Abstract

JIGJIGA UNIVERSITY
Institute of Technology
School Of Electrical And Computer Engineering
Department Of Electronics And Communication
Thesis title: Simulate And Analysis of Voip and Vlan over LAN
Group Members
Name Id-Number
1. Yared Acahlu---------------------------------0955/07
2. Cherinet Yohannis--------------------------0938/07
3. Abenezer Ayalew-----------------------------082/06
ADVISOR NAME: Mr. Dr. Anbarasu
In partial fulfillment of the requirements for the Degree of Bachelor of Science

In Electrical and Computer Engineering.
Jigjig, Ethiopia
June 2019
Simulate and analysis of VoIP and VLAN over LAN 2019
Acknowledgment
We would like to express our sincere gratitude to our advisor INSTRUCTOR Mr. Anbrasso.
Finally To our lovely family who have been a persistent source of encouragement not only
during the thesis work but also throughout our academic career. We want them to know that we
respect and always keep in our memory their boundless and invaluable support, beyond a simple
thank you and love you.
JIGJIGA UNIVERSITY I
ABSTRAACT
Technology is progressing at a faster rate each passing day. Voice over Internet Protocol (VoIP)
is one of those technological utilities that is very heavily used in the world today. A technology
called voice over Internet Protocol (VoIP), or Internet telephony means that voice is carried over
an IP network. This project entails the simulation and Analysis of voice over a Local Area
Network (LAN). This document will show simulation of VoIP and VLAN on a Local Area
Network (LAN). Before we are going to the new technology we try to explain the working
principle of old voice communication system and we compare it with the new technology which
is VOIP. Then we see the basics of VoIP such as background information about what VoIP is
and how it function, Common types of services such as softphone and IP phone, Analog phone
with ATA, VoIP Signaling and Transport Protocols and, as well as Quality of Service. Even if
we give more concern about VOIP, the document will also hold information about VLAN,
VLAN trunk, inter VLAN routing. To simulate and analysis we use different software these are
Packet Tracer, GN3, Wireshark, Putty, command promote, oracle virtual box, and cisco IP
communicator soft phone.
The traditional workplace is evolving; the way in which businesses communicate today is
different than it was in the past and yet is likely to change again in the future. Organizations are
seeking unified communications in hopes of finding innovative ways to reduce their bottom-line
communication costs. Today, many enterprise business infrastructures are comprised of separate
networks – voice, data, and mobile, yet most of the time these networks never interact. The
ability to link business application from various networks with communications proves to be
valuable and is known as convergence. (Puglia Vincent, 2010)
Even if VoIP is prohibited in Ethiopia. It give lots of advantages for Corporate organizations,
Universities, Health care, Airports, Hotels, Banks etc. Which have large LAN. This project is
economically cost effective, gives full control to the administrator and provides mobility,
feasible, Peer-to-Peer phone calls. We also believe that it will pave the way for the transition
from POTS to IP based telephone system and Converged network.
JIGJIGA UNIVERSITY II
List of Figures
Figure 1: Cisco PPDIOO Life Cycle ............................................................................................. 5

Figure 2: Evolution of Telephones ............................................................................................... 7
Figure 3: VOIP components ......................................................................................................... 11
Figure 4:VoIP Protocol Architecture ............................................................................................ 12
Figure 5: SIP Distribution Architecture ....................................................................................... 14
Figure 6: H.323 protocol header ................................................................................................. 18
Figure 7: over all VOIP header ..................................................................................................... 18
Figure 8: RTP header .................................................................................................................. 19
Figure 10:Hardware components of router ................................................................................... 29
Figure 11: Router interface ........................................................................................................... 30
Figure 12: Front panel of the Cisco 2960-24TT-L switch. ........................................................... 30
Figure 13: Back panel of the Cisco 2960-24TT-L switch. ........................................................... 31
Figure 14: Cisco IP Phone cable connections .............................................................................. 33
Figure 15: Cisco Softphone .......................................................................................................... 34
Figure 16: Cisco ATA 186 ............................................................................................................ 35
Figure 17:RJ-45 connectors. ......................................................................................................... 36
Figure 18: Our typical LAN network............................................................................................ 39
Figure 19: VLAN Tag Field ......................................................................................................... 43
Figure 20: GNS3 topology ............................................................................................................ 46
Figure 21: CIPC Phones................................................................................................................ 46
Figure 22: verify VLAN configuration. ........................................................................................ 47
Figure 23: Verify vlan Trucking ................................................................................................... 48
Figure 24: Verify DHCP ............................................................................................................... 48
Figure 25: Verify routing table ..................................................................................................... 49
Figure 26:Verify e-phone .............................................................................................................. 49
Figure 27: Test end to end connection between a) In the same vlan b) in different VLAN. ........ 50
Figure 28: Different kind of VOIP phones ................................................................................... 50
Figure 29:phone 0923 make a call to phone 0928 ........................................................................ 51
Figure 30: VOIP protocols ............................................................................................................ 52
Figure 31: Signaling protocol Sip ............................................................................................... 52
Figure 32: RTP header .................................................................................................................. 53
Figure 33: analog speech .............................................................................................................. 53
Figure 34: Global VOIP Market ................................................................................................... 54
JIGJIGA UNIVERSITY III

List of Table
Table 1: VoIP Advantages and Disadvantages ............................................................................ 10

Table 2: Different codec scheme ................................................................................................ 13
Table 3:Comparison of voip vs. PSTN ......................................................................................... 25
JIGJIGA UNIVERSITY IV
Acronyms
A
Advanced Integration Module ………………………………………………………………AIM
Advanced Research Projects Agency Network……………………………………... ARPANET
Analog Telephone Adapter………………………………………………………………….ATA
auxiliary……………………………………………………………………………………..AUX
C
Capital expenditure………………………………………………………………………CAPEX.
Central processing Unit………………………………………………………………………CPU
Cisco Internet Protocol Communicator……………………………………………………...CIPC
command line interface………………………………………………………………………..CLI
compression –decompression……………………………………………………………….Codec
content addressable memory…………………………………………………………………CAM
D
Dual Tone Multi-Frequency………………………………………………………………..DTMF
Dynamic host configuration protocol………………………………………………………DHCP
Differentiated Service…………………………………………………………………… DiffServ
E
Energy-efficient Ethernet……………………………………………………………………..EEE
Electronic Industries Alliance………………………………………….................................EIA
H
High-Speed WAN Interface Card…………………………………………………………..HWIC
Hyper Text Transfer Protocol……………………………………………………………..HTTP
I
Integrated Services Digital Network…………………………………………………………ISDN
Institute Electrical and Electronics Engineering………………………………………..……IEEE
International Telecommunication Union- Telecommunication Standardization Sector…….ITU-T
Internet engineering task force………………………………………………………………..IETF
Internet Protocol………………………………………………………………………………….IP
Internetwork Operating System………………………………………………………………..IOS
International Organization for Standardization………………………………………………..ISO
Internet of Things………………………………………………………………………………IoT
JIGJIGA UNIVERSITY V
L
Local Area Networks………………………………………………………………………...LAN
M
Metropolitan Area Networks………………………………………………………………..MANs
Multiprotocol Label Switching…………………………………………………………….MPLS
N
Non-volatile…………………………………………………………………………………RAM
O
Operation expenditure………………………………………………………………………OPEX
P
Plain Old Telephone Service………………………………………………………………...POTS
Prepare, Plan, Design, Implement, Operate, and Optimize………………………………PPDIOO
public switched telephone network………………………………………………………….PSTN
Q
Quality Of Service……………………………………………………………………………QOS
Quad Small Form-Factor Pluggable Plus……………………………………………………QSFP
R
Random access memory……………………………………………………………………...RAM
Read-only memory…………………………………………………………………………...ROM
Real-time Transport Protocol………………………………………………………………….RTP
Request for Comments………………………………………………………………………..RFC
S
Secure Shell……………………………………………………………………………………SSH
Session Initiation Protocol……………………………………………………………………...SIP
T
Transport control protocol………………………………………………………………….....TCP
Telecommunications Industry Association…………………………………………………....TIA
U
unshielded twisted pair cable………………………………………………………………….UTP
User Agent……………………………………………………………………………………...UA
JIGJIGA UNIVERSITY VI
User Agent Client……………………………………………………………………………..UAC
User Datagram Protocol……………………………………………………………………….UDP
V
Virtual Local Area Network………………………………………………………………...VLAN
Voice over Internet Protocol………………………………………………………………….VoIP
W
WideAreaNetworks…………………………………………………………………………WANs
WAN Interface Card…………………………………………………………………………WICs
JIGJIGA UNIVERSITY VII

