THE UNIVERSITY OF BAMENDA

NATIONAL HIGHER DEPARTMENT OF

POLYTECHNIC ORIGIN
INSTITUTE

DESIGN AND IMPLEMENTATION OF WIRELESS VOICE

OVER INTERNET PROTOCOL (VoIP) SOLUTION
CASE STUDY: IGT (Infogenie Technologies) YAOUNDE

A Project Submitted to the Department of Computer in the National Higher

Polytechnic Institute of the University of Bamenda in Partial Fulfillment of the
Requirements for the Award of a Bachelor of Engineering Degree in Computer

BY:

GUIMASSING DIVISSELLE

REGISTRATION NUMBER: UBa19E0329

SUPERVISOR(S):
Mr. Taku Angwa Otto Che
Mr. Tchasso Serge Paulin
Computer Engineer
Assistant Lecturer

JUNE, 2023
 THE UNIVERSITY OF BAMENDA

NATIONAL HIGHER DEPARTMENT OF

POLYTECHNIC ORIGIN
INSTITUTE

DESIGN AND IMPLEMENTATION OF WIRELESS VOICE

OVER INTERNET PROTOCOL (VoIP) SOLUTION
CASE STUDY: IGT (Infogenie Technologies) YAOUNDE

A Project Submitted to the Department of Computer in the National Higher

Polytechnic Institute of the University of Bamenda in Partial Fulfillment of the
Requirements for the Award of a Bachelor of Engineering Degree in Computer

BY:

GUIMASSING DIVISSELLE

REGISTRATION NUMBER: UBa19E0329

SUPERVISOR(S):

Mr. TCHASSO SERGE PAULIN Mr. TAKU ANGWA OTTO CHE

Assistant Lecturer Computer Engineer

JULY, 2023
ii
©Copyright by ……guimassingdivisselle……., 2023
All rights reserved

i
 DECLARATION OF ORIGINALITY OF STUDY
I, GUIMASSING DIVISSELLE, registration N◦: UBa19E0329, in the Department of
Computer Engineering, National Higher Polytechnic Institute, and The University of Bamenda
hereby declare that this work titled “DESIGN AND IMPLEMENTATION OF WIRELESS
VOICE OVER INTERNET PROTOCOL (VoIP) SOLUTION” is my original work. It has
not been presented in any application for a degree or any academic pursuit elsewhere. I have
acknowledged all borrowed ideas nationally and internationally through citations.

Date: ___________________ Signature of author ________________

ii
 CERTIFICATION OF CORRECTIONS AFTER DEFENSE
This is to certify that this project titled “DESIGN AND IMPLEMENTATION OF
WIRELESS VOICE OVER INTERNET PROTOCOL (VoIP) SOLUTION” is the original
work of GUIMASSING DIVISSELLE. This work is submitted in partial fulfillment of the
requirements for the award of a Bachelor of Engineering Degree in Computer Engineering in
the National Higher Polytechnic Institute of The University of Bamenda, Cameroon.

Supervisor: ______________________________________________________
Mr. Tchasso Serge Paulin

Co-supervisor ____________________________________________________

Mr. Taku Angwa Otto Che

The Head of Department: __________________________________________

Dr. Ndukum Pascaline Liaken Dickmu Epse

The Director: ____________________________________________________

Prof. Fidelis Cho-Ngwa

iii
 ABSTRACT

The voice over internet protocol (VOIP) allows for transmission of voice over the internet in a
more efficient and cost effective manner than traditional telephony. This is achieve through the
use of digital packets switching technology, which enables the transmission of voice and data
over the same network. VOIP technology offers a number of advantages over traditional
telephony, including lower costs, greater flexibility, better scalability, and access to advanced
communication features. However, VOIP also presents certain challenges such as the need for
quality of service mechanisms to ensure reliable voice communication, security concerns, and
regulatory issues. This technology is very useful in making cheaper calls through the internet
protocol and fine tuned data services. The objective of this study is to design an implement
VoIP solution to transmit conversations over data network using the internet protocol IP. The
research method that will be employed in this study is the Software-based deployment: This
method involves deploying VoIP software on existing hardware, such as personal computers or
mobile devices. The expected outcomes of this project include cost savings, improved
flexibility and mobility, enhanced features, scalability, and improved collaboration, leading to
improved communication efficiency, productivity, and financial performance.

Keywords: VoIP; IP phone, cisco, PSTN

iv
 RESUME

La technologie de la voix sur IP (VoIP) permet la transmission de la voix sur Internet de manière
plus efficace et économique que la téléphonie traditionnelle. Cela est rendu possible grâce à
l'utilisation de la technologie de commutation de paquets numériques, qui permet la
transmission de la voix et des données sur le même réseau. La technologie VoIP offre plusieurs
avantages par rapport à la téléphonie traditionnelle, notamment des coûts plus faibles, une plus
grande flexibilité, une meilleure évolutivité et l'accès à des fonctionnalités de communication
avancées. Cependant, la VoIP présente également certains défis tels que la nécessité de
mécanismes de qualité de service pour assurer une communication vocale fiable, des problèmes
de sécurité et des questions réglementaires. Cette technologie est très utile pour effectuer des
appels moins chers via le protocole Internet et des services de données optimisés. L'objectif de
cette étude est de concevoir et de mettre en œuvre une solution VoIP pour transmettre des
conversations sur un réseau de données en utilisant le protocole Internet IP. La méthode de
recherche qui sera employée dans cette étude est le déploiement basé sur un logiciel : cette
méthode consiste à déployer un logiciel VoIP sur un matériel existant, tel que des ordinateurs
personnels ou des appareils mobiles. Les résultats attendus de ce projet comprennent des
économies de coûts, une meilleure flexibilité et mobilité, des fonctionnalités améliorées, une
évolutivité et une collaboration améliorées, conduisant à une efficacité de communication, une
productivité et une performance financière améliorées.

Mots-clés: VoIP; téléphone IP, Cisco, RTPC

v
 DEDICATION

I dedicate this present document to my lovely mother NGUEDIA Suzanne

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 ACKNOWLEDGMENT

We take this opportunity to express our profound gratitude and respect to all those who helped
us through the duration of this thesis:
We owe sincere thanks and gratitude to our supervisor Mr. Tchasso Serge Paulin who accepted
to supervise us out of their busy schedule, and for his relentless effort to see this work succeed.

I would also extent my greater acknowledgment to my co-supervisor Mr. Taku Otto Che for his
greater guidance and technical support as well as in this project. Thanks to him to make the
completion of this report.

The Director of NAHPI, Pr. CHO-NGWA NAHPI for having thought about the opening of this
School and mainly the computer department, whose training fills us with knowledge.

The Head of the Department of computer engineering of NAHPI, Dr. NDUKUM PASCALINE
for follow-up, her advice, and the undeniable interest that she brings to all the students.

Our parents, guardians, brothers, sisters, and staff who in one way or another made this thesis
a success.

Our friends and classmates for their solidarity

To the Lord our Almighty God who graciously gave me life and who guides me, protects me,
accompanies me and supports me.

vii
 TABLE OF CONTENTS

DECLARATION OF ORIGINALITY OF STUDY .................................................................. ii

CERTIFICATION OF CORRECTIONS AFTER DEFENSE.................................................. iii

ABSTRACT .............................................................................................................................. iv

RESUME .................................................................................................................................... v

DEDICATION .......................................................................................................................... vi

ACKNOWLEDGMENT .......................................................................................................... vii

TABLE OF CONTENTS ........................................................................................................ viii

LIST OF FIGURES ................................................................................................................... xi

LIST OF TABLES .................................................................................................................. xiii

LIST OF ABBREVIATIONS ................................................................................................. xiv

Chapter 1: INTRODUCTION .................................................................................................... 1

1.1 Background of Study ................................................................................................... 1

1.2 Problem Statement ....................................................................................................... 2

1.3 Rationale ...................................................................................................................... 3

1.4 Research Questions...................................................................................................... 3

1.4.1 Main research question......................................................................................... 3

1.4.2 Specific research questions .................................................................................. 3

1.5 Objectives .................................................................................................................... 4

1.5.1 Overall Objectives ................................................................................................ 4

1.5.2 Specific Objectives ............................................................................................... 4

1.6 Significance of Study................................................................................................... 4

