How do you choose the right phone system and service for your business?
If you search the Internet for answers, you'll find there's a confusing array of choices, and a lot of outdated information. There is no comprehensive directory of providers. There are a few
comparison charts, each with affiliate links to a handful of services. There's also a lot of jargon. Before you go searching, it's helpful to know the basics.
Traditional telephone service is circuit-switched, meaning there is a dedicated line (circuit) for each call, with a high level of quality we have come to expect. Internet based telephone service, or
VoIP (Voice over Internet Protocol,) is packet-switched, sending little packets of information over a broadband Internet connection to be reassembled at the other end. Sometimes packets are lost or
delayed. The quality of your calls will be affected not only by the VoIP service itself, but also the quality of your own router, modem and Internet provider.
A circuit-switched business telephone system might be a simple key system (phones with buttons or "keys" that you push to choose a line for your call,) a PBX, or Centrex (a service provided
by the local telephone company to simulate a PBX.) Before the introduction of hybrid systems and VoIP, a small business would have a key system, a medium sized business would have Centrex, and a
large business would have a PBX.
A PBX (Private Branch Exchange) is a telephone exchange, essentially an automated switchboard, that makes connections between the internal phones of an organization, and also connects it to the
outside world via the PSTN, or public switched telephone network. With Centrex, all that switching happens at the phone company's central office. A Hybrid keyphone system may combine a keyphone
with some PBX features such as direct dialing an extension in the company without going though the public network.
A premise-based PBX is hardware that is installed and maintained on the user company's premises. A PBX offers many advanced calling features, but it is an expensive piece of equipment, and
installation and maintenance usually requires technicians specifically trained on that particular system. IP PBX uses software instead of hardware and delivers voice and video with VoIP. A
self-hosted system is on the user company's premises.
A hosted or virtual PBX is likely the best choice for a small business that needs the advanced features of a PBX without the expense. The PBX is installed and maintained on the premises of another
company that offers PBX services to many clients using VoIP. Features that were once available only with an expensive PBX system are made affordable for small businesses. Multiple extensions,
auto-attendant, voicemail, voicemail by email, conference calls, low cost international calls and internet fax are just some of the many features available.
Advantadges of VOiP
VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several
telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of
transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes,
plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional
system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.
Let's say that you and your friend both have service through a
VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched
1.You pick up the receiver, which sends a signal to the ATA.
2.The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
3.You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
4.The phone number data is sent in the form of a
request to your VoIP company's call processor. The call processor checks it to ensure that it's in a valid format.
5.The call processor determines to whom to map the phone number. In
mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your
friend's ATA, telling it to ask the connected phone to ring.
6.Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means
that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the
same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
7.You talk for a period of time. During the conversation, your system and your
friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that
you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
8.You finish talking and hang up the
9.When you hang up, the circuit is closed between your phone and the ATA.
10.The ATA sends a signal to the soft switch connecting the call, terminating the session.
The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. See the "Soft Switches" section to learn more.
Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately
gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging
technologies, there are certain hurdles that have to be overcome. We'll look at those in the next section.
The current Public Switched Telephone Network is a robust and fairly bulletproof system for delivering phone calls. Phones just work, and we've all come to depend on that. On the other hand,
computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people really panic when their e-mail goes down for 30 minutes. It's expected from time to time. On
the other hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN may lack in efficiency it more than makes up for in reliability. But the network that makes up the
Internet is far more complex and therefore functions within a far greater margin of error. What this all adds up to is one of the major flaws in VoIP: reliability.
- First of all, VoIP is dependant on wall power. Your current phone runs on phantom power that is provided over the line from the central office. Even if your power goes out, your phone (unless it
is a cordless) still works. With VoIP, no power means no phone. A stable power source must be created for VoIP.
- Another consideration is that many other systems in your home may be integrated into the phone line. Digital video recorders, digital subscription TV services and home security systems all use a
standard phone line to do their thing. There's currently no way to integrate these products with VoIP. The related industries are going to have to get together to make this work.
- Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP uses IP-addressed phone numbers, not NANP phone numbers. There's no way to associate a geographic location with
an IP address. So if the caller can't tell the 911 operator where he is located, then there's no way to know which call center to route the emergency call to and which EMS should respond.
To fix this, perhaps geographical information could somehow be integrated into the packets.
- Testing, Testing...
Wondering if your broadband connection could support VoIP service? Brix Network offers a way to test your Internet connection to see how well it works.
- Because VoIP uses an Internet connection, it's susceptible to all the hiccups normally associated with home broadband services. All of these factors affect call quality:latency, jitter and
packet loss. Phone conversations can become distorted, garbled or lost because of transmission errors. Some kind of stability in Internet data transfer needs to be guaranteed before VoIP could
truly replace traditional phones
- VoIP is susceptible to worms, viruses and hacking, although this is very rare and VoIP developers are working on VoIP encryption to counter this.
- Another issue associated with VoIP is having a phone system dependant on individual PCs of varying specifications and power. A call can be affected by processor drain. Let's say you are
chatting away on your softphone, and you decide to open a program that saps your processor. Quality loss will become immediately evident. In a worst case scenario, your system could crash in the
middle of an important call. In VoIP, all phone calls are subject to the limitations of normal computer issues.
One of the hurdles that was overcome some time ago was the conversion of the analog audio signal your phone receives into packets of data. How it is that analog audio is turned into packets for VoIP
transmission? The answer is codecs.
A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of VoIP.
Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample
into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one
continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:
- 64,000 times per second
- 32,000 times per second
- 8,000 times per second
A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.
Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the
most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states "if no
one is talking, don't send any data." The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It's Annex B in the
CS-ACELP algorithm that's responsible for that aspect of the VoIP call.
The codec works with the algorithm to convert and sort everything out, but it's not any good without knowing where to send the data. In VoIP, that task is handled by soft switches.
E.164 is the name given to the standard for the North American Numbering Plan (NANP). This is the numbering system that phone networks use to know where to route a call based on the dialed numbers. A
phone number is like an address:
313 = State
555 = City
1212 = Street address
The switches use "313" to route the phone call to the area code's region. The "555" prefix sends the call to a central office, and the network routes the call using the last
four digits, which are associated with a specific location. Based on that system, no matter where you're in the world, the number combination "(313) 555" always puts you in the same
central office, which has a switch that knows which phone is associated with "1212."
The challenge with VoIP is that IP-based networks don't read phone numbers based on NANP. They look for IP addresses, which look like this:
IP addresses correspond to a
particular device on the network like a computer, a router, a switch, a gateway or a telephone. However, IP addresses are not always static. They're assigned by a DHCP server on the network and
change with each new connection. VoIP's challenge is translating NANP phone numbers to IP addresses and then finding out the current IP address of the requested number. This mapping process is
handled by a central call processor running a soft switch.
The central call processor is hardware that runs a specialized database/mapping program called a soft switch. Think of the user and the phone or computer as one package -- man and machine. That
package is called the endpoint. The soft switch connects endpoints.
- Soft switches know:
Where the network's endpoint is
What phone number is associated with that endpoint
The endpoint's current IP address
We'll talk more about soft switches and protocols on the next page.