If you regularly make long-distance phone calls, chances
are you've already used IP telephony without even
knowing it. IP telephony, known in the industry as
Voice-over IP (VoIP), is the transmission of telephone
calls over a data network like one of the many networks that
make up the Internet.
While you probably have heard of VoIP, what you may not know
is that many traditional telephone
companies are already using it in the connections between
their regional offices.
This person is using a computer to talk to a
friend in another
In this edition of HowStuffWorks,
you'll learn about VoIP and the technology that makes it
possible. We'll talk about VoIP's major protocols, about the
various services provided and the low-cost, often free
software that allows you to take advantage of them.
But first, let's discuss the fundamental problem with
existing telephone networks -- namely, their reliance on
Circuit Switching Circuit switching is a
very basic concept that has been used by telephone
networks for over 100 years. What happens is that when a
call is made between two parties, the connection is maintained
for the entire duration of the call. Because you are
connecting two points in both directions, the connection is
called a circuit. This is the foundation of the
Public Switched Telephone Network (PSTN).
You pick up the receiver and listen for a dial tone.
This lets you know that you have a connection to the local
office of your telephone carrier.
You dial the number of the party you wish to talk to.
The call is routed through the switch at your
local carrier to the party you are calling.
A connection is made between your telephone and the
other party's line, opening the circuit.
You talk for a period of time and then hang up the
When you hang up, the circuit is closed, freeing your
Let's say that you talk for 10 minutes. During this time,
the circuit is continuously open between the two phones.
Telephone conversations over the traditional PSTN are
transmitted at a fixed rate of about 64 kilobits per second
(Kbps), or 1,024 bits per
second (bps), 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. So in
a 10-minute conversation, the total transmission is 9600 KB,
which is roughly equal to 9.4 megabytes
If you look at a typical phone conversation, much of this
transmitted data is wasted. While you are 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. 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.
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 looking at. Instead of simply sending and retrieving
data as you need it, the two computers involved in the
connection would pass data back and forth the whole time,
whether the data was useful or not. That's no way to set up an
efficient data network. Instead, data networks use a method
called packet switching.
Packet Switching While circuit switching
keeps the connection open and constant, packet switching opens
the connection just long enough to send a small chunk of data,
called a packet,
from one system to another. What happens is this: The sending
computer chops data into these small packets, with an address
on each one telling the network where to send them. When the
receiving computer gets the packets, it reassembles them into
the original data.
Packet switching is very efficient. It minimizes the time
that a connection is maintained between two systems, which
reduces the load on the network. It also frees up the two
computers communicating with each other so that they can
accept information from other computers as well.
Click "Play" to see how packet switching
VoIP technology uses this packet-switching method to
provide 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 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 your company had equipment installed and a
contract set up so that you can use VoIP. You have installed
about a dozen telephones and a digital private branch
exchange (PBX) in your office. A PBX is essentially a
switch used to connect a number of phones (extensions) to each
other and to one or more outside phone lines. In our example,
the PBX is also a gateway.
Gateways are used to connect devices on two different types
of networks so that they can communicate with each other. Our
PBX is a gateway because it converts the standard
circuit-switched signal from each phone into digital data that
can be sent over a packet-switched,
IP-based network. IP stands for "Internet
protocol," the language used by most data networks. Let's take
another look at that typical telephone call, but this time
using VoIP over a packet-switched network:
You pick up the receiver, which sends a signal to the
The PBX receives the signal and sends a dial tone. This
lets you know that you have a connection to the PBX.
You dial the number of the party you wish to talk to.
This number is then temporarily stored by the PBX.
Once you have entered the number, the PBX checks it to
ensure that it is in a valid format.
The PBX determines whom to map the number to. In
mapping, the number is attached to the IP address of
another device called the IP host. The IP host is
typically another digital PBX that is connected directly to
the phone system of the number you dialed. In some cases,
particularly if the party you are calling is using a
computer-based VoIP client, the IP host is the system you
wish to connect with.