Table of Contents
Acknowledgment ............................................................................................................................. I
ABSTRAACT ................................................................................................................................. II
List of Figures ............................................................................................................................... III
List of Table .................................................................................................................................. IV
Acronyms ....................................................................................................................................... V
CHAPTER ONE ............................................................................................................................. 1
Introduction ................................................................................................................................. 1
1.1 Background of Study ......................................................................................................... 2
1.2 Statement of the Problem .................................................................................................. 3
1.3 Study Objectives ................................................................................................................ 3
1.4 Scope and Limitation of Study .......................................................................................... 4
1.5 Research Methodology ...................................................................................................... 4
1.6 Organization of Study ........................................................................................................ 5
CHAPTER TWO ............................................................................................................................ 6
2. Literature Review .................................................................................................................... 6
Introduction ................................................................................................................................. 6
2.1 History of Telephony ......................................................................................................... 7
2.2 Data Networks ................................................................................................................... 8
2.3 Voice over Internet Protocol (VoIP) ................................................................................. 8
2.4 VoIP components ............................................................................................................ 10
2.5 VoIP compression Algorithms ........................................................................................ 12
2.6 VoIP Protocols ................................................................................................................. 13
2.7 H.323 ............................................................................................................................... 17
2.8 Transport Protocol (Real-Time Protocol (RTP)) ............................................................. 18
2.9 QOS: ................................................................................................................................ 20
2.10 Why Change to VoIP? ................................................................................................... 23
2.11 Comparison of voip vs. PSTN ....................................................................................... 24
Chapter Three................................................................................................................................ 26
3.1 System Design and Analysis ............................................................................................... 26
3.1.1 Network simulation tools.............................................................................................. 26
3.1.2 Network protocol analyzer ........................................................................................... 27
3.1.3 Virtual machine ............................................................................................................ 27
3.1.4Softphone ....................................................................................................................... 28
3.2 System components and operation ...................................................................................... 28
3.2.1 Router ........................................................................................................................... 28
3.2.2 Switches ........................................................................................................................ 30
3.2.3 Phones ........................................................................................................................... 32
3.2.4 Cables ........................................................................................................................... 35
3.2.5 Operating Systems ........................................................................................................ 36
3.2.6 IP Addressing ............................................................................................................... 38
3.3 Interconnecting and configuration Network elements ........................................................ 38
3.4 Working principle of VLAN ............................................................................................... 42
3.5 Working principle of VOIP ................................................................................................. 44
Chapter Four ................................................................................................................................. 47
4. Software Simulation Results and Discussions ...................................................................... 47
4.1 Result we get from Packet Tracer.................................................................................... 47
4.2 Result we get from GNS .................................................................................................. 52
Chapter Five .................................................................................................................................. 54
5. Conclusion and Recommendations for Future Work ............................................................ 54
5.1 Conclusion ....................................................................................................................... 54
5.2 Recommendations for Future Work ................................................................................ 55
Appendix ....................................................................................................................................... 56
Reference ...................................................................................................................................... 61
CHAPTER ONE
Introduction
Network performance is an important factor in the productivity of an organization. There are
technologies used to improve network performance one of them are vlan and voip.
Voice over Internet Protocol (VoIP) , also called IP telephony, is a methodology and group of
technologies for the delivery of voice communications microd multimedia sessions over Internet
Protocol (IP) networks, such as the Intranet, Extranet and Internet. The terms Internet telephony,
broadband telephony, and broadband phone service specifically refer to the provisioning of
communications services (voice, fax, SMS, voice-messaging) over the IP network, rather than
via the public switched telephone network (PSTN).
The steps and principals involved in originating VoIP telephone calls are similar to traditional
digital telephony and involve signaling, channel setup, digitization of the analog voice signals,
and encoding. Instead of being transmitted over a circuit-switched network, the digital
information is packetized, and transmission occurs as IP packets over a packet-switched network.
They transport media streams using special media delivery protocols that encode audio with
audio codecs.[1]
The other technology is a Virtual Local Area Network (VLAN). VLAN can be created on a
Layer 2 switch to reduce the size of broadcast domains, similar to a Layer 3 device. VLANs are
commonly incorporated into network design making it easier for a network to support the goals
of an organization. While VLANs are primarily used within switched local area networks,
modern implementations of VLANs allow them to span Metropolitan Area Networks (MANs)
and Wide Area Networks (WANs).
JIGJIGA UNIVERSITY 1
1.1 Background of Study
According to Hallock, in order for VoIP to have even been conceivable, the invention of two
very important technologies was mandatory: the telephone, and the internet. The first telephone
exchange occurred in 1878 in New Haven. In 1968 the internet was developed by Advanced
Research Projects Agency Network (ARPANET) , which was an agency that the U.S.
Department of Defense established in 1957 (2004). Greenberg states that when the telephone was
devised the voice was sent as analog packets along a land line (2013). Once these technologies
had been implemented, a small establishment titled Vocaltec, Inc. released their invention known
as Internet Phone which permitted users to call one another via computer workstations providing
a microphone and speakers were present (Hallock, 2004). By 1998 VoIP calls had barley reached
1% of all voice calls made in the world and that number reached about 25% in the year 2003.
VoIP had a major advance in 1988. Providers began constructing software which translated
digital data into analog data. This allowed VoIP calls to connect with calls made on a public
switched telephone network (PSTN). The capability of connecting to PSTNs as well as the
enhanced quality of broadband internet allowed VoIP to be operated to its full potential.
. This technology uses the Internet Protocol (IP) to transport voice signals over a data network.
Instead of using the conventional analogue voice signal (sine wave signal), human speech is
converted into a digital signal (1s and 0s) just like the data packets that travel through the
data network. Evolutionary? Yes. But IP telephony is more revolutionary than evolutionary
because it merges two very different yet critical worlds:
Voice telephony (highly cost sensitive; needs rock-solid reliability).
Data networks (accept occasional failures; subject to rapid change; need a lot of bandwidth).
A solution that merges these two worlds must simultaneously offer reliability, cost effectiveness,
a high data rate and the ability to evolve quickly. A solution that takes into account these
seemingly incompatible needs are readily needed. The pros and cons of IP telephony versus
“classic” telephony have been debated in many papers and will not be rehashed here. This
is helpful because transferring voice calls over data networks can save 75% or more compared to
traditional telephone service. (Frost & Sullivan, 2007).With a detailed network infrastructure in
place, it would not cost much to make calls through this existing data networks to reach
telephones internally using existing telephone systems and methods of calling. Calls to a host not
directly connected to the network can be made possible through the use of a gateways that
connects a voice call to a public telephone network and allows for direct communications to
future remote offices or external hosts due to expansion.
This solution will fundamentally transform the way in which from decreased carrier costs
to increased response times, thus the benefits of Unified Communications greatly outweigh the
investment. [2]
1.2 Statement of the Problem

Nowadays, there are so many LAN networks on Corporate organizations, Universities, Health
care, Airports, Hotels, Banks etc. but we observe that they only use it for simple data network
purpose even if they have high bandwidth. They also run two different network for two different
function, even if is there a possibility to converge both voice and data. Because of having two
different network they pay more money for Capital expenditure CAPEX and Operation
expenditure OPEX.
Therefore the aim of this project is to develop a communication technology, that help us to our
LAN effectively with it full capacity converge our data and telephony system
1.3 Study Objectives
1.3.1Global Objectives
The study has a general objective of developing a cost-effective IP interoperability
communication system that implements voice over a data network. It examines and integrates
different components that constitute an IP Telephony solution. A big part of the project is to also
understand the standards that are involved in a VOIP network.
1.3.2 Specific Objectives

The specific objectives of this project are to achieve the following:
 To Analysis VoIP system and Protocols.

