IRJET- Campus-Wide Internet Telephony Design and Simulation using Voice over Internet Protocol: A Case Study of Adamawa State University, Mubi, Nigeria

International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 06 | June 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 2844
Campus-wide Internet Telephony Design and Simulation using Voice
over Internet Protocol: A Case Study of Adamawa State University,
Mubi, Nigeria
Nathan Nachandiya1, Allahya Kwada2
1Research Lecturer, Dept. of Computer Science, Adamawa State University, Mubi, Nigeria
2B.sc, Dept. of Computer Science, Adamawa State University, Mubi, Nigeria
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Abstract - Voice over Internet protocol (VoIP) is a
technology also known as Internet protocol (IP) telephony
because it uses Internet protocols to make enhanced voice
communications possible. Internet protocols (IP) form the
root of IP networking, providing support to the public,
corporate, private, cable, and even wireless networks. VoIP
unites an organization’s many locations into a single
converged communication network. In light of the
objective, an efficient VoIP network which tackles Voice
communication issues was designed and developed for
Adamawa State University, Mubi. This study proposes a
Voice over Internet Protocol (VoIP) system that can help
users at the Adamawa State University campus to freely
communicate by using voice communication devices such
as the IP phone. Various sessions of the simulation were
run and configured using the Cisco packet tracer. It was
used to develop a prototype network architecture, giving
flexibility and ease to implement an efficient VoIP system.
Key Words: VoIP, QoS, Telephony, Codec, Cisco packet
tracer
1. INTRODUCTION
Nowadays, in light of the current developments in the
field of communication networks and due to the crucial
need to send data or information in a short period and at
the lowest possible cost, the VoIP technology was found.
It is the technology that transmits analog voice over a
digital network such as the Internet. VoIP increases
functionality and reduces costs because telephone calls
pass through a data network instead of the
telecommunication network of a company.
This technology is a successor to the public Switched
Transfer Network (PSTN) a connection-oriented, circuit
switched network that uses dedicated channels for
transmission. The PSTN runs over landlines or wires on
poles and underground. The inception of VoIP brings a
turnaround in communication technology in which we
no longer need to connect overhead wires over long
geographical location or distances to have access to
telephony service instead all we need now is just the
internet or our major network connection that can be in
a form of LAN, WAN, etc. Today VoIP is one among the
dominant technology in the communication world and
many organizations around the world are implementing
the technology while some are re-engineering the
traditional PSTN they have been using for years into
VoIP.
VoIP is built on open infrastructure allowing various
vendors to, provide applications and access, unlike the
public switched network, whose infrastructure is a
closed system. The PSTN technology involves vendors
only building applications specific for their equipment
and its framework hasn’t made it possible for vendors to
develop new applications for it; VoIP allows the
development and design of more creative applications as
well as the convergence of data, voice, and video in one
channel. Consequently, it is expected that the VoIP may
completely replace the circuit switched PSTN system in
the future.
Because of the efficient bandwidth and minimum cost
that VoIP technology offers, in which both voice and
data communication can be run on a single network,
several organizations ranging from small businesses to
large enterprises not excluding universities or colleges,
are adapting and deploying the technology and it has
proved to be useful in enhancing communication and
distribution of information. Currently Adamawa State
University, Mubi do not operate the VOIP technology and
the benefits of an implemented campus-wide internet
telephony are numerous. And since the university has a
good number of departments and offices, VOIP can be
used to promote inter-departmental communication and
likewise that of the offices.
This study aims at designing and simulating a VOIP-
based telephony network using the Cisco packet tracer
network simulator, for the university communication
system. I believe that if we can successfully implement
the simulation, given the advantages of VOIP, a modern
telephony system can be implemented on campus.
2. LITERATURE REVIEW
VoIP is often referred to as IP telephony (IPT) because it
uses Internet protocols to make enhanced voice
communications possible. Internet protocols are the

