How VoIP Works

Chances are good you're already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the long haul. Once the call is received by a gateway on the other side of the call, it's decompressed, reassembled and routed to a local circuit switch.

Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.

With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there's a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.

Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.

Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:

There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:

With many VoIP services, you can also check voice mail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you're interested, it's best to do a little shopping.

Now that we've looked at VoIP in a general sense, let's look more closely at the components that make the system work. To understand how VoIP really works and why it's an improvement over the traditional phone system, it helps to first understand how a traditional phone system works.

Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching.

Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. When a call is made between two parties, the connection is maintained for the duration of the call. Because you're connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN).

Here's how a typical telephone call works:

Let's say you talk for 10 minutes. During this time, the circuit is continuously open between the two phones. In the early phone system, up until 1960 or so, every call had to have a dedicated wire stretching from one end of the call to the other for the duration of the call. So if you were in New York and you wanted to call Los Angeles, the switches between New York and Los Angeles would connect pieces of copper wire all the way across the United States. You would use all those pieces of wire just for your call for the full 10 minutes. You paid a lot for the call, because you actually owned a 3,000-mile-long copper wire for 10 minutes.

Telephone conversations over today's traditional phone network are somewhat more efficient and they cost a lot less. Your voice is digitized, and your voice along with thousands of others can be combined onto a single fiber optic cable for much of the journey (there's still a dedicated piece of copper wire going into your house, though). These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open, and 960 KB every minute it's open. In a 10-minute conversation, the total transmission is 9,600 KB, which is roughly equal to 10 megabytes (check out How Bits and Bytes Work to learn about these conversions). If you look at a typical phone conversation, much of this transmitted data is wasted.

Next we'll talk about packet switching.

A packet-switched phone network is the alternative to circuit switching. It works like this: While you're talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant amount of the time in most conversations is dead air -- for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), what if we sent just the packets of noisy bytes when you created them?

Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.

While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.

Next, we'll look at the advantages of using 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 network:

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.

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:

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:

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:

We'll talk more about soft switches and protocols next.

The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.

Soft switches work in tandem with network devices to make VoIP possible. For all these devices to work together, they must communicate in the same way. This communication is one of the most important aspects that will have to be refined for VoIP to take off.

Protocols

As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls. How do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is protocols.

There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols, H.323 incorporates many individual protocols that have been developed for specific applications.

H.323 Protocol Suite

Video

Audio

Data

Transport

As you can see, H.323 is a large collection of protocols and specifications. That's what allows it to be used for so many applications. The problem with H.323 is that it's not specifically tailored to VoIP.

An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waiting. You can learn more about the architecture of these protocols at Protocols.com: Voice Over IP.

One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.

VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it's clear that developers will keep refining this technology until it eventually replaces the current phone system.

VoIP has its distinct advantages and disadvantages. The greatest advantage of VoIP is price and the greatest disadvantage is call quality. For businesses who deploy VoIP phone networks -- particularly those who operate busy call centers (customer service, tech support, telemarketing, et cetera) -- call quality issues are both inevitable and unacceptable. To analyze and fix call quality issues, most of these businesses use a technique called VoIP call monitoring.

VoIP call monitoring, also known as quality monitoring (QM), uses hardware and software solutions to test, analyze and rate the overall quality of calls made over a VoIP phone network [source: ManageEngine]. Call monitoring is a key component of a business's overall quality of service (QoS) plan.

Call monitoring hardware and software uses various mathematical algorithms to measure the quality of a VoIP call and generate a score. The most common score is called the mean opinion score (MOS). The MOS is measured on a scale of one to five, although 4.4 is technically the highest score possible on a VoIP network [source: TestYourVoIP.com]. An MOS of 3.5 or above is considered a "good call" [source: ManageEngine].

To come up with the MOS, call monitoring hardware and software analyzes several different call quality parameters, the most common being:

There are two different types of call monitoring: active and passive. Active (or subjective) call monitoring happens before a company deploys its VoIP network. Active monitoring is often done by equipment manufacturers and network specialists who use a company's VoIP network exclusively for testing purposes [source: VoIP Troubleshooter.com]. Active testing can't occur once a VoIP network is deployed and employees are already using the system.

Passive call monitoring analyzes VoIP calls in real-time while they're being made by actual users [source: VoIP Troubleshooter.com]. Passive call monitoring can detect network traffic problems, buffer overloads and other glitches that network administrators can fix in network down time.

Another method for call monitoring is recording VoIP phone calls for later analysis. This type of analysis is limited, however, to what can be heard during the call, not what's happening on the actual network. This type of monitoring is usually done by human beings, not computers, and is called quality assurance.

Next we'll talk about making VoIP calls using cell phones.

Dual-mode cell phones contain both a regular cellular radio and a Wi-Fi (802.11 b/g) radio. The Wi-Fi radio enables the cell phone to connect to a wireless Internet network through a wireless router. If you have a wireless Internet router in your home, or if you're sitting at a Starbucks with wireless Internet access, you can use your cell phone to make VoIP calls. Here's how it works:

Similar to dual-mode cell phones are Wi-Fi phones. Wi-Fi phones aren't technically cell phones because they only have a Wi-Fi radio, not a cellular radio. Wi-Fi phones look like cell phones (small, lightweight handsets), but can only make calls when connected to a wireless Internet network. That means all Wi-Fi phone calls are VoIP calls.

Wi-Fi phones are useful in large companies and offices with their own extensive wireless networks. And could prove to be the next big thing, with the expanding market for municipal Wi-Fi. [source: Dr. Dobb's Portal]. Imagine that your entire city was covered by a high-speed wireless network. That means cheap (if not free) VoIP calls wherever you go.

In England, a company called Hutchinson 3G (or simply 3) has partnered with the popular VoIP service Skype to introduce the 3 Skypephone. The Skypephone allows users to make free cell phone calls to other Skype users. The phone can also make regular cell-phone calls to non-Skype users for the normal fees. Here's how it works:

The 3 Skypephone isn't currently available in the United States.

Think of amateur radio, or ham radio, as an early version of the Internet. Using a worldwide network of radio towers, antennas and transceivers, amateur radio enthusiasts are able to communicate with fellow hobbyists around the globe, sometimes by voice and sometimes by Morse code.

Amateur radio is limited by the distance that radio waves can travel. To send a signal to the other side of the world requires calculated timing and more than a little bit of luck. Every 11 years, for example, there's a peak in the number of sunspots produced by the sun, which increases the intensity of something called ionospheric propagation [source: International Solar Terrestrial Physics Program]. By bouncing radio signals high into the ionosphere, ham radio users can send long-distance messages. During off-peak years it's much more difficult.

Now amateur radio fans are using VoIP technology to link users around the globe. Here's how it works. Ham radio has always relied on FM repeaters, large radio towers that act as base stations for accessing the radio network from home. By attaching an Internet-connected PC to these repeater stations, people can communicate with the repeater using VoIP.

Several amateur radio fans have developed special software that helps connect home radio transceivers to the Internet. Users can connect their ham radio transceivers to their PC sound card and use the computer software to search for available repeater stations across the world [source: ARRL]. No longer are ham radio fans limited to the closest repeater station. If you live in Indiana, you can call into a repeater station in Mozambique and chat with local amateur radio aficionados instantly.

There are also software programs that allow you to communicate with other amateur radio users directly from your PC, without having an actual ham radio [source: ARRL]. Some ham radio purists wouldn't call this amateur radio, while others hope that this new technology will draw more young people into the hobby.

For more information about VoIP, amateur radio and related topics, check out the links that follow.