 To simulate VLAN and VOIP and to show there result over LAN.
1.4 Scope and Limitation of Study
The research highlights converged networks with the basic LAN network that consist one router
and several switches. We focused on only on the technology of VOIP and VLAN. Under voip
technology we try to cover its protocol, router and switches configuration.
1.5 Research Methodology

The methodology used in this study is the cisco lifecycle approach to network design and
implementation. PPDIOO stands for Prepare, Plan, Design, Implement, Operate, and Optimize.
PPDIOO is a Cisco methodology that defines the continuous life cycle of services r equired for a
network. [3]. The network lifecycle approach provides several key benefits aside from keeping
the design process organized. These benefits includes: It lowers the total cost of ownership
by validating technology requirements and planning for infrastructure changes and resource
requirements. It increases network availability by producing a sound network design and
validating the network operation and improves business agility by establishing business
requirements and technology strategies. It speeds access to applications and services by
improving availability, reliability, security, scalability, and performance.
It is worth noting that as part of the design phase of the PPDIOO methodology, a top-down
approach is used, which begins with the organization’s requirements before looking at
technologies. Network designs are tested and simulated a pilot or prototype network before
moving into the implementation phase.
Figure 1: Cisco PPDIOO Life Cycle

It is worth noting that as part of the design phase of the PPDIOO methodology, a top-down
approach is used, which begins with the organization’s requirements before looking at
technologies. Network designs are tested and simulated a pilot or prototype network before
moving into the implementation phase.
1.6 Organization of Study

The project is divided into five chapters: The first chapter provides for the General
Introduction to the project; it includes the background of Voice over Internet protocol
technology, problem definition, objectives, scope and contribution of the project. The second
chapter presents the overview of Internet Telephony. Also, there are literature reviews about
existing systems and as well its component. The third chapter provides detailed description of
the proposed system. It explains the requirements and specifications as well as the design of
the new system, it also contain the detailed diagrams, Network architecture of the proposed
system and so on. The fourth chapter details the implementation, coding, documentation and the
testing of the proposed system. The fifth chapter presents the conclusion and recommendation.
CHAPTER TWO
2. Literature Review
Introduction
Telephony is the communication of spoken information between two or more participants, by
means of signals carried over electric wires or radio waves. Ever since Alexander Graham Bell
invented the telephone circuit and first envisioned the public telephone system, consumers and
businesses have relied on telephony as a staple of human interaction. With the advent of Internet
technologies and high-speed data connectivity in the enterprise, a new family of telephony
technologies began taking hold. Voice over IP, or VoIP, has significant appeal for the enterprise,
for service providers, and for end users, because it allows the Internet and commonplace data
networks, like those at offices, factories, and campuses, to become carriers for voice calls, video
conferencing, and other real-time media applications. VoIP-savvy organizations are discovering
that they can apply the paradigm of distributed, software-based networking to
voice applications and enable a new generation of telecommunications features, cost savings, and
productivity enhancements. VoIP can replace business telephone systems, or it can add value to
existing traditional telephony devices. For instance, long-distance connectivity between two
offices with traditional telephone systems can often be accomplished with a lower cost per call
when VoIP is employed.
Internet Protocol Telephony (IP Telephony) is the term commonly used to define the
transmission of phone calls over any data network that uses IP, like Internet, Intranets and
wired or wireless Local Area Networks (LAN). This is regardless of whether traditional
telephony equipment, computers and/or dedicated terminals take part in the calls and even if the
phone calls are totally or partially transmitted over the Internet. IP Telephony is, without doubt,
one of the technological developments that are being rapidly adopted by companies
nowadays.
One of the main reasons of this quick migration to Internet Telephony is that it makes the
integration of all means of communication, communication devices and media much easier. This
way users can be in touch with anyone, wherever they are, and in real time. In short, IP
Telephony allows for Unified communications to become part of the business environment,
helping companies save money and boost employee performance.
Internet protocol IP Telephony’s history is in its very early stages. It all started only a few years
ago, in 1995, when Vocal Tec launched their first Internet telephone. Before that, IP Telephony
was a field that attracted the interest of researchers; but since voice communication over the
Internet has been proved to be not only possible but also commercially viable, many are the
companies that have entered the VoIP (voice over internet protocol) Telephony market trying to
take the lead.
2.1 History of Telephony

The first voice transmission, sent by Alexander Graham Bell, was accomplished on 10th
March,1876. It gradually evolved from a one-way voice transmission to a bi-directional voice
transmission. Moving the voices across the wire requires a carbon microphone, a battery, an
electromagnet, and an iron diaphragm and a physical cable between each location when a user
wants to place a call.
Figure 2: Evolution of Telephones

Since the invention of Graham, the structure of Public line telephone network has changed
significantly. In the following paragraphs we focus on telephony network and the PBX that
traditional telephone systems run on. The acronym PSTN stands for Public Switched Telephone
Network. PSTN is the network that traditional phone systems used and was generally controlled
by the telecommunication companies. This is the network our calls are travelling over when we
pick up our handset and dial a number. This network spans the world and there are many
different interfaces to it;
POTS stand for Plain Old Telephone Service POTS. It is commonly used for residential use.
POTS is an analogue system and is controlled by electrical loops.
Integrated Services Digital Network: ISDN This is a faster and more feature-filled connection
(also more expensive). This gained some popularity within small to medium-sized businesses as
a cost-effective way of connecting to the PSTN and getting some advanced services, like many
lines to one office or voice and data lines on one service. ISDN is a digital service and offers a
few more features over POTS (BarrieDempster, 2006).
T1/E1 is a digital service used for high-volume data and voice networks and offers yet more
features than ISDN, the most important feature being increased bandwidth that translates, in
telephony, to more telephone lines (Kerry Garrison, 2006).
2.2 Data Networks

Data Networks can be classified according to coverage area or by the protocol used to transfer
data via them. By the first classification, data networks can be Wide Area Networks (WANs)that
cover a large area and distant computers or Metropolitan Area Networks (MANs) building for
example. implement vlan and voice technology on LAN.
In LAN data network, the predominant technology in the world is Ethernet. Ethernet operates in
the data link layer and the physical layer. The Ethernet protocol standards define many aspects of
network communication including frame format, frame size, timing, and encoding.
It is a family of networking technologies that are defined in the International Electrical and
Electronics Engineering IEEE 802.2 and 802.3 standards. Ethernet supports data bandwidths of:
10 Mb/s, 100 Mb/s, 1000 Mb/s (1 Gb/s), 10,000 Mb/s (10 Gb/s) , 40,000 Mb/s (40 Gb/s),and
100,000 Mb/s (100 Gb/s). We use 100Mb/s.
2.3 Voice over Internet Protocol (VoIP)

Voice over Internet Protocol (VoIP) is one of the most important technologies in the world of
communication. VoIP is simply a way to make phone calls through the internet. In other words,
VoIP transmits packet via packet-switched based network in which voice packets may take the
most efficient path. On the other hand, the traditional public switched telephone network (PSTN)
is a circuit-switched based network which requires a dedicated line for telecommunications
activity (J.B. Meisel, M. Needles, 2005).
Furthermore, Internet was initially considered to transmit data traffic and it is performing this
task really well. However, Internet is best effort network and therefore it is not sufficient enough
for the transmission of real-time traffic such as VoIP. In addition, there are about 1 billion fixed
telephone lines and 2 billion cell phones in the world that use PSTN systems. In the near future,
they will move to networks that are based on open protocols known as VoIP (V. Mockapetris,
2006).
That can be seen from the increasing number of VoIP users, for instance there are more than
eighty million subscribers of Skype; a very popular VoIP commercial application (K. Dileep, A.
Saleem and R. Yeonseung, 2008). VoIP has gained popularity due to the more advantages it
offers than PSTN systems especially that voice is transmitted in digital form which enables VoIP
to provide more features. However, VoIP still suffer few drawbacks which user should consider
when deploying VoIP system.
Advantages Disadvantages
 Low cost  Users cannot make calls during

