1

VOICE OVER INTERNET PROTOCOL - Voip

A SEMINAR REPORT
SUBMITTED TOWARDS PARTIAL FULFILLMENT FOR THE AWARD

OF DEGREE

Master of Technology

In

Electronics & Communication Engineering

Submitted By
PRAKASH RANJAN

MECE-170-2K10

ELECTRONICS AND COMMUNICATION ENGINEERING DEPARTMENT

Y.M.C.A.University of Science & Technology (Faridabad)

INDEX
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S.No. CONTENTS PAGE NUMBER

1. Abstract 3
2. Introduction 4
3. History of VoIP 5
4. How VoIP works ? 6
5. Requirements of a VoIP 9
6. H.323 Protocol 13
7. SIP Protocol 18
8. Q.923 Protocol 19
9. H.245 Protocol 20
10. Advantages of VoIP 20
11. Advanced Applications 22
12. Opportunities 22
13. Weaknesses (limitations) 23
14. Security issue 25
15. VoIP “Equipment and Solution” Vendors 28
16. Conclusion 30
17. REFERENCES 31

1) Abstract
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Ever tried placing a voice call over the Internet ? If you have, we are sure you
haven’t had a pleasant experience. You might have even promised yourself
never to try it again.
But , Stop right there!!

Now get ready to change your mind. In the near future, if you make a telephone
call, it is more than likely that it would be over the Internet or some other packet
network. But, what is it that would make this possible? It is a bunch of protocols
and standards; and years of research done by organizations all over the world
that would bring about this revolution. They call it ‘VOICE OVER IP’,
‘INTERNET TELEPHONY’ & a host of other names. VoIP (voice over IP -
that is, voice delivered using the Internet Protocol) is a term used in IP
telephony for a set of facilities for managing the delivery of voice information
using the Internet Protocol (IP). In general, this means sending voice
information in digital form in discrete packets rather than in the traditional
circuit-committed protocols of the public switched telephone network (PSTN).
A major advantage of VoIP and Internet telephony is that it avoids the tolls
charged by ordinary telephone service. VoIP is therefore telephony using a
packet based network instead of the PSTN (circuit switched).
With the passes of time VoIP replace the PSTN and people will acquaint with
the new technology which will reduce their cost and they will no more inclined
to the PSTN to make a call anywhere. Somehow it will take time to replace the
communication system invented by Bell.

2) Introduction
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Voice over IP (VoIP) is a blanket description for any service that delivers
standard voice telephone services over Internet Protocol (IP). Computers to
transfer data and files between computers normally use Internet protocol.

"Voice over IP is the technology of digitizing sound, compressing it, breaking it

up into data packets, and sending it over an IP (internet protocol) network
where it is reassembled, decompressed, and converted back into an analog
wave form.." The transmission of sound over a packet switched network in this
manner is an order of magnitude more efficient than the transmission of sound
over a circuit switched network.

VoIP saves bandwidth also by sending only the conversation data and not
sending the silence periods. This is a considerable saving because generally
only one person talks at a time while the other is listening. By removing the
VoIP packets containing silence from the overall VoIP traffic we can reach up
to 50% saving. In a circuit switched network, one call consumes the entire
circuit. That circuit can only carry one call at a time.

In a packet switched network, digital data is chopped up into packets, sent

across the network, and reassembled at the destination. This type of circuit can
accommodate many transmissions at the same time because each packet only
takes up what bandwidth that is necessary.. Internet Telephony simply takes
advantage of the efficiencies of packet switched networks.

Gateways are the key component required to facilitate IP Telephony. A gateway

is used to bridge the traditional circuit switched PSTN with the packet switched
Internet. The gateway allows the calls to transfer from one network to the other
by converting the incoming signal into the type of signal required by the
network it is required to send it on. For example, A PC user wishes to call
someone using a conventional phone. The PC sends the IP packets containing
digitized voice to the gateway.
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Applications involving voice over internet protocol technology:

 Internet Voice Telephony.

 Intranet & Enterprise network voice telephony.
 Internet fax service.
 Multimedia internet collaboration.
 Internet call centres.
 PBX intercommunications.