1.7 Scope and Limitations of This Project ........................................................................ 5

1.8 Definitions of terms ..................................................................................................... 5

viii
 1.8.1 VoIP ..................................................................................................................... 5

1.8.2 PSTN .................................................................................................................... 6

Chapter 2: LITERATURE REVIEW ......................................................................................... 7

2.1 INTRODUCTION ....................................................................................................... 7

2.2 GENERAL STUDY OF VOIP .................................................................................... 7

2.2.1 DEFINITION ....................................................................................................... 7

2.2.2 WORKING PRINCIPLE ..................................................................................... 7

2.2.3 GENERAL ARCHITECTURE OF VOIP ........................................................... 9

2.3 VoIP protocols ............................................................................................................. 9

2.3.1 Signalling protocols............................................................................................ 10

2.3.2 Comparison between H.323 and SIP .................................................................. 14

2.3.3 Transport protocols ............................................................................................ 15

2.3.4 Advantages and disadvantages of VoIP ............................................................. 18

2.4 Related works ............................................................................................................ 21

2.5 Conclusion ................................................................................................................. 22

Chapter 3: Materials and Methods ........................................................................................... 23

3.1 Introduction ............................................................................................................... 23

3.2 MATERIALS ............................................................................................................ 23

3.2.1 Integrated Development Environment (IDE) ..................................................... 23

3.2.2 Tools used .......................................................................................................... 23

3.2.3 Hardware ............................................................................................................ 25

3.3 Prototype .................................................................................................................... 25

3.4 METHODS ................................................................................................................ 26

3.4.1 Bases security configuration on the equipment.................................................. 26

3.4.2 Switches configuration ....................................................................................... 28

3.4.3 Routers configuration ......................................................................................... 30

3.4.4 Wan router configuration ................................................................................... 43

ix
 3.4.5 AAA configuration ............................................................................................. 45

Chapter 4: RESULTS AND DISCUSSION............................................................................. 47

4.1 RESULTS .................................................................................................................. 47

4.1.1 Security test ........................................................................................................ 47

4.1.2 Calls tests............................................................................................................ 47

4.1.3 AAA test ............................................................................................................. 51

4.2 Discussion .................................................................................................................. 51

Chapter 5: conclusions and recommendations ......................................................................... 54

5.1 Conclusions ............................................................................................................... 54

5.2 Recommendations ..................................................................................................... 54

5.2.1 Prioritizing the data ............................................................................................ 54

5.2.2 Wired over Wi-Fi ............................................................................................... 54

5.2.3 Maintain network security .................................................................................. 55

5.2.4 Less latency ........................................................................................................ 55

5.2.5 Network bandwidth ............................................................................................ 55

x
 LIST OF FIGURES
Figure 2.1: General architecture of VoIP ................................................................................... 9
Figure 2.2: Components of H.323 ............................................................................................ 12
Figure 3.1: Cisco router ............................................................................................................ 24
Figure 3.2: Cisco switch ........................................................................................................... 24
Figure 3.3: Cisco IP phone ....................................................................................................... 25
Figure 3.5: Prototype ................................................................................................................ 26
Figure 3.6: Rename the device ................................................................................................. 27
Figure 3.7: Password user mode .............................................................................................. 27
Figure 3.8: Password Privileged mode ..................................................................................... 28
Figure 3.9: Passwords encryption ............................................................................................ 28
Figure 3.10: Switch configuration site A ................................................................................. 29
Figure 3.11: Switch configuration site B ................................................................................. 29
Figure 3.12: Switch configuration site C ................................................................................. 30
Figure 3.13: Switch configuration site D ................................................................................. 30
Figure 3.14: Addressing and activation of the interface f0/0 site A......................................... 31
Figure 3.15: Addressing and activation of the interface f0/0 site B ......................................... 31
Figure 3.16: Addressing and activation of the interface f0/0 site C ......................................... 32
Figure 3.17: Addressing and activation of the interface f0/0 site D......................................... 32
Figure 3.18: DHCP configuration site A .................................................................................. 33
Figure 3.19: DHCP configuration site B .................................................................................. 33
Figure 3.20: DHCP configuration site C .................................................................................. 34
Figure 3.21: DHCP configuration site D .................................................................................. 34
Figure 3.22: Telephony service site A...................................................................................... 35
Figure 3.23: Telephony service site B ...................................................................................... 35
Figure 3.24: Telephony service site C ...................................................................................... 36
Figure 3.25: Telephony service site D...................................................................................... 36
Figure 3.26: Phones numbers assignation site A..................................................................... 37
Figure 3.27: Phones numbers assignation site B ...................................................................... 37
Figure 3.28: Phones numbers assignation site C ...................................................................... 38
Figure 3.29: Phones numbers assignation site D...................................................................... 38
Figure 3.30: Addressing and activation of the serial interface 0/2/0 site A ............................. 39
Figure 3.31: Addressing and activation of the serial interface 0/2/0 site B ............................. 39

xi
Figure 3.32: Addressing and activation of the serial interface 0/2/0 site C ............................. 39
Figure 3.33: Addressing and activation of the serial interface 0/2/0 site D ............................. 39
Figure 3.34: OSPF configuration site A ................................................................................... 40
Figure 3.35: OSPF configuration site B ................................................................................... 40
Figure 3.36: OSPF configuration site C ................................................................................... 40
Figure 3.37: OSPF configuration site D ................................................................................... 41
Figure 3.38: Dial peer configuration site A.............................................................................. 41
Figure 3.39: Dial peer configuration site B .............................................................................. 42
Figure 3.40: Dial peer configuration site C .............................................................................. 42
Figure 3.41: Dial peer configuration site D.............................................................................. 43
Figure 3.42: Addressing and activation of the interface serial 0/0/0 to connect to site A ....... 43
Figure 3.43: Addressing and activation of the interface serial 0/0/1 to connect to site B ........ 44
Figure 3.44: Addressing and activation of the interface serial 0/2/0 to connect to site C ........ 44
Figure 3.45: Addressing and activation of the interface serial 0/2/1 to connect to site D ....... 45
Figure 3.46: Routing protocol (OSPF) configuration .............................................................. 45
Figure 3.47: AAA configuration .............................................................................................. 46
Figure 4.1: Basis security test .................................................................................................. 47
Figure 4.2: Call from site A to site B ....................................................................................... 48
Figure 4.3: Call from site A to site C ....................................................................................... 48
Figure 4.4: Call from site A to site D ....................................................................................... 49
Figure 4.5: Call from site B to site C ....................................................................................... 49
Figure 4.6: Call from site B to site D ....................................................................................... 50
Figure 4.7: Call from site C to site D ....................................................................................... 50
Figure 4.8: AAA test ................................................................................................................ 51

xii
 LIST OF TABLES
Tableau 1.1: List of abbreviations ............................................ Error! Bookmark not defined.
Tableau 2.1: Comparison between H.323 and SIP................................................................... 14

xiii
 LIST OF ABBREVIATIONS
Abbreviations Full meaning
ATM Asynchronous Transfer Module
IP Internet Protocol
ITU International Telecommunication Union
LAN Local Area Network
MCT Multipoint Control Units
NGN Next Generation Network
OSPF Open Shortest Path First
PABx Private Automatic Branch eXchange
POTS Plain Old Telephone Service
PSTN Public Switched Telephone Network
RIP Routing Information Protocol
RAS Registration Admission Status
RTP Real-time Transport Protocol
RTCP Real-time Transport Control Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
TOIP Telephony over Internet Protocol
UAC User Agent Client
UDP User Datagram Protocol
VLAN Virtual Local Area Network
VOIP Voice over Internet Protocol
WAN Wide Area Network

xiv
 Chapter 1: INTRODUCTION
1.1 Background of Study

The origins of VoIP can be traced back to the early 1970s and the development of packet-
switched networks, which were designed to transmit data more efficiently than traditional
circuit-switched networks. In the 1980s, researchers began experimenting with voice
transmission over packet-switched networks, but the technology was not yet capable of
supporting high-quality voice calls. The term VoIP historically referred to using internet
protocols to connect PBXs but is now used interchangeably with IP telephony. Paul Baran and
other researchers worked on early developments of packet network designs. In 1973, Danny
Cohen was the first to demonstrate a form of packet voice over an early Advanced Research
Projects Agency Network. One year later, the first successful real-time conversation was had
over ARPANET. Three years after this, in 1977, User Datagram Protocol was added to carry
real-time traffic.