basis of IP networking, which supports corporate,
private, public, cable, and even wireless networks [1].
VOIP uses packet switching because circuit switching is
not the most preferred mechanism that is chosen by data
networks. Another reason is that the speed of the
internet connection would decrease by a great amount if
it had to maintain a continuous connection to the web
page that is being viewed at any given time. So as an
alternative, data networks simply send and recover data
as needed. Also, instead of choosing to route the data
over a dedicated channel, the packets of data flow
through a hectic network that consists of various
possible routes [2]. In the past, the circuit switching
process was the most widely used to build a
communications network it was used for ordinary
telephone calls and allowed the sharing of
communication equipment and circuits among users.
The connection was established first between source and
destination and after the transfer of information, it was
terminated [2]
Public switched telephone networks (PSTN) is a
traditional phone network system, using a circuit
switching mechanism for voice transmissions. Basically,
in circuit switching, resources are reserved along the
entire communication channel for the duration of the
call, whereas in packet switching, information is digitally
transmitted into one or more packets [2].
2.1 VoIP REAL-TIME PROTOCOLS
The protocols used to send real-time streams of data
across a network are called real-time protocols. Real-
time protocols deliver audio and video over IP networks.
RTP (Real Time Protocol) and RTCP (Real Time Control
Protocol) comes under VoIP real-time protocols that run
on the top of the User Datagram Protocol [3].
2.1.1 Real-Time Protocol (RTP)
Real-Time Protocol (RTP) a network protocol for
delivering audio and video over IP networks. RTP is used
extensively in communication and entertainment
systems that involve media, such as telephony, video
teleconference. RTP is used in alliance with a signaling
protocol that assists in build-up connections across the
network [4].
2.1.2 Real-Time Control Protocol (RTCP)
Real-Time Control Protocol (RTCP) is a control protocol
and works in combination with RTP. RTCP provides
Quality of Service (QoS) feedback and session
information [5]. RTCP can monitor the fraction lost,
jitter, packet loss and one-way delay [4].
RTCP allows participants to indicate that they are
leaving a session with the use of the BYE packet. It
partners with RTP in the delivery and packaging of
Multimedia data but does not transport any media data
itself [3].
2.2 Related Work
[6] Has demonstrated a survey on VOIP over WLAN, its
advantages and challenges and also VOIP capacity over
WLAN and the number of calls for different voice codecs
and intervals based on IEEE 802.11b standard.
[7] Evaluates the quality of service of video transmission
on Differentiated Services (Diff-Serv) with Multiprotocol
Label Switching (MPLS) network is being simulated. The
objective of this work is to study the influence of the QOS
mechanism via DiffServ-MPLS on network parameters
such as packet loss, delay and throughput for different
video resolutions.
[8] Proposes how the implementation of voice over
internet protocol (VoIP) system in UUM campus can help
users to freely communicate by using the VoIP
technique. According to him the proposed system also
helps to increase the effectiveness of using the Internet
bandwidth; since the users can communicate with each
other without the need to have an Internet access.
[9] Proposed the optimization techniques that can be
used to analyze and optimize the performance of wired
and wireless networks of a campus area. Cisco packet
tracer was used for the simulation
[10] Attempts to identify some of the network
performance parameters that service providers will
focus on to develop a VOIP over WIMAX communication
tool that will serve as a voice communication broadband
replacement technology to old circuit switch voice
communication.
[11] Provides the quick and technical overview of
concept, standard, technology and architecture for IEEE
802.16 WiMAX.
[12] Presented a Media Access Control Protocol that
provides the quality of service for VoIP over wlan. In
this, the characteristics of our proposed protocol are No
hardware modification of VOIP STA. Backward
compatibility in order to minimize the cost of
development no modification of access points.
[13] Provides focusing on quality of service scheduling
services and performance related metrics such as jitter,
packet end to end delay and MOS (mean opinion score).
[14] Evaluated the performance measures such as delay
variation, delay, page response time, throughput and
packet drop for different types of traffic such as voice,