3) History of VoIP

During the early 90's the Internet was beginning its commercial spread. The
Internet Protocol (IP), part of the TCP/IP suite (developed by the U.S.
Department of Defense to link dissimilar computers across many kinds of data
networks) seemed to have the necessary qualities to become the successor of the
PSTN.

The first VoIP as a technology demonstration, was introduced in 1995 - an

"Internet Phone". An Israeli company by the name of "VocalTec" was the one
developing this application. The application was designed to run on a basic PC.
The idea was to compress the voice signal and translate it into IP packets for
transmission over the Internet.

VocalTec's Internet phone was a significant breakthrough, although the

application's many problems prevented it from becoming a popular product.
Since this step IP telephony has developed rapidly. The most significant
development is gateways that act as an interface between IP and PSTN
networks.
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This "first generation" VoIP application suffered from delays (due to

congestion), disconnection, low quality (both due to lost and out of order
packets) and incompatibility.

4) How VoIP works ?

i. Part one :- Technology overview

Let us look at very simple VoIP call. Consider two VoIP telephones connected
via an IP network .In this example both VoIP telephones are connected to a
local LAN. Sally’s phone has an IP address of 192.168.1.1 ,Bill’s phone is
192.168.1.2, the IP addresses uniquely identify the telephones. Both our phones
are configured to use a widely used VoIP standard called H.323.

Bill wants to talk to Sally and his phone knows the IP address of Sally’s phone.
Bill lifts the handset and 'dials' Sally, the phone sends a call setup request
packet to Sally's phone, Sally’s phone starts to ring, and responds to Bill's phone
with a call proceeding message. When Sally lifts the handset the phone sends a
connect message to Bill's phone. The two phones will now exchange the data
packets containing the speech. At the end of the call Bill replaces his handset
and phone stops sending voice data sends a disconnect message and Sally's
phone responds with a release message.
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IP PBX A traditional Private Branch Exchange (PBX) connects all the phones
within an organization to the public telephone network. Essentially IP PBX
replaces all the internal phones with VoIP telephones. The IP PBX has standard
telephone trunk connections to the public telephone network. The IP PBX is a
PBX with VoIP, but it also has the ability to support VoIP over the Internet and
Office to Office VoIP.

ii. Part 2 : The Protocols.

I have made an assumption that both ends of a VoIP telephone conversation are
compatible. This compatibility only happens if both ends agree to use the same
protocol. All manufacturers who claim to be producing industry standard voice
over IP either support SIP or H.323 protocol.

iii. Part 3 : Encoding

The call control part of H.323 sets up the parameters for the full duplex voice
path between source telephone and destination telephone. I will continue with
my analogies to explain how your voice gets transported across the Internet. In
terms of H.323 there is a trade off between call quality and bandwidth, in
general the higher the quality the greater the bandwidth required. During the
call setup portion of H.323 the phones have to decide which speech
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encoder/decoder to use when they send the speech to the other phone, Bill and
Sally both have phones that support G.723.1, G.711 and G729. The main
difference between each of these encoders is the amount of bandwidth they use,
G.711 uses 64kbit/s and G.723.1 can use as little as 5.3kbit/s. Although it would
seem obvious to use the encoder with the lowest bandwidth, there is a loss of
quality with a lower bandwidth.. At the same time a stream of G723.1 encoded
voice data starts being sent from each phone to the other phone.

G.723.1 defines how an audio signal with a bandwidth of 3.4KHz should be

encoded for transmission at data rates of 5.3Kbps and 6.4Kbps. G.723.1
requires a very low transmission rate and delivers near carrier class quality. The
VoIP Forum as the baseline Codec for low bit rate IP Telephony has chosen this
encoding technique.

G.711. The ITU standardised PCM (Pulse Code Modulation) as G.711. This
allows carrier class quality audio signals to be encoded for transmission at data
rates of 56Kbps or 64Kbps. G.711 uses A-Law or Mu-Law for amplitude
compression and is the baseline requirement for most ITU multimedia
communications standards.

iv. Part 4 : Hear the Quality

The performance or quality of the speech depends up on encoders at each end,

the number of packets lost on route, Latency and Jitter. As a general rule the
occasional lost packet will not affect too drastically the quality of a call, but lose
in 5 a row the entire word is lost and this will be a problem. So if you are going
to have lost packets make sure they are only lost in a regular distributed manner.
5% lost packets distributed evenly will not result in the loss of words lose .
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Quality also vary with manufacturer . If a manufacturer has a G.723.1 encoder

it may not sound the same as another manufacturer who claims to have G.723.1,
quality does vary.