It was not until the 1990s that VoIP technology became more practical and affordable, thanks
in part to the development of the H.323 protocol by the International Telecommunication Union
(ITU). The H.323 protocol enabled real-time voice and video communication over IP networks,
and it became the first widely adopted standard for VoIP.

In 1991, the first VoIP application release was Speak Freely. A year later, InSoft launched a
desktop conferencing product named Communique. Communique notably included options for
video conferences. InSoft is often credited for creating the first generation of commercial VoIP
services in the United States.
In 1994, the FCC placed a requirement on VoIP providers to comply with the Communications
Assistance for Law Enforcement Act of 1994. In addition, VoIP providers had to now contribute
to the Universal Service Fund.

In 1995, Intel, Microsoft and Radvision began to standardize VoIP systems. One year, later the
ITU-T developed standards for transmission and signaling voice over IP networks, creating the
H.323 standard. The G.729 standard is also introduced. SIP was standardized in

1
In the early 2000s, the emergence of broadband internet and the adoption of the SIP (Session
Initiation Protocol) standard led to a rapid expansion of the VoIP market. SIP is a more flexible
and scalable protocol than H.323, and it has become the de facto standard for VoIP
communication.

In 2005, the FCC began imposing VoIP providers to provide 911 emergency call abilities. This
began opening up the ability for VoIP to make and receive calls from traditional telephone
networks. Emergency calls do work differently with VoIP, however. For example, a provider
with the right hardware infrastructure can find the approximate location of the calling device
by using the IP address that is allocated to the network router.
Another codec, the G.729.1 protocol, was unveiled in 2006. A year after this, VoIP device
manufacturers began to expand in Asia. The SILK codec was introduced in 2009, notable for
being used for voice calling in Skype.
In 2010, Apple introduced the LD-MDCT-based AAC-LD codec, which is notable for being
used in FaceTime.

Today, VoIP is a mature technology that is used by millions of people around the world for
both personal and business communications. It has also paved the way for other IP-based
communication technologies, such as video conferencing and unified communications. With
the continued growth of high-speed internet access and the increasing adoption of cloud-based
services, the future of VoIP looks bright.

1.2 Problem Statement

Traditional telephony systems are expensive and inflexible, with limited features and high cost
of maintenance. Voice over Internet Protocol (VoIP) is a promising alternative that offers cost
savings, flexibility, and advanced features. However, implementing VoIP presents several
challenges, including ensuring reliable voice quality, minimizing latency, managing security
risks, and ensuring interoperability with existing communication systems. Moreover, VoIP
systems require specialized skills and expertise in network design, security, and performance
optimization. Therefore, To address this problem, this study aims design and implement a VoIP
solution that meets the requirements of reliable voice communication, low latency, high

2
security, and interoperability, while minimizing implementation and maintenance costs, and
ensuring ease of use and scalability."

1.3 Rationale

For decades, landline phones have been integral to organizations. From connecting with
customers to internal meetings, phone communication is critical to business operations.
However, landlines are limited in their ability to connect you wherever you are. While mobile
devices have overcome this issue, they are still a costly solution. With voice over internet
protocol (VoIP), you aren’t tethered to traditional networks and can make and receive calls
online with a more seamless experience. With this technology, calls are made via the internet.

VoIP allows you to embrace the benefits of moving telecommunications. A few advantages of
moving from traditional phone systems to the internet are reduced operating and capital
expenses, and improved business continuity and flexibility.

The goal of this study is design and implement voice over internet protocol (VoIP) solution in
order to improve cost savings, Complete Portability, Scalability, and Multitasking.
Organizations will be able to make and receive internal, inbound and outbound calls, which
will help to reduce expenses buy using the same network for data and VoIP (voice over internet
protocol), and enable to make calls wherever you are.

1.4 Research Questions

1.4.1 Main research question

What is the most effective or efficient way to implement a voice over internet protocol solution
and how can organizations ensure that their networks are properly configured to support VoIP?

1.4.2 Specific research questions

How can we reduce network infrastructure costs and enables providers to deliver voice over
Broadband and private networks?

3
How can we provide an alternative way of transmitting voice communication using digital
signals rather than traditional phones?
How can VoIP be configured and customized to meet the specific needs and requirements for
users and organizations?
What are the potential challenges and limitations of implementing VoIP and how can they be
addressed?

1.5 Objectives

1.5.1 Overall Objectives

The overall objective of this project was to design and implement a voice over internet protocol
solution between multiple remote sites and configure networks.

1.5.2 Specific Objectives

The specific objectives of this project were:

1. To evaluate the existing telephony infrastructure and identify areas where VoIP can
provide cost savings and efficiency gains.
2. To design a VoIP system that meets the specific needs and requirements of the
organization, including capacity, scalability, and security.
3. To authenticate users and devices to prevent unauthorized access to sensitive
information.

1.6 Significance of Study

VoIP consolidates communication technologies into one unified system, meaning that VoIP can
enable several audio, video or text-based communication methods. This can be particularly
useful for businesses so teams don't have to work with multiple different applications to
communicate with one another effectively. This study is significant for several reasons

4
Cost savings: VoIP is significantly less expensive than traditional phone systems, as it uses the
internet to transmit voice data instead of dedicated phone lines. This can lead to reduced costs
for long-distance calls, international calls, and even local calls.

Increased flexibility and mobility: With VoIP, users make and receive calls from anywhere with
an internet connection, which lead to increased productivity and flexibility for businesses. VoIP
also allows for features like call forwarding, voicemail-to-email, and video conferencing, which
can enhance communication and collaboration.

1.7 Scope and Limitations of This Project

The design and implementation of VoIP is a complex and multidisciplinary field that requires
the thorough understanding of network architecture, protocols and standards used to enable
voice communication over internet. However the complexity of the underlying technology and
the need for specialized knowledge and expertise to fully understand and analyse the technical
aspects of VoIP may limit its scope or require more time and resources than anticipated. The
project will not include a physical implementation, as the realization will be done using cisco
packet tracer software. This may limit the ability to test and validate the communication in a
real-world setting.

1.8 Definitions of terms

1.8.1 VoIP

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of
technologies for voice calls, the delivery of voice communication sessions over Internet
Protocol (IP) networks, such as the Internet.
The broader terms Internet telephony, broadband telephony, and broadband phone
service specifically refer to the provisioning of voice and other communications services
(fax, SMS, voice messaging) over the Internet, rather than via the public switched telephone
network (PSTN), also known as plain old telephone service (POTS).

5
 1.8.2 PSTN

PSTN stands for Public Switched Telephone Network, or the traditional circuit-switched
telephone network. This is the system that has been in general use since the late 1800s.
Using underground copper wires, this legacy platform has provided businesses and households
alike with a reliable means to communicate with anyone around the world for generations.
The phones themselves are known by several names, such as PSTN, landlines, Plain Old
Telephone Service (POTS), or fixed-line telephones.
PSTN phones are widely used and generally still accepted as a standard form of communication.
However, they have seen a steady decline over the last decade. In fact, there are currently
just 972 million fixed-line telephone subscriptions in use worldwide, the lowest tally this
century so far.

6
 Chapter 2: LITERATURE REVIEW

2.1 INTRODUCTION

Voice over IP (VoIP) is the name of a new emerging voice telecommunications technology that
is transforming telephony. This technology marks a turning point in the world of
communication by making it possible to transmit voice over a digital network and the Internet.
In 1996, the first version of Voice over IP was born, called H323. Since then, Voice over IP
technology has progressed as companies have discovered its advantages in increasing the
productivity and efficiency of their networks. The objective of Voice over IP is to apply to voice
the same treatment as other types of data circulating on the Internet. Thanks to the IP protocol,
data packets, consisting of digitized voice, are transported there. Indeed, by dint of transferring
information files in increasingly real time, Internet users came to transfer voice, in real time
enough to compete on the telephone. In this trivialization of voice data, two major constraints
are present: transmitting these packets in the right order and doing so within a reasonable time.
IP telephony and mobile telephony, two technologies set to become widespread over the next
few years, will have a major impact on the way people communicate, both in the office and at
home. The objective of this chapter is the study of this technology and its different aspects. We
will discuss in detail the architecture of VoIP, its elements, its operating principle and the main
advantages and disadvantages of IP telephony.

