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What is VoIP? .. Explaining Voice Over IP
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Simply put, voice over IP (VoIP)
technology, or IP telephony, as it
is often called, is a
system for transmitting telephone
calls over data networks,
such as the ones that make up the
Internet.
While VoIP technology is set to
revolutionize communications, and
is already being used by a number
of traditional telephone companies
to connect their regional offices,
on a smaller scale it can also be
a useful solution for businesses
looking to trim their telephone
expenses.
The advantages of using VoIP
technology are simple: its use can
result in huge savings on the
amount of physical and resources
required to communicate by voice
over long distances. It does so by
working around circuit switching
architecture, one of the
fundamental drawbacks of
traditional telephone networks.
Traditional circuit
switching-based telephone networks
operate by opening a circuit
between two points, identified by
their telephone numbers. This
circuit remains open, and
transferring at its full capacity
for the duration of the call,
until somebody disconnects it by
hanging up. Much of this capacity
is wasted during a normal
telephone conversation, because
while the line is working at full
capacity, not all of each user’s
time is spent transferring data,
or talking. Normal telephone
users, of course, spend much of
their time listening, or receiving
data. Furthermore, during the
course of a normal phone call,
there is often dead air. All of
these things are wasted capacity.
Data networks operate in an
entirely different way. They
communicate through packet
switching, a much more efficient
scheme for exchanging data.
Instead of keeping a circuit open
constantly, they send and receive
data only as needed, a bit at a
time, in data packets. By doing
so, packet switching-based data
networks free up network
resources, as well as the
resources of the computers sending
and receiving information.
VoIP technology uses packet
switching to minimize the amount
of resources used in a telephone
connection by exchanging the
information in packets over a data
network. This allows several phone
calls to use the space that just
one call would have occupied in a
circuit-switched network.
In the case of an office,
telephones might be connected to a
private branch exchange (PBX), a
device designed to connect a
number of phones or extensions to
an outside line. Using a gateway,
a device used to translate the
standard circuit-switched signal
generated by the telephones into
digital information that can be
sent over the data network. This
signal is usually an IP signal,
the standard protocol used by most
data networks.
One of the advantages of using
packet-switched networks to carry
telephone communication is that
the infrastructure is already
largely in place in the form of
the many data networks that make
up the Internet, and that
infrastructure is already
understands the technology.
There are two major protocols used
by VoIP technology to allow
telephones, computers and other
devices on the data network to
communicate with each other: H.323
and SIP. The H.323 standard, a
suite of protocols created by the
International Telecommunications
Union is a very wide-ranging and
very complicated protocol,
providing specifications for a
range of communication including
video conferencing, data sharing
and VoIP, and it can be
complicated to set up. The Session
Initiation Protocol (SIP) emerged
after H.323 as an alternative,
guided by the Internet Engineering
Task Force. SIP is a much simpler,
more streamlined protocol
developed specifically for VoIP
use, and designed to employ other
protocols in handling parts of the
communication process.
Using either gateway devices, or
software applications on
computers, VoIP technology can
allow users to communicate by
voice from computer to computer,
computer to telephone, telephone
to computer or telephone to
telephone. Businesses can take
advantage of the technology by
using it to route voice data over
their existing data networks, or
by purchasing VoIP services from
IP service providers.
VoIP technology is growing in
acceptance, and it seems
inevitable that the cheaper, more
efficient technology will play an
important role in the world’s
telephone communications. But it
can also mean immediate cost
savings and improvement in
efficiency for businesses that
chose to implement it now.
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Technology Behind
VoIP (Internet
Protocol) technology unites the telephony and data worlds.
VoIP allows phone calls, faxes and voice traffic to be
relayed over corporate Intranets or at home across the
Internet/Intranet. VoIP technologies convert digitized
voice into data packets that are encapsulated in Internet
protocol. This form of data is then routed between
multiple DVFG’s, providing toll-quality voice, directory
services, and complex voice quality assurance
capabilities.
Two standards have
emerged for signaling and control of VoIP telephony: ITU-T
H.323 and the IETF Session Initiated Protocol (SIP). These
protocols, although resulting in the same end-user service
(telephony), differ in the approach to providing signaling
functions. H.323 is based more on a monolithic bloc
derived from H.320 for traditional of the traditional
circuit-switched ISDN multimedia, and SIP favors a more
lightweight approach based on HTTP.
ITU - T H.323
H.323 is an ITU
recommendation for packet-based multimedia communications
systems. It was originally developed for multimedia
communications over LANs. H.323 addresses the transport of
audio, video, and data over any underlying packet network.
Today, one of the most common uses of H.323 is the call
control structure for VoIP networks.