video, data in their movement in a congested network
for both MPLS-TE and Conventional IP Network.
[15] Studies VOIP to a level that allows discussion of
security purposes and concerns. In this work, VOIP
components will include network components, gateway,
end user equipment, call processors and two of common
architectures.
[16] Work on the network performance analysis to
evaluate the effects of the application of different voice
encoder schemes on quality of service of VOIP system
which is deployed with the UMTS network.
2.3 Conceptual Framework
In this research work, a conceptual VoIP
model/framework was designed for Adamawa State
University, Mubi. The implementation was carried out
using Packet Tracer and the network is based on the
Hierarchical Network Design Model and mesh topology.
The infrastructure of the campus was considered during
the framework design.
The following attributes were also considered during the
implementation of the framework.
i. Placement of VoIP for internal use over the
current network.
ii. VoIP signaling, control, and management of calls
would be done by using the Cisco 2811 router’s
telephony service.
iii. Users can receive/make calls by using IP
Phones.
Fig -1: Conceptual framework of the system
3. METHODOLOGY
The mode of communication in Adamawa State
University campus does not involve any communication
system other than the current call services provided by
Internet Service Providers (ISP) which involves the use
of mobile Phones or smartphones with the induced
service cost. For any form of communication to occur
between offices or departments or units, mobile phones
or smartphones are the major alternative means to
communicate. Consequently, without service charges or
costs, it will be difficult to communicate. Network
Systems such as the VoIP system is being utilized by
numerous organizations to save the cost charges
induced by ISP for their call services. Therefore for
effective and efficient voice communication to take place
on campus between various offices, departments, and
units, etc. a telephony system needs to be implemented.
Deployment of this system will make communication a
lot easier since no cost charges will be attached to voice
communication, instead, the voice calls will be cost-free.
With such an advantage, Adamawa state university can
benefit greatly from this technology that has been
around for more than a decade. The prototype network
developed in this study can be used to implement this
technology in Adamawa state university. Apart from
that, the prototype will also address major VOIP voice
quality issue that usually affects VOIP systems.
3.1 System Development
The Cisco Packet Tracer Simulation of the VoIP
Telephony system is given below:
Step 1: Network devices were selected from the Cisco
packet tracer device database and connected.
Step 2: The VLAN’s and IP networks were created for
different user segments and network traffic.
Step 3: IP addresses were assigned to the seven
different endpoints on the campus via the DHCP
(Dynamic Host Configuration Protocol).
Step 4: Static and dynamic routing protocols were
configured on the routers with dual Internet access
connection.
Step 5: Telephony-service was then implemented by
configuring the cisco 2811 routers to support the Cisco
IP phones, through directory assignments and call
connections using VoIP dial-peers.
3.1.1 System Flowchart
The first telephony process represented in the flowchart
below involves the opening of a data connection,
followed by sending a telephone call request. When data

is returned, it means a call is completed and the
connection will be closed or else the call request will be
sent again until a connection is returned.
Fig -2: Flowchart for VoIP System
3.1.2 System’s Use Case Diagram
Fig-3: Use case diagram of the VoIP System
3.1.3 System’s State Diagram
Fig-4: State Diagram for the Simulation of the Cisco IP
phone calling process
4. RESULTS
In this research work, the results of the study are
presented and discussed concerning the aim of the
study, which is to determine the influence of using
quality of service (QoS) Configurations to improve Voice
communication efficiency for the VoIP network
simulation of Adamawa state university, Mubi.
Table-1: The IP address and VLAN’s
Table 1. Above gives the list of designated network
addresses and VLAN’s of faculty buildings and other
sections of the University. Each one of them has a
designated IP address for both Voice and data
communication in the system. For example, the faculty
of the science IP address is 192.168.10.0/23 for data
communication and 192.168.20.0/23 for voice
communication.

Fig-5: Connected routers of the network simulation with
their designation.
Fig-5. Above shows, the designated routers of each
faculty and various sections of the University connected
in a mesh topology.
Fig -6: The simulation of ICT and Science complex VoIP
clusters with the library VoIP network.
Fig-6. Above is a portion of the larger network, showing
the clustered form of ICT Centre and science complex
connected on a multilayer switch, together with the
library Unit.
Fig-7: view of Economics department VoIP network
Fig 7. Above shows a view of the Economics department
network and the various connected devices.
Fig-8: Calling from one IP phone to another
In Fig-8. Above the process of making a call between two
destinations was depicted using a Cisco IP phone, in
which the sender dials the destination phone number
while the receiver sees an incoming call.
5. CONCLUSION
In conclusion, a prototype campus network was
designed and implemented in this research work using
the Cisco Packet Tracer software. Our objective was to
design and simulate an efficient VoIP network scenario
for the case study Adamawa State University and also to
configure the virtual network devices of the simulation,
evaluating point-to-point connections to ensure proper
communication between various offices and
departments. To implement this topology, we had to
study the whole VoIP scenario, VoIP background, its
features, benefits, drawbacks and its future in the
networking world. Overall, this study improved our
understanding of the whole concept of VoIP and its ever-
increasing demand in present times.
5.1 Recommendations
i. The access list should be implemented in the network
to provide more security to control which packets or
routing updates are permitted or denied in or out of the
network.
ii. Network security infrastructure such as VPNs,
firewalls, etc. optimize voice and therefore should be
implemented because they are capable of supporting the
advance security requirement of VOIP.
iii. VoIP comes with new complex threats, therefore, it is
highly recommended that network security upgrades be
carried out.

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IRJET- Campus-Wide Internet Telephony Design and Simulation using Voice over Internet Protocol: A Case Study of Adamawa State University, Mubi, Nigeria

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IRJET- Campus-Wide Internet Telephony Design and Simulation using Voice over Internet Protocol: A Case Study of Adamawa State University, Mubi, Nigeria