 Flexibility. poweroutages.
 Provides voice mail and call  Connection limitation to
forwarding. emergencyservices.
 Easy to implement and install  Depends on Internet connection
 Free services gained usually condition.IP network that does not
whenconnecting from PC to PC (G. guarantee Quality of Service for voice
Samrat, B. Sudeept, 2008) communication (J.M. Lozano-
 Network Capacity utilization Gendreau, A.Z. Halabi, M. Choueiri
 Users can make VoIP calls from and V. Besong, 2006).
anywherefor long distance or
international calls.
 Integration with other available
servicesover the Internet
Table 1: VoIP Advantages and Disadvantages
Basically, VoIP system can be configured in these connection modes respectively; PC to PC,
Telephony to Telephony and PC to Telephony (H. Yong-feng, Z. Jiang-ling, 2000). Moreover,
telephony can be digital type or analogue type. In case of analogue phone, it would be connected
to the system via adapters which convert the analogue signals into digital format.
2.4 VoIP components

VoIP consists of three essential components: Coder/Decoder (CODEC) , Packetizer and play out
buffer (C. Lin, X. Yang, S. Xuemin and W.M. Jon, 2006). At the sender side, an adequate
sample of analogue voice signals are converted to digital signals, compressed and then encoded
into a predetermined format using voice codec. There are various voice codecs developed and
standardized by International Telecommunication Union-Telecommunication (ITU-T) such as
G.711, G.729, G.723.1a, etc.
Next, packetization process is performed which fragment encoded voice into equal size of
packets. Furthermore, in each packet, some protocol headers from different layers are attached to
the encoded voice. Protocols headers added to voice packets are of Real-time Transport Protocol
(RTP), User Datagram Protocol (UDP), and Internet Protocol (IP) as well as data link layer
header. In addition, RTP and Real-Time Control Protocol (RTCP) were designed at the
application layer to support real-time applications. Although Transport control protocol TCP
transport protocol is commonly used in the internet, UDP protocol is preferred in VoIP and other
delay-sensitive real-time applications. TCP protocol is suitable for less delay sensitive data
packets and not for delay-sensitive packets due to the acknowledgement (ACK) scheme that TCP
applies. This scheme introduces delay as receiver has to notify the sender for each received
packet by sending an ACK. On the other hand, UDP does not apply this scheme and thus, it is
more suitable for VoIP applications. The packets are then sent out over IP network to its
destination where the reverse process of decoding and DE packetizing of the received packets is
carried out. During the transmission process, time variations of packets delivery (jitter) may
occur. Hence, a playout buffer is used at the receiver end to smoothen the playout by mitigating
the incurred jitter. Packets are queued at the playout buffer for a playout time before being
played. However, packets arriving later than the playout time are discarded. The principle
components of a VoIP system, which covers the end-to-end transmission of voice, are illustrated
in Figure below.
Figure 3: VOIP components

Besides, there are signaling protocols of VoIP namely Session Initiation Protocol (SIP) and
H.323. These signaling protocols are required at the very beginning to establish VoIP calls and at
the end to close the media streams between the clients (R. P. Swale, 2001). H.323 was
standardized by International Telecommunication Union- Telecommunication Standardization
Sector ITU-T specifically to smoothly work together with PSTN while SIP was
standardized by Internet engineering task force (IETF) to support internet applications such as
telephony.
In figure 2.3, VoIP protocol stack is illustrated. Furthermore, in IP networks, IP addresses can be
changed from one session to another, especially in dial-up case. Therefore, there is a need for a
common meeting point shared among users to enable them finding each other at the
establishment stage of communication. This common meeting point is generically known as a
call server.
Figure 4:VoIP Protocol Architecture
2.5 VoIP compression Algorithms

compression –decompression (Codec) is used to convert analog signal (video, voice) to digital
data or convert digital data stream to analog signal. This helps to provide compression capability
to save network bandwidth. There are different audio codec available for voice application
(Hersent, 2011). They include:
G.711: approved in 1965 and is the simplest way of digitizing analog signal. The algorithm uses
Pulse Code Modulation with less than 1% acceptable packet loss factor. The encoded audio
stream of G.711 is 64kbit/s, so is worst in terms of bandwidth but the best in quality among the
entire scheme.
G.722 was approved in 1988 and provides higher quality digital coding at 7 KHz of audio
spectrum at only 48, 56, or 64bits. It is mainly used for all professional conversation voice
application such as video conference and IP phone applications
G.722.1 is a wideband coder designed by Picturetel, which operate at 24bit/s or 32bit/s. It
encodes a frames of 20ms with workload of 20ms, 16kbit/s version of G.722.1 supports windows
messenger.
G.723.1 was approved in 1995 for use in H.323 communication and UMTS 99 video cell
phones. It uses frame length of 30ms and needs a workload of 7.5ms in 64kbt/s or 5.3kbit/s
operation modes. The algorithm is not designed for music and its difficult to be used in a fax and
modem signal transmission. The International Telecommunication Union and Telephony
recommends it for use in narrow band video conferencing and 3G wireless multimedia devices.
G.726 was approved in 1990 and uses Adaptive differential Code modulation (ADPCM)
techniques to encode G.711 bit stream in words of 2, 3, or 4 bits resulting in bit rate of 16, 24, 32
or 64kbits/s .
G.728 uses low delay, codec executed linear prediction (LD CELP) coding techniques with a
mean opinion score (MOS) similar to G.726. The algorithm is used for Fax and modem
transmission, and also for H.323 video conference.
G.729 is conjugate-structure, Algebraic Code Excited Linear Prediction (CS-ACELP) speech
compression algorithm approved by ITU-T for use in voice over frame relay application. It
produce 80-bits frame encoding 10ms of speech at a bit rate of 8kbit/s. The scheme is not
designed for music and does not support Dual-Tone Multi-Frequency (DTMF) signalling tones
reliably. Listed in table 3 are properties of common voice codec schemes.
Codec Bit Rate Payload Packets per Quality bandwidth Sample algorithm
Seconds (pps) period
G.711 64kbit/s 160bytes 50pps Excellent 95.2kbps 20ms PCM
G.729 8kbit/s 20bytes 50pps Good 39.2kbps 10ms CS-ACELP
G.723.1 6.3kbit/s 24bytes 34pps Good 27.2kbps 30ms MPC-MLQ
G.723.1 5.3kbit/s 20bytes 34pps Good 26.1kbps 30ms ACEP
Table 2: Different codec scheme
2.6 VoIP Protocols

There are two types of VoIP protocols: signaling and transport. The signaling protocols handle
the functions derived from the telephone system architecture, and the transport protocols carry
the voice packets generated from the codec. Phones use the signaling protocol to register with the
call server, set up, and tear down calls. Signaling protocols are also used for features such as
directory services and screen displays. Once a call has been established, the voice data packets
are typically sent directly between the phones using RTP encapsulation.
2.6.1 Signaling Protocols

Even though the VoIP architecture is completely different from that used by traditional
telephony, we still have the basic requirement of signaling. Somehow, phones have to ring,
numbers must be communicated, and routes have to be set up, and these functions are handled by
the signaling protocol. The three most common types are H.323, Skinny, and the Session
Initiation Protocol, or SIP.
2.6.1.1 Session Initiation Protocol