5) Requirements of a VoIP

The requirements for implementing an IP Telephony solution to support Voice

Over IP varies from organization to organization, and depends on the vendor
and product chosen. The following section aims to identify the fundamental
requirements :-

Software Requirements

Hardware Requirements

Protocol Requirements

i. Software Requirements

The software package chosen will reflect the organizational needs, but should
contain the following modules - Voice Over IP Publication, and other sources.

Voice Processing Module:- This aspect of the software is required to prepare

voice samples for transmission. The functionality provided by the voice
processing module should support:

A PCM Interface is required to receive samples from the telephony

interface (e.g. a voice card) and forward them to the Voice Over IP software for
further processing.

Echo Cancellation is required to reduce or eliminate the echo introduced as a

result of the round trip exceeding 50 milliseconds.
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Idle Noise Detection is required to suppress packet transmission on the

network when there are no voice signals to be sent. This helps to reduce
network traffic as up to 60% of voice calls are silence and there is no point in
sending silence.

A Tone Detector is required to discriminate between voice and fax signals

by detecting DTMF (Dial Tone Multi frequency) signals.

The Packet Voice Protocol is required to encapsulate compressed voice and

fax data for transmission over the network.

A Voice Playback Module is required at the destination to buffer the

incoming packets before they are sent to the Codec for decompression.

Call Signaling Module. This is required to serve as a signalling gateway

which allows calls to be established over a packet switched network as opposed
to a circuit switched network (PSTN for example).

Packet Processing Module. This module is required to process the voice and
signaling packets ready for transmission on the IP based network.

Network Management Protocol. Allows for fault, accounting and

configuration management to be performed.

ii. Hardware Requirements

The exact hardware, which would be required, again, depends on organizational

needs and budget. The list below highlights the most general hardware required.

The most obvious requirement is the existence (or installation) of an IP based

network within the branch office gateway is required to bridge the differences
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between the protocols used on an IP based network and the protocols used on
the PSTN.

The gateway takes a standard telephone signal and digitizes it before

compressing it using a Codec. The compressed data is put into IP packets and
these packets are routed over the network to the intended destination.

The PC's attached to the IP based network require the voice/fax software
outlined above. They also require Full Duplex Voice Cards which allow both
communicating parties to speak at the same time - as often happens in reality.

As an alternative to installing Voice Cards, IP Telephones can be attached to the

network to facilitate Voice Over IP. A secondary gateway should be considered
as a backup in the event of the failure of the primary gateway.

iii. Protocol Requirements There are many protocols in existence but

the main ones are considered to be the following:
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H.323 is an ITU (International Telecommunications Union) approved standard

which defines how audio /visual conferencing data is transmitted across a
network. H.323 relies on the RTP (Real-Time Transport Protocol) and RTCP
(Real Time Control Protocol) on top of UDP (User Datagram Protocol) to
deliver audio streams across packet based networks.

a) Real-Time Transport Protocol (RTP)

It defines a standardized packet format for delivering audio and video over IP
networks. It is the standard protocol for streaming applications developed
within the IETF (Internet Engineering Task Force).

RTP is used extensively in communication and entertainment systems that

involve streaming media, such as telephony, video teleconference applications
and web-based push-to-talk features.
RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP
carries the media streams (e.g., audio and video), RTCP is used to monitor
transmission statistics and quality of service (QoS) and aids synchronization of
multiple streams. When both protocols are used in conjunction, RTP is
originated and received on even port numbers and the associated RTCP
communication uses the next higher odd port number.
It is one of the technical foundations of Voice over IP and in this context is
often used in conjunction with a signaling protocol which assists in setting up
connections across the network. RTP was developed by the Audio-Video
Transport Working Group of the Internet Engineering Task Force (IETF) and
first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003.

b) Resource Reservation Protocol (RSVP) is the protocol which supports

the reservation of resources across an IP network. RSVP can be used to indicate
the nature of the packet streams that a node is prepared to receive.
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Fig :- OSI reference model layer in VOIP

6) What is H.323 ?