H.323 is often called
an "umbrella" recommendation. This means it provides an
overall framework for the recommendation, but actually
relies on many other recommendations to specify the
details of operation. For example, H.323 specifies the
components of an H.323 network and a call model for
logical connections, but doesn't specify the details of
signaling or audio and video coding. For these additional
functions H.323 merely provides a reference to the
appropriate recommendation. Therefore, H.323 is an
"umbrella" that covers many recommendations.
Components of H.323
H.323 defines four
logical components viz., Terminals, Gateways, Gatekeepers
and Multipoint Control Units (MCUs). Terminals, gateways
and MCUs are known as endpoints.
Terminal:
It is any end-user client device that uses the H.323
protocol suite for communication. A terminal must support
audio (voice capability is a requirement for a terminal);
video and data capabilities are optional. For control, the
terminal must support the H.323 referenced standards,
H.225 and H.245. For example, a minimal H.323 terminal
would be a telephone that uses standard G.711 Pulse Code
Modulation (PCM) plugged into a packet a network (e.g., an
Ethernet) and uses H.225 and H.245 for connection control.
More capable terminals might include some standard speech
compression capabilities, along with standard video or
data communications capabilities.
MCU:
The MCU is required to support conferencing of three or
more terminals. The MCU provides classic conferencing
functions like audio bridging and video or data switching.
The MCU may be a stand-alone device supporting the full
H.323 control stack, or the MCU functions may be
integrated into other H.323 components. In order to
facilitate MCU distribution, its operations are broken
into two functional parts: a multipoint controller (MC)
and a multipoint processor (MP).
Gateway:
This is the key component in today's VoIP implementations.
The gateway's purpose is to provide all the necessary
translations between an H.323 network and any external
network. A gateway must support the full H.323 control
stack for communication within the H.323 network. The
gateway must also support the signaling and communication
protocols of the external network to which it's connected.
Gatekeeper:
The
gatekeeper is the manager of the components that make up
the network. It's responsible for things such as admission
and bandwidth control. Admission control means that other
components (such as terminals and gateways) must ask the
gatekeeper for permission to participate in the network
when they are first connected (similar to a registration
server), as well as on a regular basis to keep records
current.
Session Initiation
Protocol (SIP)
Many consider the
Session Initiation Protocol (SIP) a powerful alternative
to H.323. They say SIP is a more flexible solution,
simpler than H.323, easier to implement, better suited to
the support of intelligent user devices, and better suited
to the implementation of advanced features. These factors
are of major importance to any equipment vendor or network
operator. Simplicity means that products and advanced
services can be developed faster and made available to
subscribers more quickly. The features themselves mean
that operators are better able to attract and retain
customers and also to offer the potential for new revenue
streams.
SIP is designed to be a
part of the overall Internet Engineering Task Force (IETF)
multimedia data and control architecture. As such, SIP is
used in conjunction with several other IETF protocols,
such as the Session Description Protocol (SDP), the
Real-Time Streaming Protocol (RTSP), and the Session
Announcement Protocol (SAP).
Advantages of VoIP
Voice over IP Telephony
has been fairly popular in last few years and there are
several reasons carriers should be interested:
First, it offers a
short/medium term arbitrage opportunity. This means that
for example it is cheaper to make IP telephony calls than
a circuit call because the operators avoid paying
interconnect charges.
Second, because of
engineering economics: A circuit switched telephony call
takes up 64 KBPS while an IP telephony call takes up 6-8
KBPS.
Third, it offers
exciting new added values in the long term. Such value
added opportunities include: IP multicast conferencing and
telephony distance learning applications, phone
directories and screen popping via IP, and “voice web
browsing” where the caller can interact with a web page by
speaking commands.
Lastly, IP telephony
gives carriers ability to manage a single network handling
both voice and data.
Future Challenges
for VoIP
The main problems with
the VoIP technology as it is today are the
interoperability, security, and bandwidth management
issues. All three of these are major stumbling blocks that
will keep VoIP technology from being implemented
immediately into large corporations; until these problems
are fixed, standard PBX’s will remain the norm for voice
communications.
Conclusion
With the advent of
Internet telephony the idea of Voice over Internet
Protocol (VoIP) has begun to gain notice not only from
companies and universities, but also from private
individuals.
The main reason being
that VoIP allows for telephone calls to be made over the
internet, thus eliminating the cost of long distance phone
charges that the phone companies charge of long distance
phone calls. While this money saving technology is good
news for small and large consumers, there are still many
problems that have to be worked out before this becomes a
viable alternative to everyday telecommunications.
Thus, as more
technology is developed and price for VoIP products become
less expensive, VoIP will become a reality for everyone.
VoIP technology will revolutionize the telecommunications
industry and if used properly can create many positives
for companies and private individuals alike. |