A session is established between your company's
PBX and the other party's IP host. This means that each
system knows to expect packets of data from the other
system. Each system must use the same protocol to
communicate. The systems will implement two channels, one
for each direction, as part of the session.
You talk for a period of time. During the conversation,
your company's PBX and the other party's IP host transmit
packets back and forth when there is data to be sent. The
PBX at your end keeps the circuit open between itself and
your phone extension while it forwards packets to and from
the IP host at the other end.
You finish talking and hang up the receiver.
When you hang up, the circuit is closed between your
phone and the PBX, freeing your line.
The PBX sends a signal to the IP host of the party you
called that it is terminating the session. The IP host
terminates the session at its end, too.
Once the session is terminated, the PBX removes the
number-to-IP-host mapping from memory.
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. Of course, having the ability to communicate and
understanding the methods of communication are two very
different things. For telephones to communicate with each
other and with other devices, such as computers, over a data
network, they need to speak a common language called a
Protocols There are two major protocols
being used for VoIP. Both protocols define ways for devices to
connect to each other using VoIP. Also, they include
specifications for audio codecs. A codec, which stands
for coder-decoder, converts an audio signal into
a compressed digital form for transmission and back into an
uncompressed audio signal for replay.
The first protocol is H.323, a standard created by
the International Telecommunications Union (ITU). H.323
is a comprehensive and very complex protocol. It provides
specifications for real-time, interactive videoconferencing,
data sharing and audio applications such as IP telephony.
Actually a suite of protocols, H.323 incorporates many
individual protocols that have been developed for specific
As you can see, full implementation of H.323 requires a lot
of overhead. This
page provides detailed information about the entire H.323
suite of protocols and how they relate to the OSI Reference
An alternative to H.323 emerged with the development of
Session Initiation Protocol (SIP) under the auspices of
the Internet Engineering Task Force (IETF). SIP is a
much more streamlined protocol, developed specifically for IP
telephony. Smaller and more efficient than H.323, SIP takes
advantage of existing protocols to handle certain parts of the
process. For example, Media Gateway Control Protocol
(MGCP) is used by SIP to establish a gateway connecting to
the PSTN system. You can learn more about the architecture of
SIP on this
Let's take a quick look at the various ways you can connect
Calling There are four ways that you might
talk to someone using VoIP. If you've got a computer or a
telephone, you can use at least one of these methods without
buying any new equipment:
Computer-to-computer - This is certainly the
easiest way to use VoIP. You don't even have to pay for
long-distance calls. There are several companies offering
free or very low-cost software that you can use for this
type of VoIP. All you need is the software, a microphone,
card and an Internet
connection, preferably a fast one like you would get
through a cable
modem. Except for your normal monthly ISP fee, there is
usually no charge for computer-to-computer calls, no matter
the distance. A good example of this software is MSN
The Net2Phone software client is easy to
set up and
Computer-to-telephone - This method allows you to
call anyone (who has a phone) from your computer. Like
computer-to-computer calling, it requires a software client.
The software is typically free, but the calls may have a
small per-minute charge. For example, Net2Phone
offers free calls to anywhere in the United States for the
first five minutes. If the call is over five minutes, a rate
of 3.9 cents per minute kicks in. Net2Phone's international
rates vary widely, ranging from 3.9 cents to $7.52 per
minute, depending on where you call.
Telephone-to-computer - A few companies are
providing special numbers or calling cards that allow a
standard telephone user to initiate a call to a computer
user. The caveat is that the computer user must have the
vendor's software installed and running on his or her
computer. The good news is that the cost of the call is
normally much cheaper than a traditional long-distance call.
Telephone-to-telephone - Through the use of
gateways, you can connect directly with any other standard
telephone in the world. To use the discounted services
offered by several companies, you must call in to one of
their gateways. Then, you enter the number you wish to call,
and they connect you through their IP-based network. The
downside is that you have to call a special number first.
The upside is that the rates are typically much lower than
standard long distance.
Although it will take some time to happen, you can be sure
that, eventually, all of the circuit-switched networks will be
replaced with packet-switching technology. 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.