It is designed to act above transport layer and is mainly used for transmission of
voice, data and video conferencing over packet networks .Over the next few
years, the industry will address the bandwidth limitations by upgrading the
Internet backbone to asynchronous transfer mode (ATM), the switching fabric
designed to handle voice, data, and video traffic. Such network optimization
will go a long way toward eliminating network congestion and the associated
packet loss. The Internet industry also is tackling the problems of network
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reliability and sound quality on the Internet through the gradual adoption of
standards.

H.323 can be applied in a variety of mechanisms

 Audio only (IP telephony),

 Audio and Video (Video telephony),
 Audio, Video, Data.
H.323 can also be applied to multipoint-multimedia communications. H.323
provides myriad services and therefore can be applied in a wide variety of
areas : Consumer, Business and Entertainment.

H.323 Components :-

The H.323 standard specifies 4 kinds of components, which when networked

together, provide the point–to-point and point-to-point-to-multipoint multimedia
– communication services :

 Terminals
 Gateways,
 Gatekeepers,
 Multipoint Control Units (MCU’s).
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i. Terminals:

Used for real-time bidirectional multimedia communications, an H.323 terminal

can either be a personal computer (PC) or a stand-alone device, running an
H.323 & the multimedia applications. It supports audio communication and can
optionally support video or data communications. As the basic service provided
by an H.323 terminal is audio communications, an H.323 terminal plays a key
role in IP-telephony services.

H.323 terminals are the client endpoints on the LAN that provide real-time,
two-way communication. They can be realized either as SW-Clients running on
a PC or workstation or as a dedicated HW-devices. All terminals must support
voice communication; video and are optional.

Function of the Terminal:

The terminal is the user or the end point.

 Provides real-time 2 way communication.

 Optional : Video & data streaming.
 Mandatory voice streaming.

Protocols Supported by the Terminal

 H.232
 H.245 (Control channel usage & capabilities)
 sQ.931 (Call setup & signaling )
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 RAS (for use with Gatekeepers/Registration/ Admission/Status)

 RTC/RTCP (sequence Audio & status video packets)

ii. Gateways :

A gateway connects two dissimilar networks. An H.323 gateway provides

connectivity between an H.323 network and a non-H.323 network. For example,
a gateway can connect and provide communication between an H.323 terminal
and SCN networks (SCN networks includes all switched telephony networks,
e.g. public switched telephone network [PSTN] ).

This connectivity of dissimilar networks is achieved by translating protocols for

call setup and release, converting media between different networks, and
transferring information between the networks connected by the gateway. A
gateway is not required, however, for communication between two terminals on
an H.323 network.

The Gateway also translates between audio and video codec’s and performs call
setup and clearing on both the LAN side and the PSTN side.

The Functions of the gateway can be stated as follows:

 Task-translation.
 Audio Codec
 Video codec
 H.245  H.221 (ISDN Conference)
H.245  H.242 (Audio-Visual terminals)
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iii. Gatekeepers :-
A gatekeeper can be considered the brain of the H.323 network. It is the most
important component of an H.323 enabled network.

It is the focal point for all calls within the H.323 network.

Although they are not required, gatekeepers provide important services such as
addressing, authorization and authentication of terminals and gateways;
bandwidth management; accounting; billing; and charging. Gatekeepers may
also provide call-routing service.

The Functions of the Gatekeeper can be stated as follows :

 Task: user information / “name server”

 Gatekeeper is optional but essential.
 Managing communications.
 Address translation.
 Call Control.
 Routing services.
 System management.
 Security policies.

iv. Multipoint Control Unit:-

MCU’s provide support for conferences of three or more H.323 terminals. All
terminals participating in the conference establish a connection with the MCU.

The MCU manages conference resources, negotiates between terminals for the
purpose of determining the audio or video coder/decoder (CODEC) to use, and
may handle the media stream.
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The gatekeepers, gateways and MCU’s are logically separate components of the
H.323 standard but can be implemented as a single physical device.