2.2 GENERAL STUDY OF VOIP

2.2.1 DEFINITION

VOIP stands for Voice Over Internet Protocol or voice over IP, is a technique that allows
communication by voice over IP-compatible networks, whether private networks or the
Internet, wired or not. It concerns the transport of voice over an IP network.

2.2.2 WORKING PRINCIPLE

7
Unlike traditional circuit-switched telephony, which relies exclusively on a switched telephone
network, VoIP technology makes it possible to telephone on specialized or wireless networks,
including computer networks. These new types of networks use "packet switching" protocols.
In addition to the voice data (digitized voice), a packet includes the network addresses of the
sender and the recipient. VoIP packets are transmitted through any VoIP compatible network
and can be routed by different paths: VoIP is therefore interoperable. Subsequently, an
application will take care of the reverse transformation (from packets to voice). "In simpler
terms, you pick up, dial, and the call goes through the Internet rather than through traditional
channels. Not to mention any features like the link between voicemail and the computer."
Indeed, all the information to be transmitted on the network is divided into data packets. Each
package consists of:
A header indicating its source and destination
From a sequence number
From a data block
An error checking code

Routers and servers route these packets through the network to their destination. When the
packets arrive at their destination, the sequence number allows the packets to be reordered in
the original order. Unlike PSTN telephony which dedicates a circuit to a telephone call, data
packets share a circuit with other transmissions. Indeed, Bertrand Chauvet, Director of Business
and International Development at NetCentrex, explains that circuit telephony, like the
traditional switched telephone network (PSTN), "... consists of opening a communication
channel between two people and reserving the "all of that bandwidth has those two
parties...even if you're not speaking, bandwidth is used and it's wasted whether it's analog or
digital." In "packet" mode telephony, a session - and not a connection - is established between
two users. Depending on the available protocols, LAN-based telephony uses VoIP (Voice over
IP) or ATM (Asynchronous Transfer Mode) to transmit voice calls over the LAN. The
connection to the traditional telephone system is provided by a PSTN (Public Commute
Telephone Network) gateway on a server. LAN-based phone systems are useful for both small
and large organizations. The physical location of telephony devices does not matter. In fact, a
LAN-based telephony system works perfectly in a corporate environment where remote sites
are connected to the main office via a wide area network (WAN). The telephony resources
available at a specific point may be available over the wide area network. In addition, IP
telephony makes it possible to combine the telephone and the computer, thanks to an integrated
8
infrastructure based on the Internet protocol (IP). In this way, it is possible to process and
transmit on the same infrastructure communications of different types, whether voice, data,
images or video. This new technology could allow an organization to merge, on a single
network, the computer network and the switched telephone network. In addition, IP telephony
enables the most popular features of the PABX traditional (Private Automatic Branch eXchange
or private telephone exchange). These include:
Call forwarding (all, on busy, on no answer)
Call waiting.
Display of caller's number and name.
Distinctive ringtone (internal vs. external call).
Message waiting indicator.
Conference and transfer.

2.2.3 GENERAL ARCHITECTURE OF VOIP

Figure 2.1: General architecture of VoIP

2.3 VoIP protocols

9
 2.3.1 Signalling protocols

2.3.1.1 H.323 STANDARD

This is the ITU-T (International Telecommunications Union) standard that vendors should
comply while providing Voice over IP service. This recommendation provides the technical
requirements for voice communication over LANs while assuming that no Quality of Service
(QoS) is being provided by LANs. It was originally developed for multimedia conferencing on
LANs, but was later extended to cover Voice over IP. The first version was released in 1996
while the second version of H.323 came into effect in January 1998. The standard encompasses
both point to point communications and multipoint conferences. The products and applications
of different vendors can interoperate if they abide by the H.323 specification.

2.3.1.1.1 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. These are
discussed below:

Terminals

These are the LAN client endpoints that provide real time, two way communications. All H.323
terminals have to support H.245, Q.931, Registration Admission Status (RAS) and Real Time
Transport Protocol (RTP). H.245 is used for allowing the usage of the channels, Q.931 is
required for call signaling and setting up the call, RTP is the real time transport protocol that
carries voice packets while RAS is used for interacting with the gatekeeper. These protocols
have been discussed later in the paper. H.323 terminals may also include T.120 data
conferencing protocols, video codecs and support for MCU. A H.323 terminal can
communicate with either another H.323 terminal, a H.323 gateway or a MCU

Gateways

An H.323 gateway is an endpoint on the network which provides for real-time, two-way
communications between H.323 terminals on the IP network and other ITU terminals on a

10
switched based network, or to another H.323 gateway. They perform the function of a
"translator" i.e. they perform the translation between different transmission formats, e.g. from
H.225 to H.221. They are also capable of translating between audio and video codecs. The
gateway is the interface between the PSTN and the Internet. They take voice from circuit
switched PSTN and place it on the public Internet and vice versa. Gateways are optional in that
terminals in a single LAN can communicate with each other directly. When the terminals on a
network need to communicate with an endpoint in some other network, then they communicate
via gateways using the H.245 and Q.931 protocols.

Gatekeepers

It is the most vital component of the H.323 system and dispatches the duties of a "manager". It
acts as the central point for all calls within its zone (A zone is the aggregation of the gatekeeper
and the endpoints registered with it) and provides services to the registered endpoints. Some of
the functionalities that gatekeepers provide are listed below:
Address Translation: Translation of an alias address to the transport address. This is done using
the translation table which is updated using the Registration messages.
Admissions Control: Gatekeepers can either grant or deny access based on call authorization,
source and destination addresses or some other criteria.
Call signaling: The Gatekeeper may choose to complete the call signaling with the endpoints
and may process the call signaling itself. Alternatively, the Gatekeeper may direct the endpoints
to connect the Call Signaling Channel directly to each other.
Call Authorization: The Gatekeeper may reject calls from a terminal due to authorization failure
through the use of H.225 signaling. The reasons for rejection could be restricted access during
some time periods or restricted access to/from particular terminals or Gateways.
Bandwidth Management: Control of the number of H.323 terminals permitted simultaneously
access to the network. Through the use of H.225 signaling, the Gatekeeper may reject calls from
a terminal due to bandwidth limitations.
Call Management: The gatekeeper may maintain a list of ongoing H.323 calls. This information
may be necessary to indicate that a called terminal is busy, and to provide information for the
Bandwidth Management function.

Multipoint Control Units (MCU)

11
The MCU is an endpoint on the network that provides the capability for three or more terminals
and gateways to participate in a multipoint conference. The MCU consists of a mandatory
Multipoint Controller (MC) and optional Multipoint Processors (MP). The MC determines the
common capabilities of the terminals by using H.245 but it does not perform the multiplexing
of audio, video and data. The multiplexing of media streams is handled by the MP under the
control of the MC. The following figure [Fig1] shows the interaction between all the H.323
components.

Figure 2.2: Components of H.323

2.3.1.2 SESSION INITIATION PROTOCOL (SIP)

This is the IETF standard for establishing VOIP connections. It is an application layer control
protocol for creating, modifying and terminating sessions with one or more participants. The
architecture of SIP is similar to that of HTTP (client-server protocol). Requests are generated
by the client and sent to the server. The server processes the requests and then sends a response
to the client. A request and the responses for that request make a transaction. SIP has INVITE
and ACK messages which define the process of opening a reliable channel over which call
control messages may be passed. SIP makes minimal assumptions about the underlying
transport protocol. This protocol itself provides reliability and does not depend on TCP for
reliability. SIP depends on the Session Description Protocol (SDP) for carrying out the
negotiation for codec identification. SIP supports session descriptions that allow participants to
agree on a set of compatible media types. It also supports user mobility by proxing and
redirecting requests to the user’s current location. The services that SIP provide include:

• User Location: determination of the end system to be used for communication

• Call Setup: ringing and establishing call parameters at both called and calling party

12
 • User Availability: determination of the willingness of the called party to engage in
communications
• User Capabilities: determination of the media and media parameters to be used
• Call handling: the transfer and termination of calls

2.3.1.2.1 Components of SIP

The SIP System consists of two components

User Agents:

A user agent is an end system acting on behalf of a user. There are two parts to it: a client and
a server. The client portion is called the User Agent Client (UAC) while the server portion is
called User Agent Server (UAS). The UAC is used to initiate a SIP request while the UAS is
used to receive requests and return responses on behalf of the user.
Network Servers:

There are 3 types of servers within a network. A registration server receives updates concerning
the current locations of users. A proxy server on receiving requests, forwards them to the next-
hop server, which has more information about the location of the called party. A redirect server
on receiving requests, determines the next-hop server and returns the address of the next-hop
server to the client instead of forwarding the request.
2.3.1.2.2 SIP Messages (requests) and these responses

SIP defines a lot of messages. These messages are used for communicating between the client
and the SIP server. These messages are:

INVITE: for inviting a user to a call

BYE: for terminating a connection between the two end points

ACK: for reliable exchange of invitation messages

OPTIONS: for getting information about the capabilities of a call

REGISTER: gives information about the location of a user to the SIP registration server.