SIP is an Internet Engineering Task Force (IETF) defined signalling protocol, widely used for
controlling multimedia communication sessions such as voice and video calls over Internet
Protocol. The protocol can be used for creating, modif ying and terminating of multimedia
communication sessions between end users. Session is considered to be a communication states
kept between senders and receivers during communication.
Figure 5: SIP Distribution Architecture
Examples of communication sessions are Internet telephone calls, distribution of multimedia etc.
The modification can involve changing addresses or ports, inviting more participants, and
adding or deleting media streams. SIP clients typically use TCP (Transmission Control Protocol)
or UDP on por t numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints.
Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically
used for encrypted traffic. In the next paragraphs, the structural, functional and characteristics
features of SIP are explained in detail.
SIP distributed architecture
As shown in Figure 2.4 the SIP protocol defines a collection of entities that take part on a SIP
communication, which are, User Agents, Proxy Server, Location Server, Registrar and Redirect
Server. All these elements work together on one computer to perform specific task. Installation
of these elements on the same machine increases the speed and processing between the
network elements.
Components of SIP
User Agent Client (UAC): It is an entity that makes a call or request to call. User Agent Server
(UAS): It's a server at application level, which contacts the user when a SIP request is received
and responses on the user's name. The response to the request is accepted, rejected or
redirected. User Agent (UA): It's an application, which contains both the UAC and UAS. When
users want to talk with another, it executes a program that contains a UA. They can reside on the
user computer in the form of an application, but they can be cellular phones, PSTN
gateways, PDA’S and IVR (Interactive Voice Response) systems and so on. All the interactions
between users and the SIP protocol are done through UA. When UAC sends request to UAS,
UAS respond to that request and the session is established between them.
Registrar Server
Registrar server is a logical SIP entity that accepts the registration requests from senders extract
registration information about current location (IP address, port and username) and store that
information into location database. At the completion of registration process, Registrar Server
sends the ACK 200 message to the requestor. Registrar is very important entity that helps in
storing current information in location database, which further use for forking by proxy or
7redirect server (Jan Janak, 2003).
Redirect Server
A Redirect server is a user agent server that accepts and receive SIP request. Redirect server
checks the request from the location database and creates the list of current location of the user
and send back the request to the originator within a 3xx response (detail in next section). The
user receives the list of current destinations and send request directly to required destination. A
general SIP transaction model consists of sequence of SIP messages (request and responses)
between SIP network elements; which describe the SIP calls setup and teardown process.
Sequence of requests and their responses are used in number of steps to complete the call
process (Jan Janak, 2003).
Location Server
A location server is a SIP entity used by a proxy and redirect server to obtain the information
about the called party possible location. Location server stores the current location of the users
by registration process.
Proxy server
The proxy servers accept session requests generated by UA and request the address information
about the destination user to the registrar server. Then, it redirects the invitation directly to the
destination user if it's located on the same domain, or redirects it to another proxy of the
corresponding domain (Jan Janak, 2003).
SIP Addresses
The SIP network has the address attribute: SIP URL (SIP Uniform Resource Locator) to be
easily recognizable. SIP URLs used in SIP networ ks follows the structure of an email address; a
user @ host where user can be any user name, phone number, or the name of the agency. The
host can be either a domain name or an IP address. SIP address with the form phone number @
gateway shows the phone number on the network the General Switched Telephone Network
(GSTN) which can be contacted with a known gateway name.
SIP messages
SIP is a text-based protocol with syntax similar to that of Hyper Text Transfer Protocol
(HTTP). There are two different types of SIP messages: requests and responses. The first line of
a request has a method, defining the nature of the request, and a Request-URI (Uniform
Resource Indictor), indicating where the request should be sent. The first line of a response has a
response code [10]. Method is an important entity in the request line and used to decide the
function of request, six types of methods are defined: REGISTER, INVITE, ACK, CANCEL,
BYE, OPTIONS.
SIP Responses
Every request needs a response, when a user agent receives a request it replies the response.
Response methods are similar to request, except to the first line. First line of the response
contains protocol version (SIP/2.0), reply code, and reason phrase. The reply code is integer
number from 100 to 699 and indicates type of response.
2.7 H.323
H.323 is a standard approved by International Telecommunication Union (ITU) in 1996 to
promote compatibility in videoconference transmissions over IP networks. It consists of groups
of protocols that are used for call set-up, call termination, registration, authentication and other
functions. These protocols are transported over TCP or UDP protocols stack. The H.323 family
of protocol consists of H.225 for registration, admission, and call signaling. H.245 is used to
establish and control the media sessions. T.120 is used for conferencing applications in which
desktop and white-board application can be shared. The audio codec is defined by G.711, G.722,
G.729 and G.723.1, while video codec is defined by H.261 and H.263 series of specifications.
H.323 deploys RTP for media transport and RTCP is used for purpose of controlling RTP
sessions.
Figure 6: H.323 protocol header
2.8 Transport Protocol (Real-Time Protocol (RTP))

RTP provides end-to-end network transport functions suitable for applications transmitting real-
time data, such as audio, video or simulation data, over multicast or unicast network services.
RTP does not address resource reservation and does not guarantee quality-of-service for real-
time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring
of the data delivery in a manner scalable to large multicast networks, and to provide minimal
control and identification functionality. RTP and RTCP are designed to be independent of the
underlying transport and network layers. The protocol supports the use of RTP-level translators
and mixers.
Figure 7: over all VOIP header
Figure 8: RTP header
Ver, Version. 2 bits: RTP version number. Always set to 2.
P, Padding. 1 bit: If set, this packet contains one or more additional padding bytes at the end
which are not part of the payload. The last byte of the padding contains a count of how many
padding bytes should be ignored. Padding may be needed by some encryption algorithms with
fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit.
X, Extension. 1 bit: If set, the fixed header is followed by exactly one header extension.
CC, CSRC count. 4 bits: The number of CSRC identifiers that follow the fixed header.
M, Marker. 1 bit: The interpretation of the marker is defined by a profile. It is intended to allow
significant events such as frame boundaries to be marked in the packet stream. A profile may
define additional marker bits or specify that there is no marker bit by changing the number of
bits in the payload type field.
PT, Payload Type. 7 bits: Identifies the format of the RTP payload and determines its
interpretation by the application. A profile specifies a default static mapping of payload type
codes to payload formats. Additional payload type codes may be defined dynamically through
non-RTP means. An RTP sender emits a single RTP payload type at any given time; this field is
not intended for multiplexing separate media streams.
Sequence Number. 16 bits: The sequence number increments by one for each RTP data packet
sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The
initial value of the sequence number is random (unpredictable) to make known-plaintext attacks
on encryption more difficult, even if the source itself does not encrypt, because the packets may
flow through a translator that does.
Timestamp. 32 bits: The timestamp reflects the sampling instant of the first octet in the RTP
data packet. The sampling instant must be derived from a clock that increments monotonically
and linearly in time to allow synchronization and jitter calculations. The resolution of the clock
must be sufficient for the desired synchronization accuracy and for measuring packet arrival
jitter (one tick per video frame is typically not sufficient). The clock frequency is dependent on
the format of data carried as payload and is specified statically in the profile or payload format
specification that defines the format, or may be specified dynamically for payload formats
defined through non-RTP means. If RTP packets are generated periodically, the nominal
sampling instant as determined from the sampling clock is to be used, not a reading of the system
clock. As an example, for fixed-rate audio the timestamp clock would likely increment by one
for each sampling period. If an audio application reads blocks covering 160 sampling periods
from the input device, the timestamp would be increased by 160 for each such block, regardless
of whether the block is transmitted in a packet or dropped as silent.
SSRC, Synchronization source. 32: bitsIdentifies the synchronization source. The value is
chosen randomly, with the intent that no two synchronization sources within the same RTP
session will have the same SSRC. Although the probability of multiple sources choosing the
same identifier is low, all RTP implementations must be prepared to detect and resolve
collisions. If a source changes its source transport address, it must also choose a new SSRC to
avoid being interpreted as a looped source.
CSRC, Contributing source. 32 bits: An array of 0 to 15 CSRC elements identifying the

contributing sources for the payload contained in this packet. The number of identifiers is given
by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC
identifiers are inserted by mixers, using the SSRC identifiers of contributing sources. For
example, for audio packets the SSRC identifiers of all sources that were mixed together to create
a packet are listed, allowing correct talker indication at the receiver.
2.9 Quality Of Service QOS:

