail was usually a completely separate machine), and smaller systems simply did not have enough computing power to provide more than a basic set of features.

The open source revolution in telecom, spearheaded by the Asterisk project—but complemented by many other projects such as FreePBX, OpenSIPS, FreeSwitch, Kamailio, and so forth—drove the cost of telecommunication systems down, and raised the bar on what sort of features could be delivered on a computer-based platform (as opposed to the old, proprietary hardware-based platforms). Suddenly, it was possible to download a complete PBX in software, for free, and get features that were unheard of in even the most expensive systems only a few years previous. These new open source PBXs rapidly gained popularity, as many felt they were the best phone systems for small business operations.

The challenge, unfortunately, was that these new innovations didn’t change the fact that the design and implementation of the system still required either a) an extensive skillset, or b) the time and curiousity needed to achive a result based on trial-and-error.

The best phone systems for small business operations are not so much based on hardware and software as they are based on the team that ensures their success.

Find out what happens when a rich set of features is combined with decades of experience in the design, implementation and support of business telecommunication systems. Your telecommunications challenges can be solved. We can help.

If one asks the question: What is a PBX System? One will typically get the answer: A Private Branch Exchange. What is more important of course is understanding what it does. The PBX emerged from a desire of businesses to reduce the cost of telephone services, coupled with a need for more control over how phone services worked in their business.

Prior to the PBX, every phone in an office required a separate circuit from the phone company (in exactly the same way as home telephone service is delivered)— one pair of wires from the Central Office to each and every desk that needed a phone. Since many (or perhaps most) of these phones would be unused most of the time, and since many (or perhaps most) calls happened between phones in the office, it became obvious that an office of, say, 100 people might only need a dozen or so lines to the outside world.

The PBX answered this need by connecting trunks from the Central Office into the premises, and then provided extensions to each desk. When a call was placed, the PBX would determine whether it was a desk-to-desk call, or an external call, and route accordingly.

For incoming calls, the PBX requires intelligence in order to determine what needs to happen to each telephone number as it arrives in the system. This is perhaps the most complex part of PBX design, however in essence what we are doing is answering the question “When somebody calls this number, what experience do we want them to have?”

The question of “What is a PBX System?” (now a hosted PBX system) sometimes also includes mention of the KTS, or Key Telephone System. These were generally used in smaller companies (a handful of lines and maybe a dozen phones). In a Key System, the incoming lines appeared on buttons on each phone (sometimes called lamps because each button had a light under it). Each phone could see each line. This obviously would not scale well as a phone can only have so many buttons on it. As electronics improved, Key Systems often provided some of the line pooling features of their bigger PBX siblings, and most of the larger, better quality PBXs are also able to emulate a small Key System (which might be useful for example in an executive assistant or departmental reception) by providing a Shared Line Appearance (SLA) feature.

So, what is a PBX system? Well, these days, it’s something far more than just a phone system from the past, and with new, standards-based and open-source platforms underpinning many PBX products, we’re only beginning to scratch the surface of what is possible.

If you want to learn more, call 1-877-267-3835 for a free consultation (I’ll be happy to get on the call if you’d like), or check out our Hosted PBX Business Telephone Systems page.

In its early days, VoIP was touted for many reasons, but quality was seldom one of them. VoIP was traditionally seen as a riskier technology, with near-certain problems with quality and reliability. Land lines (especially digital circuits such as PRI) were so reliable that one could reasonably expect decades of trouble-free service from them.

Carriers would put their brightest and best on the PRI side of the business; that’s where all the big customers were, and that’s where the money was to be made. Today, the old-school technical people are retiring, and the newcomers to the industry have little or no interest in obsolete technologies such as PRI.

Today, I find I can no longer recommend PRI service as the best choice for on-site trunking. The reason is not so much technical as it is cultural: Younger, career-minded people rising through the ranks in this industry cut their teeth on IP-based technologies, and old-school legacy voice is neither interesting to them, nor in many cases even comprehensible.

In the very recent past, the best way to ensure trouble free PSTN connectivity was PRI. Today, that is no longer a safe assumption. As the carriers work to cut costs, and as the decision-makers in these organizations are people with a career history of learning and working with VoIP, the older, more expensive technologies have become little more than peripherals, reluctantly installed to deliver backwards-compatibility. In many cases if you order a PRI circut, what you’ll find is delivered is a SIP-based gateway, providing VoIP the whole last mile to your equipment room, with a 6 foot cable of PRI into your phone system. If your PBX is VoIP-capable, you might as well cut out the 6 feet of PRI and go SIP the whole way.

Note that this is more of a trend that I’ve been seeing rather than a hard rule that applies in all situations. Still, I have seen increasing problems with PRI, and decreasing problems with VoIP (not just with the technology, but also with the people involved in providing it). The tipping point was already several years ago, and whereas five years ago I would have always recommended PRI, today, I very seldom do. My experience is that the best and brightest people on the carrier side of things are working in the VoIP side of the house, and the PRI side of the business is mostly old timers nearing retirement, or folks who do not seem to understand the technology very well.

The bottom line is that while a PRI circuit was always a very low-risk way to deliver trunking to your premises, that is no longer a certainty. The best minds are working in VoIP now; from product development, through solution engineering, to installation and support.

I’ve never been much of a fan of videoconferencing.

While shows like Star Trek have set the expectation in us that ‘in the future, this is how people will communicate’, I don’t see any signs that we’ll ever be replacing all phones (or travel) with video.

I have noted two problems with videoconferencing in the past, and my thoughts about that haven’t changed one whit.

Today, I read an article that I thought presented an interesting perspective. The thought is that the marketing efforts for video products is having to resort to FUD to pitch their technology. It’s an interesting perspective.

Last week I had a conference call with a customer, and they were showing off their new Lync deployment complete with videoconferencing. The video did not facilitate the discussion in any way, and in fact since not all of us were feeding video, those with cameras on them ended up looking rather awkward when they were not speaking, as they had to figure out what to do with their faces even though they were peripheral to that part of the discussion. It felt forced and awkward.

Even in Star Trek, the videoconferencing was mostly reserved for formal conversations in the main bridge area (and had full eye-to-eye contact with everyone dressed in their best uniform). Communications between shipmates was all done by voice only.