CANCEL: for terminating the search for a user

13
 2.3.1.2.3 Responses to these SIP queries:

A response to a request is characterized by a code and a reason, called status code and reason
sentence respectively. A status code is a 3-bit integer indicating a result after receiving a request.
This result is specified by a sentence, text based (UTF-8), explaining the reason for refusing or
accepting the request. The status code is therefore intended for the automaton managing the
establishment of sessions SIP and patterns to programmers. There are 6 classes of responses
and therefore of status codes, represented by the first bit:

1xx: Information - The request has been received by the recipient and continues to
be processed (ex: 180 = 'ringing')
2xx: Success (ex: 200='OK', 202='accepted')
3xx: Redirect - Another action must take place in order to validate the request
4xx: Client error - The request contains incorrect syntax or it cannot be processed
by this server (ex: 404 = 'Not found')
5xx: Server Error - The server failed to process a request that seems to be correct.
6xx: General failure, the request cannot be processed by any server

2.3.2 Comparison between H.323 and SIP

Comparison of SIP and H.323 Protocols. Both H.323 and SIP are protocols for supporting VoIP
technology, but their usage and design are quite different. H.323 focuses on replacing circuit
lines in existing Telecom grade telephone networks with IP lines. SIP is more likely to use the
IP phone as an application on the Internet. In addition, based on the next generation network
(NGN), which is gradually entering people’s mind, more widely adopted large-scale integrated
IP products and IP gateways in the network, so that end-to-end systems can adopt IP to achieve
pure IP Business Applications. At this time, IP-based SIP protocol will show great advantages.
At the same time, SIP also provides good QoS support, enabling SIP to carry out high-quality
transmission over the IP network. In combination with the above analysis, the performance of
the two protocols can be compared and summarized as shown in Table 2.1
Tableau 2.1: Comparison between H.323 and SIP

S.NO H.323 SIP

1. H.323 is monolithic architecture. SIP is modular architecture.

14
 S.NO H.323 SIP
2. The scalability of H.323 is limited. SIP is better scalable.
3. H.323 is a little bit flexible. It is more flexible.
H.323 does not provide the facility of
4. SIP provides the facility of instant messaging.
instant messaging.
H.323 is absolute complex in terms of
5. It is moderate complex in terms of complexity.
complexity.
The message format of H.323 is in binary While the message format of sip is in ASCII
6.
form. format.
7. It is not compatible with internet. While it is compatible with internet.
H.323 is built entirely on telephone While SIP completely depends on internet
8.
systems. connection.
9. It is quite compatible with PSTN. It is not compatible with PSTN.
10. It was designed by ITU. It was designed by IETF.
For the location of the endpoint, it is For the location of the endpoint, it uses SIP
11.
mapped by gatekeepers. Uniform Resource Locator link.
12. It has limited services. It provides better services.

H.323 and SIP are specifically known for the IP signalling standards. The H.323 and SIP
describe multimedia communication systems and protocols. These protocol suites differ in
many ways. Essentially, H.323 is derived by ITU before the advent of SIP while SIP is
acknowledged by IETF standard.

2.3.3 Transport protocols

We describe two transport protocols used in voice over IP namely RTP and R TCP
2.3.3.1 The RTP protocol

2.3.3.1.1 General description of RTP

15
RTP (Real time Transport Protocol), standardized in 1996, is a protocol which was developed
by the IETF in order to facilitate the end-to-end real-time transport of audio and video data
streams on IP networks, c i.e. on packet networks. RTP is an application-level protocol that
uses the underlying transport protocols TCP or UDP. But the use of RTP is usually done at-
over UDP which can makes it easier to reach real time. Real-time applications such as digital
speech or videoconferencing constitute a real problem for the Internet. Who says real-time
application, says the presence of a certain quality of service (QoS) that RTP does not guarantee
because it operates at the Application level. Moreover RTP is a protocol which is in a multipoint
environment, therefore one can say that RTP has in its charge, the management of the real time,
but also the administration of the multipoint session.

2.3.3.1.2 The functions of RTP

The purpose of the RTP protocol is to organize the packets entering the network and to control
them at the exit. This in order to reform the flows with its initial characteristics. RTP is an end-
to-end protocol, intentionally incomplete and malleable to adapt to the needs of applications. It
will be integrated into the core of the application. It leaves the control responsibility to the end
devices. It is also a protocol suitable for applications with real-time properties. It thus allows
to:
Set up a sequencing of packets by numbering in order to allow the detection of lost
packets. This is a key point in data reconstruction. But you should still know that
the loss of a packet is not a big problem if the packets are not lost in too large
numbers. However, it is very important to know which packet has been lost in order
to be able to compensate for this loss.
Identify the content of the data to associate them with a secure transport and
reconstitute the time base of the flows (timestamp of the packets: possibility of
resynchronization of the flows by the receiver)
The identification of the source, i.e. the identification of the sender of the packet. In
a multicast the identity of the source must be known and determined.
Transport audio and video applications in frames (with dimensions that are
dependent on the codecs that perform the digitization). These frames are included
in packets in order to be transported and must, therefore, be easily recovered at the
time of the packet segmentation phase so that the application is decoded correctly.

16
 2.3.3.2 The RTCP protocol

2.3.3.2.1 General description of RTCP

The RTCP protocol is based on the periodic transmission of control packets to all participants
of a session. It is the UDP protocol (for example) which allows the multiplexing of RTP data
packets and RTCP control packets. The RTP protocol uses the RTCP protocol, Real-time
Transport Control Protocol, which carries the following additional information for session
management. Receivers use RTCP to send a QoS report back to senders. These reports include
the number of lost packets, the parameter indicating the variance of a distribution (more
commonly known as jitter: i.e. packets that arrive regularly or irregularly) and the round-trip
delay. This information allows the source to adapt, for example, to modify the level of
compression to maintain a QoS. Among the main functions offered by the RTCP protocol are
the following:
Additional synchronization between media: Multimedia applications are often
transported by separate streams. For example, voice, image or even digitized
applications on several hierarchical levels can see managed streams and follow different
paths.
Identification of participants in a session: indeed, RTCP packets contain address
information, such as the address of an electronic message, a telephone number or the
name of a participant in a conference call.
Session control: in fact, the RTCP protocol allows participants to indicate their
departure from a conference call (RTCP Bye packet) or simply to provide an indication
of their behavior. The RTCP protocol asks session participants to periodically send the
information mentioned above.
The periodicity is calculated according to the number of participants of the application.
It can be said that RTP packets carry only user data. While RTCP packets only carry
real-time supervision. We can detail the monitoring packages in 5 types:
SR (Sender Report): This report gathers statistics concerning the transmission
(percentage of loss, cumulative number of lost packets, delay variation (jitter), etc.).
These reports are from active senders in a session.
RR (Receiver Report): Set of statistics relating to communication between participants.
These reports come from the receivers of a session.

17
 SDES (Source Description): Source business card (name, e-mail, location).
BYE: Message of end of participation in a session.
APP: Application-specific functions.

2.3.4 Advantages and disadvantages of VoIP

2.3.4.1 Advantages

Lower Costs

The bottom line is vital for every business, large or small. So, you have to consider every cost-
saving opportunity. One way companies can realize significant cost savings is by adopting a
VoIP phone system.
Cost savings in VoIP come in two ways: direct and indirect.
 Direct Cost Savings

When it comes to traditional phone service, a business incurs massive initial costs. Especially
in the name of business phones and PBX hardware.

• PBX Costs

A PBX (private branch exchange) is an on premise physical piece of hardware. It connects

many landline phones in an office and can cost a huge sum of money. We are talking tens of
thousands of dollars. An amount you can amortize over several years.
You may argue that analog phones cost about the same as IP phones. The exact price will differ
based on the desired features.