The functions of the MCU can be stated as follows:

 Task : maintain all audio, video data & control streams

mandatory for conferences.

7) What is SIP ?
The Session Initiation Protocol (SIP) is a signaling protocol for initiating,
managing and terminating voice and video sessions across packet networks.
SIP sessions involve one or more participants and can use unicast or multicast
communication. Borrowing from ubiquitous Internet protocols, such as HTTP
and SMTP, SIP is text-encoded and highly extensible
A SIP network is composed of four types of logical SIP entities. Each entity has
specific functions and participates in SIP communication as a client (initiates
requests), as a server (responds to requests), or as both.

i. User Agent (UA):-

In SIP, a User Agent (UA) is the endpoint entity. User Agents initiate and
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terminate sessions by exchanging requests and responses.

User Agent as an application, which contains both a User Agent client and User
Agent server, as follows:6 RADVISION SIP Overview
User Agent Client (UAC)—a client application that initiates SIP requests.
User Agent Server (UAS)—a server application that contacts the user when a
SIP request is received and that returns a response on behalf of the user.
ii. Proxy Server
A Proxy Server is an intermediary entity that acts as both a server and a client
for the purpose of making requests on behalf of other clients.

iii. Redirect Server

A Redirect Server is a server that accepts a SIP request, maps the SIP address of
the called party into zero (if there is no known address) or more new addresses
and returns them to the client.
iv. Registrar
A Registrar is a server that accepts REGISTER requests for the purpose of
updating a location database with the contact information of the user specified
in the request.

8) What is Q.923 ?

Network Layer is specified by the ITU Q-series documents Q.923 through

Q.939. Q.931, the layer 3 protocol, is used for the ISDN call establishment,
maintenance, and termination of logical network connections between two
devices. During the layer 3 call setup, three parties are involved, and are where
messages sent and received
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a) the Caller,
b) the ISDN Switch, and
c) the Receiver.

9) What is H.245  ?

It is a control channel protocol  used with [in] e.g.  H.323  and H.324 

communication sessions, and involves the line transmission of non-
telephone signals .  It also defines separate  send and receive capabilities and the
means to send these details to other devices that support H.323. One major
drawback within the initial version of H.323 was the lengthy, four-way H.245
protocol handshake required during the opening up the logical channels of a
telephony session. Later versions of H.323 introduced the Fast
Connect procedure, using the fastStart element of an H.225.0 message. Fast
Connect brought the negotiation down to a two-way handshake. 

10) Advantages of VoIP

There are many advantages to be gained from implementing an IP Telephony

solution within the organization. The following list aims to highlight some of
the advantages of such a strategy:
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Single network infrastructure. When installing VoIP in the office only a

single cable is required to the desk, for both telephone and data. Eliminating
separate telephone wiring.

VoIP uses "soft" switching which eliminates most of the legacy PBX
equipment. Reducing the cost of installing a communications infra-structure and
the maintenance cost once installed.

Simple upgrade path. The VoIP PBX technology is software based. It is easier
to expand, upgrade and maintain than its traditional telephony counterparts.

Bandwidth efficiency. VoIP can compress more voice calls into available
bandwidth than legacy telephony.. IP Telephony helps to eliminate wasted
bandwidth by not transporting the 60% of normal speech which is silence
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IP - the underlying protocol - is supported by most platforms and is

independent of the transport protocol used.

Only one physical network is required to deal with both voice/fax and
data traffic instead of two physical networks. Having only one physical network
has the following advantages:

lower physical equipment cost ,lower maintenance costs.

11) Advanced Applications:

The most compelling aspect of converged voice/data networking may well be

the new generation of applications it enables. These applications include Web
enabled call canters, unified messaging and real-time collaboration. Other
examples include real-time multimedia video/audio conferencing, distance
learning, and the embedding of voice links into electronic documents. Three or
four years ago, the Internet was not ready for prime time as a medium for
commerce. But now it is. VoIP will be a valuable enhancement.