Voice Traffic
Voice traffic refers to the amount of calls made on a VoIP network infrastructure. The
key QoS requirements and recommendations to ensure respectable quality VoIP service include
DSCP EF voice traffic marking, a Loss percentage which is less than one percent, One way
latency should not exceed 150 milliseconds, the average one-way jitter should be targeted at less
than 30 milliseconds, and a guaranteed priority bandwidth range of 21 to 320 kilobits per second
is required per call. Loss, Latency, and Jitter are factors that directly affect the quality of voice
calls
Loss
During calls - skips and voice clippings are both caused by Loss. Packet Loss
Concealment (PLC) is used to mask the effects of VoIP packets that encounter Loss. Every PLC
method is different and the one used depends on the type of codec. Waveform codecs use a
simple method that replays the last sample with increasing attenuation at every repeat. This
technique can be effective at some instances of Loss. The loss of two or more packets yields a
noticeable difference in voice quality, which is why it is recommended that a loss rate of less
than one percent is present
Latency
In VoIP, Latency is described as the amount of time it takes for the sound to travel from
the individual who articulates it, to the individual who is receiving it. Three types of latency exist
in VoIP: Propagation delay, Handling delay, and Serialization delay. Propagation delay is caused
by the distance a signal must travel as light through a fiber optic cable or electricity in copper
based wire; Handling delay refers to the amount of time it takes for compression and
decompression of packets (packetization) as well as packet switching to occur and is also known
as processing delay; Serialization delay is the amount of time it takes to place data (a bit or byte)
onto an interface. Serialization delay influences overall latency at a fairly minimal level
Jitter
is the variation of packet interval time. Jitter is an issue which is only prevalent in packet-
based networks. The difference of when the packet is expected to arrive versus when it actually
is received is called Jitter (2006). A good way to prevent Jitter is by utilizing a Jitter Buffer. A
Jitter Buffer is also known as a Playout Buffer and is used to change asynchronous packets into a
synchronous stream. The buffer does this by changing variable delays in a network into constant
delays at the end systems
Services and Protocols
Quality of Service is dignified by how well a service within the network matches the
performance that is expected. Many networking specialists have developed techniques to
overcome QoS challenges in VoIP as well as Video-over IP. The three techniques are known as
RSVP (Resource Reservation Protocol), DiffServ (Differentiated Service), and MPLS
(Multiprotocol Label Switching). Each one of these three techniques has been standardized by
the IETF, formally known as the Internet Engineering Task Force
RSVP is specified in RFC 2205, this protocol resides in the Transport Layer (Layer 4)
and its purpose is to reserve specific amounts of network resources for a particular transmission
prior to its occurrence. The transmission pathway is established from the sending node issuing a
PATH statement through the RSVP to the receiving node on the network infrastructure. The
PATH message signposts the level of service it anticipates, as well as the amount of bandwidth
that needs to be allotted for the transmission to occur. Guaranteed service and controlled-load
service are two service types that RSVP permits. Guaranteed service ensures the transmission
will be free of packet loss and will only experience minimal delay, and controlled-load service
provides the type of QoS a transmission would experience if the network carried a diminutive
amount of traffic. Once the transmission destination node has received the Path message, it
answer back with a RESV (Reservation Request) message. The RESV message traces the path
taken by the PATH message in reverse. If the routers do not have sufficient bandwidth to
allocate for the transition, they reject the RESV. RSVP is a very useful tool on small networks,
but less popular in larger infrastructures. Bigger networks prefer more streamlined Quality of
Service techniques.
A great example of a streamlined technique used on larger networks is DiffServ. DiffServ

is used to prioritize traffic contained within the network. It modifies the IP datagrams that
contain payload data. When prioritizing traffic, DiffServ inserts its information within the
DiffServ field in an IPv4 Datagram. In IPv6 Datagrams Differentiated Service information is
placed within the Traffic Class field. DiffServ utilizes two diverse forwarding types: Expedited
Forwarding (EF) or Assured Forwarding (AF). When DiffServ is utilizing EF, the stream is
designated a minimum departure rate. When AF is utilized, one can assign different levels of
router resources to the data stream. Although AF prioritizes data, it does not guarantee that
packets will arrive on time or in sequence. DiffServ’s simplicity as well as its low overhead
makes it much better suited for large networking infrastructures. The final QoS technique which
belongs in the Network Layer (Layer 3) is known as the Multiprotocol Label Switching (MPLS)
technique. MPLS is used to indicate where data is to be forwarded. MPLS does this by replacing
the IP datagram header with a label at the first router a data stream encounters. The label
contains information about the location the packet is to be forwarded to. MPLS [5]
2.10 Why Change to VoIP?

With all of this disruption, why would we switch to Voice over IP? Probably the biggest reason
for adopting a VoIP-based architecture is money. Instead of paying for a series of telephone lines
or circuits, customers need only pay for a data connection. This is because the VoIP traffic
travels in IP packets that can share the data connection. In addition, IP packets can flow to any
destination connected to the Internet, and toll charges are much reduced. There are several
business cases in which forklift (removing everything in favor of the new equipment) changes to
telephony infrastructure are justified based on the savings in toll charges alone. VoIP
architectures can pay for themselves in a relatively short period of time, giving the company a
good Return on Investment, or ROI. There are several other, less obvious, opportunities to save
money with an IP-based VoIP solution. Networks deploying VoIP are often called converged
networks because they share the data network. Once the data network is installed, all other
devices are connected to it. This actually extends to other systems such as heating and cooling
systems, security, and video cameras. The impact of this change is hard to overestimate:
 Single network to support
 Single set of devices
 Single set of maintenance requirements
 Single set of employee skills
 Many ―off the shelf‖ components
 Single cable infrastructure
 Easier moves/adds/changes
All of these lead to a lower total cost of ownership, or TCO, for the network. This is not to say
that switching to VoIP eliminates specialized or expensive components. Indeed, some of the
pricing structures or licensing fees for VoIP phones or PBXs are very similar to their traditional
counterparts. VoIP desktop phones do not come cheap, with the more advanced models running
hundreds of dollars. However, one advantage is the ability to deploy softphones instead of
physical units. Softphones (phone software running on a laptop or handheld device) can be much
less expensive and easier to manage.
The single set of employee skills is worth another look. VoIP systems run on the data network
but are telephony systems that have been converted to IPbased protocols. The ideas and
functions are the same. Companies consolidating infrastructure sometimes find themselves with
a collection of employees that no longer possess the skills for the current infrastructure. As
mentioned earlier, they may lack a background in the protocols and hardware associated
with a data network. However, these employees are also the ones that understand the telephony
side of things. On the other hand, data network administrators may have little or no knowledge of
telephony. So a conversion to VoIP may require different types of training: vendor specific, basic
network, and VoIP specific. Leveraging both groups of employees may provide the best possible
outcome for the deployment.
2.11 Comparison of voip vs. PSTN
Table 3: Comparison of voip vs. PSTN
Chapter Three
3.1 System Design and Analysis

In chapter one and two we try to explore different VOIP and VLAN theoretical terms. In this
chapter we try to elaborate software component we used, network component we used in the
simulation and configuration we do on the network elements.
We use different Network simulation tools, Network protocol analyzer, Virtual machine,
Softphones.
3.1.1 Network simulation tools

3.1.1.1 Packet Tracer (Version 7.2): Packet Tracer is a medium fidelity, network-capable,
simulation-based learning environment for networking novices to design, configure, and
troubleshoot computer networks at a CCNA-level of complexity. Packet Tracer is an integrated
simulation, visualization, collaboration, and assessment environment. Packet Tracer supports
student and instructor creation of simulations, visualizations, and animations of networking
phenomena. Like any simulation, Packet Tracer relies on a simplified model of networking
devices and protocols. Real computer networks, experienced both in-person/hands-on and
remotely, remain the benchmark for understanding network behavior and developing networking
skills. Packet trace.
In this simulation tool we interconnect and configure our LAN. We use router and switches and
Straight through cable and Crossover unshielded twisted pair cable (UTP), computers, Laptop, IP
Phones and analog phone form Packet tracer.
3.1.1.2 GNS3 (version 0.8.7): GNS3 is a software emulator for networks that allows the
combination of virtual and real devices to simulate complex networks. It is based on a
combination of Dynamips emulation software to host Cisco IOS images and Virtual PC
Simulator to simulate network hosts. It also integrates with QEMU and VirtualBox virtual
machines.
We use GNS3, to simulate real softphone connection and to facilitate the analysis protocols that
used in VoIP communication with some details.
In this Software I used Cisco IOS Software, 3700 Software (C3745-ADVENTERPRISEK9-M),
Version 12.4(9)T4, and putty emulation software
3.1.2 Network protocol analyzer

3.1.2.1 WIRESHARK (Version 2.6.6): Wireshark is a data capturing program that
"understands" the structure (encapsulation) of different networking protocols. It can parse and
display the fields, along with their meanings as specified by different networking protocols.
Wireshark uses pcap to capture packets, so it can only capture packets on the types of networks
that pcap supports.
 Data can be captured "from the wire" from a live network connection or read from a file
of already-captured packets.
 Live data can be read from different types of networks, including Ethernet, IEEE 802.11,
PPP, and loopback.
 Captured network data can be browsed via a GUI, or via the terminal (command line)
version of the utility, TShark.
 Captured files can be programmatically edited or converted via command-line switches
to the "editcap" program.
 Data display can be refined using a display filter.
 Plug-ins can be created for dissecting new protocols.[20]
 VoIP calls in the captured traffic can be detected. If encoded in a compatible encoding,
the media flow can even be played.
 Raw USB traffic can be captured.[21]
 Wireless connections can also be filtered as long as they traverse the monitored Ethernet.
 Various settings, timers, and filters can be set to provide the facility of filtering the output
of the captured traffic. This simulation tool help us to analysis VOIP network protocols.
3.1.3 Virtual machine