• Copper Wiring Charges

Broadband connections also do away with the extra wiring because VoIP networks allow both
voice and data on the same channel.

• Calling Expenses

Direct costs also come in the form of the cost of calling. VoIP calls are cheaper compared to
the Public Switched Telephone Network (PSTN) or the traditional circuit-switched telephone
network by a stretch.
18
  Indirect Cost Savings

Indirect savings are more difficult to quantify, but that doesn’t make them any less critical for
your business. Below are some of the most common areas where organizations save money
long-term.

• Savings with Remote Work

Switching to VoIP lets employees stay connected to the corporate phone system while working
remotely. This is thanks to the long list of VoIP phone features like call waiting, auto-attendant,
instant video calling, conference calling, and others not provided by traditional phones.

• Add-On Features at No Extra Cost

VoIP phone services include many of these features at no additional cost. No need to pay
extra for whatever feature you think could be useful for your network.

Increased Accessibility

Accessibility is one of the biggest benefits of VoIP for business. If you have a decent data
connection, you can make and receive calls for your business. And when you’re unable to
answer the call, you can direct calls to another person or get voicemails emailed to you.

Complete Portability

VoIP is not distance or location-dependent, is completely portable. This means you can use the
same number wherever you go.

Scalability

VoIP gives businesses the potential for high-quality, large-scale conference calls. Users can be
easily added or removed to your network as and when required

Supports Multitasking

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Along with traditional phone calls, VoIP allows you to send documents, images, and videos all
while simultaneously engaging in a conversation. So you can seamlessly hold more integrated
meetings with clients or staff from other corners of the globe.

Easy to install

Unlike traditional phone systems that require a local phone company to set up and install a
physical PBX system within your office premises, a VoIP system is much easier to install.

All you have to do is download an app or work with your VoIP software provider to port over
your existing business number.

No hardware needs to be installed, which can be challenging to manage since the phone
company technicians need to be physically present.

2.3.4.2 Disadvantages

Latency

Problems with latency can affect the quality of calls made through the VoIP system, particularly
if you’re relying on a shared internet connection. However, if you have a private internet
connection or a dedicated business broadband, the quality of calls should not suffer. By opting
for a leased line or quality ISP that provides high bandwidth availability, the risk of low-quality
sound is mitigated.

Emergency situations

If there is a loss of power during an emergency, you won’t be able to make an urgent call
using the VoIP system. This means businesses are often required to have a back-up landline
telephone for emergency situations, as the local telephone exchange network is not affected
by power outages. However, with most people now having mobile phones, the likelihood
of being unable to contact an emergency operator is low.

VoIP Needs a Fast and Reliable Internet Connection

Compared to a traditional phone line, VoIP relies entirely on an internet connection to make
calls. So, if your internet is down or incredibly slow, you won't be able to use VoIP properly.
20
 Susceptibility to cyber attacks

Security is paramount to internet communications. The possibility of hackers listening,

distorting, or forwarding your calls for malicious purposes should be taken seriously.

A small handful of providers have not yet implemented stronger security measures to protect
their private VoIP calls from getting breached during transmission.

Fortunately, most VoIP vendors use high-level end-to-end encryption strategies to ensure that
your data is always secure from intruders. Be sure to do your due diligence and find the best
VoIP service providers before committing.

2.4 Related works

Several studies have been carried out to design and implement a voice over internet protocol
solution. Some of these studies includes;
(Nitthita Chirdchoo et al., 2013). Improved how Voice over Internet Protocol (VoIP)
technology has continually gained popularity and has been widely adopted for personal and
enterprise usage over the past decade by proposing new VoIP architecture for a campus usage.
He also improved how the system’s performance is evaluated in terms of voice quality and the
sufficiency of the number of trunks required, which reveals that the designed system operates
with satisfactory performance.
A study (Nitthita Chirdchoo et al. (2016). Improved that Voice over IP (VoIP) is a technology
that is increasingly recognized in all sectors, it is to channel telephone calls through the IP
network. Today, this technology is increasingly deployed in institutions for the Technology,
Education and Research in Western countries, telemedicine and interconnection of
administrative services, education and health. This project is carried out by the will of the
Aviation Authority is equipped with a VoIP phone system to reduce the cost of communication
was too expensive.
This paper (“Design and Implementation of a VoIP PBX Integrated Vietnamese Virtual
Assistant: A Case Study”, 2023), improved how the application of Artificial Intelligence (AI)
technology into the Private Branch Exchange (PBX) has played a pivotal role in enhancing the
customer experience and is able to unite employees in any company. This article also compare
VoIP with a traditional PBX, analyzes, evaluates and optimizes an automatic PBX system with
integrated VVA, thereby offering efficient solutions for interest companies.
21
Most of the above studies were done between two ends points or networks and focused on the
communication between two ends points and no security measure was taken into consideration
to protect the company or campus network from unauthorized access or user

Therefore, we planed to present a four remote networks and evaluate the influence of VoIP over
traditional phones. Moreover, we placed the emphasis on voice call and some security
measures.

2.5 Conclusion

As we have seen throughout this chapter, VoIP is the most cost-effective solution for carrying
out conversations. Currently it is obvious that VoIP will continue to evolve. IP telephony is a
good solution in terms of integration, reliability and cost. We have seen that voice over IP being
a new communication technology, it does not yet have a single standard. Each standard has its
own characteristics to guarantee a good quality of service. Indeed, respecting time constraints
is the most important factor when transporting voice. Although standardization has not reached
sufficient maturity for its general realization at the level of IP networks, it is not dangerous to
bet on these standards since they have been accepted by the entire telephony community.
Finally, when implementing this technology, we must ask the question next: does the
development of this technology represent a risk or an opportunity for users and telephone
operators?

22
 Chapter 3: Materials and Methods

3.1 Introduction

In order to ensure the accuracy and reliability of the research findings, it is essential to have a
well-defined methodology. This chapter provides a detailed account of the steps taken to carry
out the research, including the tools and techniques utilized to collect and analyse data. This
section therefore serves as the foundation for the dissertation, offering a comprehensive
overview of the technical aspects of the research, including the materials and methods used to
collect and analyse data. It also highlights the reasons why certain methods were chosen over
others, providing by presenting a thorough overview of the methodology, this section enables
readers to understand the research process and the validity of the findings.

3.2 MATERIALS

In this section, we will look at the materials used in the implementation of this project.
Software materials ranging from tools, IDE, equipment and hardware materials.

3.2.1 Integrated Development Environment (IDE)

Cisco packet tracer, as the name suggests, is a tool built by Cisco. This tool provides a network
simulation to practice simple and complex networks. Packet Tracer is a cross-
platform visual simulation tool designed by Cisco Systems that allows users to create network
topologies and imitate modern computer networks. The software allows users to simulate the
configuration of Cisco routers and switches using a simulated command line interface. Packet
Tracer makes use of a drag and drop user interface, allowing users to add and remove simulated
network devices as they see fit. The software is mainly focused towards Cisco Networking
Academy students as an educational tool for helping them learn fundamental CCNA concepts.

3.2.2 Tools used

23
 • Cisco router

It’s a networking device operating at layer 3 or a network layer of the OSI model. They are
responsible for receiving, analysing, and forwarding data packets among the connected computer
networks. When a data packet arrives, the router inspects the destination address, consults its
routing tables to decide the optimal route and then transfers the packet along this route.

Figure 3.3: Cisco router

• Cisco switch 2960 series

The Switch is a network device that is used to segment the networks into different subnetworks
called subnets or LAN segments. It is responsible for filtering and forwarding the packets
between LAN segments based on MAC address.
Switches have many ports, and when data arrives at any port, the destination address is
examined first and some checks are also done and then it is processed to the devices.

Figure 3.4: Cisco switch

• IP phone

Also called VoIP phone, refers to any phone system that uses internet connection to send and
receive data. Unlike a regular telephone that uses landlines to transmit analog signals, IP phones

24
connect to internet via router and modem. A VoIP system converts analog voice signals into
digital signals over the Broadband connection.