12) Opportunities

Many vendors offer the ability to incorporate Virtual Private Networking (VPN)
with relative ease into the IP Telephony solutions they provide. This allows any
transmission to be encrypted using a number of cryptographic techniques and
providing security by transmitting the communications through a 'tunnel' which
is set up using PPTP (Point-to-Point Tunnelling Protocol) before commencing
communications.IP Telephony allows companies to exploit Computer
Telephony Integration to its full extent.
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The convergence of communications technologies allows greater control over

communications, most vendors provide logging and accounting facilities
whereby all usage can be monitored.

13) Weaknesses (limitations):-

While there are many aspects of VoIP which provide considerable benefits, the
technology is still very young and problems remain. The following section
looks at some of the weaknesses of this technology and their consequences.

The Internet is not the best medium for real time communications.
Individual packets can take different routes and varying delays can be
encountered and packets lost in transit. Waiting for delayed packets or
retransmission of lost packets can result in considerable degradation of quality.
Long delays in transit can affect quality so much that the technology can
become unusable, though many vendors do have solutions which aim to negate
the degradation suffered due to transit delays.

While some standards have been set by the ITU, the technology is not fully
standardized and there is no guarantee that products from different vendors will
be interoperable. Some vendors are trying to resolve this problem by forming
groups and making guarantees about the products in the group but this is only a
partial solution - vendors out with the group cannot guarantee interoperability.

Jitter: Jitter in technical terms is the deviation in or displacement of some

aspect of the pulses in a high-frequency digital signal. As the name suggests,
jitter can be thought of as shaky pulses. The deviation can be in terms of
amplitude, phase timing, or the width of the signal pulse.
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Jitter refers to non-uniform packet delays. It is often caused by low bandwidth

situations in VOIP and can be exceptionally detrimental to the overall QoS.
Variations in delays can be more detrimental to QoS than the actual delays
themselves . Jitter can cause packets to arrive and be processed out of sequence.
RTP, the protocol used to transport voice media, is based on UDP so packets
out of order are not reassembled at the protocol level. However, RTP allows
applications to do the reordering using the sequence number and timestamp
fields.

Heavy congestion on the network can result in considerable degradation of

service as IP is not good at providing QoS (Quality of Service) guarantees.
Feedback to Lucent Technologies customers reflect this worry. Major
companies are planning to install IP Telephony capabilities at some point and
have carried out initial investigations, however:

Since only one physical network for both data and voice/fax transmissions is
required, failure of the network could be catastrophic, as all communications
capabilities are lost.
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14) Security issue:-

a) The concern:-

VoIP networks and devices are vulnerable to unique security threats and
exploits, because of the real-time nature of the traffic.  In addition, many new
protocols and products make up the VoIP infrastructure, which traditional
security products and approaches are not designed to protect.

The flaws were discovered by VoIP security solutions vendors (like VoIPshield)

which revealed the vulnerabilities to the public today.

Since VoIPshield Labs and like that labs are continuously finding new

vulnerabilities, they plan on monthly disclosures to VoIP equipment vendors
followed by public disclosure.

An interesting example of an identified Cisco VoIP vulnerability  is shown

below:

In the above example, a potential attacker exploiting the Cisco Unified

Communication Manager (UCM) vulnerability related to its Disaster Recovery
 26

Network, could obtain full access to the UCM by getting the remote shell on the
attacker's machine.

Subsequently the attacker could either disable UCM completely, download all
the information from UCM to the attacker's machine or upload an executable
file to the UCM.

Then the attacker could force all the Cisco softphones connected to this UCM to
reboot and download that executable file.

It could be a bot, Trojan or worm.

Once the executable is downloaded and executed an attacker is able to have full
access to the user’s laptop running the softphone.

b) The counter measure products:-

Each product draws on VoIPshield’s proprietary database of VoIP-specific

vulnerabilities and signatures. They are some what analogues to Anti viruses.
Some of them are:-

   
VoIP Vulnerability Assessment (VVA) and penetration testing tool has been
deployed by enterprise security and telecom professionals in many network
scenarios and is being used for both risk assessment, compliance focused
auditing as well as operationally by security staff to ensure their Voice over IP
network are secure at all times. 

