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
state.
In this article, 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
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 receiver.
When you hang up, the circuit is closed, freeing
your line.
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
(MB).
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 works.
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 PBX.
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 protocol.
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 applications.
As you can see, full implementation of H.323
requires a lot of overhead. Protocols.com:
H.323 provides detailed information about
the entire H.323 suite of protocols and how they
relate to the OSI
Reference Model.
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
at Protocols.com:
SIP.
Let's take a quick look at the various ways
you can connect using VoIP.
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,
speakers,
a sound
card and an Internet connection, preferably
a fast one like you would get through a cable
or DSL
modem. Except for your normal monthly ISP
fee, there is usually no charge for computer-to-computer
calls, no matter the distance.
The Net2Phone software
client is easy to set up and use.
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.
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.