Figure 3.5: Cisco IP phone

3.2.3 Hardware
In order to carry out this project, we used a Dell laptop with the following configuration
• Processor Intel Core i7 2670QM CPU 2.3 GHZ (64 bits).
• 8 Go RAM.
• Hard drive: 500Go.
• Operating system: Windows 10 Professional (64bit)

3.3 Prototype

Setting up a network is the final step in many processes including hardware purchasing,
topology planning, and addressing plan. But, before any full-scale implementation, a prototype
or sketch of the topology must be made to ensure the feasibility of the project. For this case, we
used simulation software called Cisco Packet Tracer. The following image shows us the
network prototype.

25
Figure 3.6: Prototype

3.4 METHODS

3.4.1 Bases security configuration on the equipment

The definition of a security policy in a computer network is a significant and even crucial aspect
for its proper functioning. Regarding the site of password configurations in privileged and user
execution mode, it is important to note that security configuration helps to ensure the integrity
of the equipment itself. The following images illustrate the configurations made on the
equipment.

NB: To configure a Cisco router/switch, you must first open your emulator which is an
application that provides the graphical interface to connect to the router. Once the router is
turned on, we see the writings “router>” which shows that we are in “user execution mode”. In
this mode, no configuration can be performed. The next mode is the “privileged execution
mode” you are only allowed to a limited number of basic monitoring: verification commands
for example. We recognize this mode with the sign “router#”. To switch from user EXEC mode
to privileged EXEC mode, you execute the “enable” command. We finally have the last mode
which is the global configuration mode. It is in this mode that the operation of the equipment

26
can be modified. It is recognized by the sign “router (config) #”. To switch from privileged
EXEC mode to global configuration mode, you execute the “configure terminal” command.

• Rename the equipment /device

To rename a Cisco device means to change the name that appears in the device's hostname. The
hostname is the name that identifies the device on the network. By default, Cisco devices are
given a hostname based on their model number, such as "Switch" or "Router". Renaming a
Cisco device is useful for several reasons, such as to make it easier to identify the device on the
network or to give it a more meaningful name that reflects its location or purpose.

Figure 3.7: Rename the device

The definition of a security policy in a computer network is a significant and even crucial
aspect for its proper functioning. With regards to the site of user and privileged execution

• Password user mode

To set a password for user mode you configure the console port to require local authentication
using the following command in global configuration mode:

Figure 3.8: Password user mode

• Password privileged mode

To set a password for privileged mode, you use the following command in global configuration
mode: “enable secret” followed by the desired password
27
Figure 3.9: Password Privileged mode

• Passwords encryption

This help to encrypt passwords of the previous which means putting in the form that can not
be readable

Figure 3.10: Passwords encryption

3.4.2 Switches configuration

To configure our switches, two mode are going to be configure on the specific ports
Access mode (ports 2&3)
Trunk mode (port 1)
Using the “Switchport mode access” command forces the port to be an access port while and
any device plugged into this port will only be able to communicate with other devices that are
in the same VLAN.
Using the “Switchport mode trunk” command forces the port to be trunk port. C-a-d Puts the
interface into permanent trunking mode and negotiates to convert the neighboring link into a
trunk link. The interface becomes a trunk interface even if the neighboring interface is not a
trunk interface.
The images below illustrate the configuration of these switches in a detailed and hierarchical
fashion.

Site A

28
Figure 3.11: Switch configuration site A

Site B

Figure 3.12: Switch configuration site B

Site C

29
Figure 3.13: Switch configuration site C

Site D

Figure 3.14: Switch configuration site D

3.4.3 Routers configuration

3.4.3.1 Addressing and activation of the interface f0/0

30
To activate and configure an interface for the local network, you need to assign it an IP address
and subnet mask that are compatible with the network, and then activate the interface. This
allows the device to communicate with other devices on the local network.
Once the interface is activated, you also need to configure other settings such as default gateway
to ensure proper connectivity to the local network and the internet.

Site A

Figure 3.15: Addressing and activation of the interface f0/0 site A

Site B

Figure 3.16: Addressing and activation of the interface f0/0 site B

31
 Site C

Figure 3.17: Addressing and activation of the interface f0/0 site C

Site D

Figure 3.18: Addressing and activation of the interface f0/0 site D

3.4.3.2 DHCP configuration

DHCP (Dynamic Host Configuration Protocol) is a network protocol whose role is to ensure
the automatic configuration of the IP parameters of a station or machine, in particular by
automatically assigning it an IP address and a subnet mask. . DHCP provides a solution to these
three drawbacks:
32
 Only working computers use an address from the address space;
Any modification of the parameters (gateway address, name servers) is passed
on to the stations during the restart
The modification of these parameters is centralized on the DHCP servers.
Using a centralized DHCP server allows companies to manage all IP address assignments
dynamically from a single server. This practice helps optimize IP address management and
ensures enterprise-wide consistency.

Site A

Figure 3.19: DHCP configuration site A

Site B

Figure 3.20: DHCP configuration site B

Site C

33
 Figure 3.21: DHCP configuration site C

Site D

Figure 3.22: DHCP configuration site D

3.4.3.3 Call manager or telephony service configuration

The steps are shown below:

a. From global configuration mode, enter telephony service mode.

Router(config)# telephony-service

b. To see the maximum allowable number of ephones (another name for IP phones)
that this system use enter the command max-ephones?
c. In the space provided, write the maximum number of phones supported by the
current router:

Router(config-telephony)# max-ephones ?

d. Set the maximum number of IP phones to two, as this will be sufficient for this
project.
34
Router(config-telephony)# max-ephones 2

e. To see the maximum allowable number of directory numbers that this system can
use enter the command max-dn ?

Router(config-telephony)# max-dn ?

f. In the space provided, write the maximum number of directory numbers:

Router(config-telephony)# max-dn 2 for this project

Site A

Figure 3.23: Telephony service site A

Site B

Figure 3.24: Telephony service site B

Site C

35
 Figure 3.25: Telephony service site C

Site D

Figure 3.26: Telephony service site D

3.4.3.4 Phones number assignation

We will now assign numbers to our two IP Phones which will be assigned automatically.
Enter the configuration of the 1st telephone number.

ephone-dn 1

Set an (internal) phone number of your choice.

number 2001

Repeat the manipulation with the “ephone-dn 2” and the number “2002“

ephone-dn 2

36
 number 2002

Site A

Figure 3.27: Phones numbers assignation site A

Site B

Figure 3.28: Phones numbers assignation site B

Site C

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 Figure 3.29: Phones numbers assignation site C

Site D

Figure 3.30: Phones numbers assignation site D

3.4.3.5 Addressing and activation of the serial interface 0/2/0

To activate and configure an interface on a Cisco device to communicate with a remote network,
you need to assign it an IP address that is compatible with the remote network, and then activate
the interface. Additionally, you need to configure the default gateway on the device to specify
the IP address of the next-hop router that will forward packets to the remote network.

Site A

38
Figure 3.31: Addressing and activation of the serial interface 0/2/0 site A

Site B

Figure 3.32: Addressing and activation of the serial interface 0/2/0 site B

Site C

Figure 3.33: Addressing and activation of the serial interface 0/2/0 site C

Site D

Figure 3.34: Addressing and activation of the serial interface 0/2/0 site D

39
 3.4.3.6 Routing protocol configuration (ospf)

Open Shortest Path First (OSPF) is a routing protocol used to exchange routing information
between routers in an IP network. OSPF is used in enterprise networks where multiple routers
are connected together.

Site A

Figure 3.35: OSPF configuration site A

Site B

Figure 3.36: OSPF configuration site B

Site C

Figure 3.37: OSPF configuration site C

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 Site D

Figure 3.38: OSPF configuration site D

3.4.3.7 Dial peer

Dial peers are virtual phone lines in a cisco voice over internet protocol (VOIP) network that
allow two endpoints to communicate. Dial peers allow VoIP calls to traverse the network from
one endpoint to another by establishing and routing calls. They are configured on cisco routers
and consist of a dial peer identifier, a destination pattern and destination or session target. The
destination pattern is used to match an incoming call and route it to the destination assigned to
the dial peer.