3.1.3.1 Oracle VirtualBox (Version 6.0.6 Edition): Oracle VM VirtualBox (formerly
Sun VirtualBox, Sun xVM VirtualBox and Innotek VirtualBox) is a free and open-source hosted
hypervisor for x86 virtualization, developed by Oracle Corporation. Created by Innotek GmbH,
it was acquired by Sun Microsystems in 2008, which was, in turn, acquired by Oracle in 2010.
VirtualBox may be installed on Windows, machos, Linux, Solaris and Open Solaris. There are
also ports to FreeBSDand GenodeIt supports the creation and management of guest virtual
machines running Windows, Linux, BSD, OS/2, Solaris, Haiku, and OSx86, as well as limited
virtualization of macOS guests on Apple hardware. For some guest operating systems, a "Guest
Additions" package of device drivers and system applications is available, which typically
improves performance, especially that of graphics.
This platform helps us to create a virtual machine. That act as other computer on the same
network with its on network interface card and Softphone.
3.1.4Softphone
3.1.4.1 Cisco IP Communicator (Version 7.0.6.0): Cisco IP Communicator is a
Windows PC-based softphone application that lets you use your personal computer to make
premium voice and video calls. Offering the latest in IP communications technology, it is easy to
acquire, deploy, and use.
3.2 System components and operation

To build a computer network (Data Network) with capability of carrying voice. We use Router,
Switches, phones (it can be IP-phone, Softphone and Analog phone with Analog Telephone
Adapter ATA), Computers and Cat5 UTP cable.
In packet tracer we try to demonstrate VOIP as well as VLAN. In packet trace rwe use the
following components:
3.2.1 Router
Router is a key component; in this project we use one router. Routers not only used to connect
different networks, but also as Dynamic host configuration protocol DHCP and PBX server.
Router is a computer. It has hardware part and software part. Router hardware part are Central
processing Unit CPU and memory to temporarily and permanently store data to execute
operating system instructions, such as system initialization, routing functions, and switching
functions.
The router we used in our project called as Cisco 2811 router. it is one of 2800 cisco series
routers. Router components and their functions: CPU - Executes operating system instructions,
Random access memory (RAM): Contains the running copy of configuration file and Stores
routing table. RAM contents lost when power is off, Read-only memory (ROM) - Holds
diagnostic software used when router is powered up. Stores the router’s bootstrap program, Non-
volatile RAM (NVRAM) - Stores startup configuration. This may include IP addresses (Routing
protocol, Hostname of router), Flash memory - Contains the operating system (Cisco IOS).
Unlike a computer, a router does not have video adapters or sound card adapters. Instead, routers
have specialized ports and network interface cards to interconnect devices to other networks.
Figure 9: Hardware components of router
There exist multiple physical interfaces that are used to connect network. Examples of interface
types: Ethernet / fast Ethernet interfaces Serial interfaces Management interfaces Router
provides two fixed 10/100 (100BASE-TX) Ethernet ports, four integrated High-Speed WAN
Interface Card (HWIC) slots that are compatible with WAN Interface Card (WICs), and
Advanced Integration Module (AIM) slot. This router has ability to support up to 96 IP phones.
Figure 10: Router interface
3.2.2 Switches
3.2.2.1 Cisco 2960-24TT: For this project we use Cisco 2960-24TT and one cisco 3560-
24PS switches We configure on them, VLAN and Trucking. C2960-24TT-L is one of the Cisco
Catalyst 2960 Series switches. Cisco Catalyst 2960 Series switches support voice, video, data,
and highly secure access. They also deliver scalable management as your business needs change.
The Common Features are included: Enhanced security including Cisco Trust Sec for providing
authentication, access control, and security policy administration, Multiple Fast or Gigabit
Ethernet performance options, Cisco Energy Wise for power management, Scalable network
management.
Figure 11: Front panel of the Cisco 2960-24TT-L switch.
(1) LED Area
(2) 24 x Ethernet 10/100 ports
(3) 2 x 10/100/1000 TX uplinks
Figure 12: Back panel of the Cisco 2960-24TT-L switch.
(1) RJ45 Console Port (3) RPS Connector
(2) Fan Exhaust (4) AC power connector
3.2.2.2 Cisco Catalyst 3650 Series Switches: is the next generation of enterprise-class
standalone and stackable access-layer switches that provide the foundation for full convergence
between wired and wireless on a single platform.
Integrated wireless controller capability with:
◦ Up to 40G of wireless capacity per switch (48-port models)
◦ Support for up to 50 access points and 1000 wireless clients on each switching entity (switch or
stack)
● 24 and 48 10/100/1000 data and PoE+ models with energy-efficient Ethernet (EEE) supported
ports
● 24 and 48 100-Mbps and 1-, 2.5-, 5-, and 10-Gbps (multigigabit) Cisco UPOE and PoE+
models with EEE1
● Five fixed-uplink models with four Gigabit Ethernet, two 10 Gigabit Ethernet, four 10 Gigabit
Ethernet, eight 10 Gigabit
Ethernet, or two 40 Gigabit Ethernet Quad Small Form-Factor Pluggable Plus (QSFP+) ports
● 24-port and 48-port 10/100/1000 PoE+ models with lower noise and reduced depth of 11.62
inches for shallow depth
cabinets in enterprise, retail, and branch-office environments
● Optional Cisco StackWise-160 technology that provides scalability and resiliency with 160
Gbps of stack throughput
● Dual redundant, modular power supplies and three modular fans providing redundancy2
● Support for external power system RPS 2300 on the 3650 mini SKUs for power redundancy
Switches are used to connect multiple devices together on the same network. In a properly
designed network, LAN switches are responsible for directing and controlling the data flow at
the access layer to networked resources.
Switches use MAC addresses to direct network communications through the switch, to the
appropriate port, toward the destination. A switch is made up of integrated circuits and the
accompanying software that controls the data paths through the switch. For a switch to know
which port to use to transmit a frame, it must first learn which devices exist on each port. As the
switch learns the relationship of ports to devices, it builds a table called a MAC address, or
content addressable memory (CAM) table.
3.2.3 Phones
One of the advantages of VOIP is its user friendly features. We can get VOIP phones in different
ways; it can be in IP-Phone, Softphone and Analog phone with ATA.
3.2.3.1 IP-Phone
We use cisco the Cisco IP Phone 7960. It provides six programmable line/feature buttons and
four interactive soft keys that guide a user through call features and functions. The Cisco IP
Phone 7960 also features a large, pixel-based LCD display. The display provides features such as
date and time, calling party name, calling party number, and digits dialed. The graphic capability
of the display allows for the inclusion of present and future features.
Figure 13: Cisco IP Phone cable connections
1 DC adapter port (DC48V) 5 Access port (10/100 PC)
2 Power supply with AC adapter port plug 6 Handset port
3 Power cable with wall socket plug 7 Headset port
4 Network port (10/100 SW)
3.2.3.2 Softphones: A softphone (software telephone) is an application program that enables