   
VoIPguard is the industry’s first VoIP Intrusion Prevention System (VIPS) with
comprehensive protection for Voice over IP systems from the leading VoIP
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vendors. VoIPguard  provides effective protection against known and new

attacks aimed at compromising the security of VoIP networks.

   

VoIP PBX monitoring tool provides around the clock monitoring  for hardware
failures and systems events generated by VoIP IP Public branch exchanges
(PBX). VoIPmonitor empowers PBX administrators to selectively forward
events in real-time to destinations such as in-house Network Operations Centers
(NOC), Network Management Systems (NMS) and Operational Support
Systems (OSS) while preserving local administration processes without the
need for external monitoring services. 

   
VoIP PBX configuration management tool enables PBX administrators to
quickly catalog, assess and report on configuration status and configuration
changes as they apply to regulator requirements. It allows the PBX
administrators detect configuration issues and changes improving overall
security and reliability of VoIP installations. 

…….. and many more companies are providing the similar solutions.
some of them are:-

Nextiva 

Vocality

DCP SERVICES INC

Quintum Technologies, LLC.

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SecureWorks, Inc.

CREDANT Technologies 

15) VoIP “Equipment and Solution” Vendors

Avaya
Avaya is the leader on the VoIP market. It has a vast variety of equipment and
solutions including IP phones, media gateways, routers, communication servers,
PBXs, wireless phones, voice application etc. Its hottest product is the IP Office
platform and communications system.

Nortel
Nortel is one of the oldest vendors around - it has been here since 1995. The
range of products it offers is more or less like Avaya, but it is more reputed for
its PBXs. Nortel is the first vendor to provide an end-to-end IP telephony
solution certified by U.S. Defence Department Joint Interoperability Test
Command (JITC) in 2004.

Cisco
Before VoIP, Cisco has been around as a networking vendor giant. Now, it is a
leading player in the VoIP equipment arena, with good IP Phones, routers and
other devices and solutions. Cisco owns Linksys, which is also well known on
the market.

Shoretel
Shoretel is a fast-growing vendor providing complete IP telephony solutions
and PBXs. Shoretel's strong points are performance and ease of installation and
use.
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Siemens
Siemens is another giant in technology which has extended its reach to VoIP
and unified communications. It provides phones, PBXs and ... just anything
concerning communication.

Mitel
Mitel is a global provider of communications solutions for enterprises and small
business. It aims at blending good human interface with powerful infrastructure.
Mitel brands include SX and ICP series.

Asterisk
Asterisk is not hardware, but a software PBX that can run on Linux, Unix,
MacOS and some other operating systems. It is open source (meaning free) and
is extremely flexible and very robust.

Taridium
Taridium is a fast growing open standards VoIP solution provider. Taridium's
offering ranges from managed VoIP services for small and medium sized
businesses through to high capacity telephony solutions for large enterprises and
service providers.

Here are some of the other providers:

Talkswitch
Toshiba
Fonality 
3Com
Polycom
Alcatel
Lucent (now owned by Avaya)
Linksys (now owned by Cisco)
D-Link
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Sipura (now owned by Cisco)

Grandstream
Snom
NetGear
ZyXEL
Belkin

16) Conclusion :-
Without a doubt, the data revolution will only gain momentum in the coming
years, with more and more voice traffic moving onto data networks. Vendors of
voice equipment will continue to develop integrated voice and data devices
based on packetized technology. Users with ubiquitous voice and data service
integrated over one universal infrastructure will benefit from true, seamless,
transparent interworking between voice and all types of data. “It is not Voice
over IP, it is Everything over IP…”
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17) REFERENCES

Journals

An OPNET-based Simulation Approach for Deploying VoIP by K. Salah and

A. Alkhoraidly

Voice over IP (VoIP) by Spirent Communications

Conference Recomendations

Internet telephony fall 2005- conference - LOS ANGELES

ITEXPO- Conference

Books

Switching to VOIP by Ted Wallingford – 2005 O’RELLY Publications.

Hacking Exposed –VOIP by David Endler, Mark Collier, Endler – 2007 TMH

Publications

Website:-

http://www.voip.com

http://www.networkworld.com

http://www.itu.int
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“VoIP Equipment and Solution Vendors” By Nadeem Unuth, About.com

Guide