Site A

Figure 3.39: Dial peer configuration site A

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 Site B

Figure 3.40: Dial peer configuration site B

Site C

Figure 3.41: Dial peer configuration site C

42
 Site D

Figure 3.42: Dial peer configuration site D

3.4.4 Wan router configuration

3.4.4.1 Addressing and activation of the interfaces
3.4.4.1.1 Addressing and activation of the interface serial 0/0/0 to
connect to site A

Figure 3.43: Addressing and activation of the interface serial 0/0/0 to connect to site A

43
 3.4.4.1.2 Addressing and activation of the interface serial 0/0/1 to
connect to site B

Figure 3.44: Addressing and activation of the interface serial 0/0/1 to connect to site B

3.4.4.1.3 Addressing and activation of the interface serial 0/2/0 to

connect to site C

Figure 3.45: Addressing and activation of the interface serial 0/2/0 to connect to site C

3.4.4.1.4 Addressing and activation of the interface serial 0/2/1 to

connect to site D

44
 Figure 3.46: Addressing and activation of the interface serial 0/2/1 to connect to site D

3.4.4.2 Routing protocol (OSPF) configuration to connect all the sites

Figure 3.47: Routing protocol (OSPF) configuration

3.4.5 AAA configuration

AAA (Authentication, Authorization, Accounting) is a standard-based framework used to

control who is permitted to use network resources (through authentication), what they are
authorized to do (through authorization), and capture the actions performed while accessing
the network (through accounting).

Authentication
The process by which it can be identified that the user, which wants to access the network
resources, valid or not by asking some credentials such as username and password. Common
methods are to put authentication on console port.

Authorization

45
It provides capabilities to enforce policies on network resources after the user has gained
access to the network resources through authentication. After the authentication is successful,
authorization can be used to determine what resources the user is allowed to access and the
operations that can be performed.

Accounting
It provides means of monitoring and capturing the events done by the user while accessing
the network resources. It even monitors how long the user has access to the network. The
administrator can create an accounting method list to specify what should be accounted for
and to whom the accounting records should be sent.

Figure 3.48: AAA configuration

46
 Chapter 4: RESULTS AND DISCUSSION

4.1 RESULTS

After the different configurations, we will pass to the test phase which will be done in three
stages

4.1.1 Security test

Regarding the network security test, we performed a functional test of the security of the
equipment

Figure 4.49: Basis Security test

4.1.2 Calls tests

As for the communication test in the network, we carried out a series of connectivity tests
between the IP telephones. The following illustrations represent the different calls between
the phones

47
 4.1.2.1 Call from site A to site B

p
Figure 4.50: Call from site A to site B

4.1.2.2 Call from site A to site C

Figure 4.51: Call from site A to site C

48
 4.1.2.3 Call from site A to site D

Figure 4.52: Call from site A to site D

4.1.2.4 Call from site B to site C

Figure 4.53: Call from site B to site C

49
 4.1.2.5 Call from site B to site D

Figure 4.54: Call from site B to site D

4.1.2.6 Call from site C to site D

Figure 4.55: Call from site C to site D

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 4.1.3 AAA test

Here, we have to verify who has authorisation to access our network. The only person that can
access the network is the one having credentials for authentication

Figure 4.56: AAA test

4.2 Discussion

Voice over Internet Protocol (VoIP) works by converting voice data into digital data. VoIP
technology allows people to complete calls over an existing internet connection.
When users bypass the Public-Switched Telephone Network (PSTN), they supply the internet
connection. A typical broadband connection provides all the bandwidth needed for high-quality
phone calls.
As you plan your VoIP deployment, you’ll want to design with growth in mind. Planning ahead
is crucial when setting up a VoIP phone system.

51
Use the following checklist to help you assess your current needs while you plan for the future.

VoIP Phones

Identify the number of phones and which must-have features you need. Should everyone should
get the same type of phone? Are there certain users who want to receive a premium business
phone? Visit our VoIP phone guide to view top recommendations.

Network Connectivity

Calculate how much bandwidth you need by estimating 100 kbps per voice line. Always factor
in an extra 15% headroom for data overhead. Align your implementation plan with the
company’s staffing plan, so it’s growth-oriented.

Communication Costs

Understand how much the company spends on its existing telephone service. Consider these
costs as you look into an alternative business phone provider.
Switching to VoIP (or adding a SIP Trunk to your IP PBX) can save your organization up to
70% off its voice service.
Understand what your company actually spends on business communications. The entire
organization rarely feels these individual costs, but they add up. Gather invoices for all your
business apps so you can look at the bigger picture and trim expenses.

Number Portability

Identify any issues you may have with porting your number to a VoIP number. In most cases,
you will want to speak with your phone provider to understand what you need to port out your
number.
It takes about 2-4 weeks to migrate a number from a previous phone provider.

Get Familiar with the Portal (Control Panel)

Gain confidence in administering your VoIP phone system. People may ask you to add lines,
adjust features, and more. Check with your VoIP provider to adopt their recommendations.

52
Dedicate a couple of hours for this purpose so you will save you time when you need to make
adjustments later.

Develop Training Materials

Expect that people will ask you about common phone functions like accessing voicemail. It will
happen no matter how simple it is. Make your VoIP implementation successful by developing
a handy guide for staff.
Provide a slideshow showing significant features and functions. Think about their everyday
needs. This document empowers your staff to enjoy their phone system without any hiccups.
Provide a one-page guide to show how to manage voicemail, transfer calls, and use the hold
function.

Test to Discover and Fix Issues

Give yourself enough time to test your new VoIP business phone system. Identify any problems
upfront and troubleshoot VoIP issues right away.
If you find that you need to upgrade your broadband service, do it soon. It’s not uncommon for
service upgrades to take a couple of days to complete.
Consider testing your voice calls at various times throughout the workday. Testing like this
simulates real-world conditions so you can fix issues earlier.

53
 CHAPTER 5: CONCLUSIONS AND RECOMMENDATIONS

5.1 Conclusions

The development of Voice over Internet Protocol network is a complex process that must be
handled with great care. The network managers can hardly detect the numerous challenges that
affect data packets. The incorporation of a VoIP system into an already overworked network
may be detrimental to an organization. There is no approved solution to the challenges, which
affect voice data that is in transit via a VoIP network.
Organizations must use different solutions selectively according to the nature of VoIP system
that they use. Institutions can run secure VoIP systems; however, they have to incur a high cost
with respect to installing security measures. Until the experts come up with a reliable and
efficient method of running a VoIP network, organizations should operate their systems
cautiously. They should ensure that they acquire the correct software and establish the
appropriate network infrastructure.

5.2 Recommendations

Implementing VoIP requires you to follow certain criteria in order for it to function properly.
Below are the main recommendations you need to consider when implementing VoIP in your
business or company or organization

5.2.1 Prioritizing the data

VoIP traffic must always precede other kinds of data traffic because it is sensitive to delays and
interruptions. This can be accomplished by enabling Quality of Service (QoS) settings on the
network, which prioritize voice traffic over other kinds of traffic. Depending on the sort of
traffic and the capacity of the network, QoS settings can be set up to give various types of traffic
priority.

5.2.2 Wired over Wi-Fi

It’s advised to use a wired network connection when adopting VoIP technology instead of Wi-
Fi. This is due to the possibility of interference and signal strength affecting Wi-Fi connections,

54
which may result in dropouts and bad call clarity. Wired links provide a stronger, more
dependable connection that is essential for VoIP to function properly.

5.2.3 Maintain network security

VoIP communication is susceptible to security risks like hacking and eavesdropping.

Implementing security steps like firewalls, encryption, and internet protocol is crucial for
network security. Encryption can be used to safeguard the privacy of VoIP traffic, while
firewalls can be used to prevent unauthorized entry to the network. You need to ensure that the
session initiation protocol is meeting the required criteria.

5.2.4 Less latency

The delay between the time that a voice signal is transmitted and the time the receiver hears it
is called latency. Use a low-delay network to ensure there is as little latency as possible. A
network with fast connections and minimal packet loss rates can accomplish this. VoIP requires
low latency networks in order to function properly because they guarantee real-time voice data
transmission.

5.2.5 Network bandwidth

VoIP needs a certain quantity of bandwidth to function properly. The number of simultaneous
phone calls made and the connection quality determine how much bandwidth is needed. Make
sure there is sufficient bandwidth to support the number of concurrent contacts that will be
placed. Bandwidth can be raised by modernizing the network’s infrastructure, such as boosting
internet speed or implementing more effective codecs.
Implementing VoIP into practice necessitates carefully evaluating several factors, including
giving the data priority, using a wired network connection, ensuring network security, reducing
latency, and ensuring enough bandwidth. Adhering to these criteria guarantees a successful
VoIP implementation with uninterrupted and clear voice communication.

55
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