voice over Internet Protocol (VoIP) telephone calls from computing devices. In the enterprise,
softphones are sometimes referred to as soft clients. We use in this project CIPC stands for Cisco
Internet Protocol Communicator CIPC (Cisco Softphone).
Figure 14: Cisco Softphone
CIPC is a Microsoft Windows-based soft-phone application that brings your work telephone to
your personal computer. It is easy to deploy and includes some of the latest technology and
advancements available for IP communications today. This application gives computers the
features of IP phones, enabling high-quality voice calls on the road, in the office, or from
wherever you have access to the corporate network.
3.2.3.3 Analog phones with ATA: An analog telephone adapter (ATA) is a device for
connecting traditional analog telephones, fax machines, and similar customer-premises devices
to a digital telephone system or a voice over IP telephony network. ATA usually has multiple
telephone jacks and an RJ-45 connection to a 10/100BaseT Ethernet hub or switch, and is used to
connect to a local area network (LAN). Such an ATA digitizes voice data, and uses protocols
such as such as H.323 or SIP to communicate directly with a VoIP server so that a softphone is
not required. An ATA that connects telephones to a LAN is sometimes called a VoIP gateway.
The Cisco ATA 186 Analog Telephone Adaptor is a handset-to-Ethernet adaptor that turns
traditional telephone devices into IP devices.
Figure 15: Cisco ATA 186
3.2.4 Cables : Networks use copper media because it is inexpensive, easy to install, and has
low resistance to electrical current. However, copper media is limited by distance and signal
interference.
There are three main types of copper media used in networking: Unshielded Twisted-Pair (UTP)
and Shielded Twisted-Pair (STP). In this project we use Unshielded Twisted-Pair (UTP)
Unshielded twisted-pair (UTP) cabling is the most common networking media. UTP cabling,
terminated with RJ-45 connectors, is used for interconnecting network hosts with intermediate
networking devices, such as switches and routers.
UTP cable is usually terminated with an ISO 8877 specified RJ-45 connector. This connector is
used for a range of physical layer specifications, one of which is Ethernet. The TIA/EIA 568
standard describes the wire color codes to pin assignments (pinouts) for Ethernet cables.
Figure 16:RJ-45 connectors.
Different situations may require UTP cables to be wired according to different wiring
conventions. This means that the individual wires in the cable have to be connected in different
orders to different sets of pins in the RJ-45 connectors.
The following are main cable types that are obtained by using specific wiring conventions:
 Ethernet Straight-through: The most common type of networking cable. It is commonly

used to interconnect a host to a switch and a switch to a router.
 Ethernet Crossover: An uncommon cable used to interconnect similar devices together.
For example to connect a switch to a switch, a host to a host, or a router to a router.
The minimum cabling standard for VoIP is cat5 (Fast Ethernet: 100 Mb/s Ethernet over twisted
pair).
We use also one RJ-11 cable to connect Analog phone with ATA.
3.2.5 Operating Systems

The operating system on a network device is known as a network operating system. The Cisco
Internetwork Operating System (IOS) is a generic term for the collection of network operating
systems used on Cisco networking devices. Cisco IOS is used for most Cisco devices regardless
of the type or size of the device. IOS is a package of routing, switching, security, and other
internetworking technologies integrated into a single multitasking operating system. The IOS file
itself is several megabytes in size and is stored in a semi-permanent memory area called flash.
The switch and router, we use on packet tracer have an IOS Version 12.2(25)FX and Version
12.4(15)T1 respectively.
Cisco IOS routers and switches perform functions that network professionals depend upon to
make their networks operate as expected. Different feature or service has an associated collection
of configuration commands that allow a network technician to implement it. The services
provided by the Cisco IOS are generally accessed using a CLI.
There are several ways to access the CLI environment. The most common methods are:
Console, Telnet or Secure Shell (SSH) and AUX port
Condole: The console port is a management port that provides out-of-band access to Cisco
device. Out-of-band access refers to access via a dedicated management channel that is used for
device maintenance purposes only.
Telnet:Telnet is a method for remotely establishing a command line interface (CLI) session of a
device, through a virtual interface, over a network. Unlike the console connection, Telnet
sessions require active networking services on the device.
Secure Shell (SSH) : The Secure Shell (SSH) protocol provides a remote login similar to Telnet,
except that it uses more secure network services. SSH provides stronger password authentication
than Telnet and uses encryption when transporting session data.
AUX: An older way to establish a CLI session remotely is via a telephone dialup connection
using a modem connected to the auxiliary (AUX) port of a router. Similar to the console
connection the AUX method is also an out-of-band connection and does not require any
networking services to be configured or available on the device. In the event that network
services have failed, it may be possible for a remote administrator to access the switch or router
over a telephone line.
After we connected to a device, it is possible to configure it. On CLI we found different

hierarchical modes this are User executive (User EXEC) mode, Privileged executive (Privileged
EXEC) mode, Global configuration mode, and Other specific configuration modes, such as
interface configuration mode.
To go from one mode to another we use the following scripts:
The user EXEC mode is Identified by “ Switch> “if we want go from this mode to Privileged
EXEC mode We use the following command “ Switch>enable “ . The Privileged EXEC Mode is
identified by “Switch# “. If we want to go from Privileged mode to Global Configuration Mode ,
we use the following commands “Switch# configure terminal”. The Global Configuration Mode
is identified by “Switch(config)#”.
The Cisco IOS modes are quite similar for switches and routers. We perform all of our
configuration on Global configuration mode and we use Privileged executive mode to see what
we do so far.
3.2.6 IP Addressing
Before we start the configuration part. let us talk about IP addressing. We use IPv4,
192.168.0.0/24 , class C class full subletting.
We have four different network this are:
192.168.3.0/24 for Faculty, 192.168.4.0/24 for Students ,192.168.6.0/24 for Management and
192.168.10.0/24 for Voice
3.3 Interconnecting and configuration Network elements

To demonstrate VOIP and VLAN we use a typical LAN. Our typical LAN network consists one
router, one multilayer central switches and 5 switches find on different geographical location, 15
desktop computer plus one laptop with softphone, six IP-phone, printer and one analog phone
with ATA.
After we place the above devices on the working ground , we start the connection. We use
Straight-through cable to interconnect router with switches and switches with computer, printer,
IP phone and ATA. To interconnect switches with switches we use crossover cable. The
interface we use is Fast-Ethernet/IEEE 802.3 interface on router as well as on switches when we
interconnect network elements. Then finally create a topology like as shown in figure below
Figure 17: Our typical LAN network
The next step is configuring the devices. We start our configuration by setting all PC’s on DHCP
mode. IP-Phones on packet tracer they are by default on. Then we go to switches configuration,
on switches we configure VLAN and VLAN trunk.
To configure VLAN on switches we use the following command syntax format steps:
The first step is creating a VLAN.
Switch(config)#VLAN VLAN-id
Switch(config-VLAN)#name VLAN-name
Switch(config-VLAN)#end
The second or the final steps on configuring VLAN is assigning port to VLANS:
Switch(config)#interface interface_id
Switch(config-if)#swithport mode access
Switch(config-if)#switchport access VLAN VLAN_id
Switch(config-if)#end
To configure trunk we use the following Syntax
Switch(config)# interface interface_id
Switch(config-if)#swithport mode trunk
Switch(config-if)#swithport trunk native VLAN VLAN_id
Switch(config-if)#end
After we finished the configuration on switches we go for router. On router we configure DHCP,
Inter-VLAN routing and VOIP.
In our project we create five DHCP POOL for different VLAN’s . Our DHCP pool have the
following name FACL3, MANA6, STUDE4, and VOICE10. Each VLAN found on in different
network. On each network we have ten excluded IP address. To configure DHCP we use the
follow the following steps:
On steps one, we excluding some of IP4 addresses. We assigned them, to network devices that
require static address assignments. Therefore, these IPv4 addresses should not be assigned to
other devices. Excluded addresses should include the addresses assigned to routers, servers,
printers, and other devices that have been or will be manually configured.we use the following
syntax:
Router(config)# ip dhcp excluded-address low-address and high-address
On step two, we configuring a DHCPv4 pool with it’s the range of available addresses and the
default router getways . We use the following syntax
Router(config)# ip dhcp pool pool-name
Router(dhcp-config)# network network-number [mask /prefix-length]
Router(dhcp-config)#default-router address [address2….address8]
The second configuration we do on router is Inter-VLAN routing. As we mentioned earlier we

have 5 VLAN. Each VLAN found on different network. To send packet on different VLAN we
need router that is configured with inter-VLAN routing.
To configer Inter-VLAN routing we use the following syntax format:
Router(config)#interface interface_id subinterface_number
Router(config-subif)#encapsulation dot1Q VLAN_id
Router(config-subif)#ip address ip_address subnet_mask
Router(config-subif)#exit
In our project our router interface_id is FastEthernet0/0 and subinterface_number and VLAN-id
are 3, 4, 6, 10, and 8.
The last configuration we do on router is VOIP. Our router act as soft-PBX.
We follow the following steps
Step one,
Router(config)#telephony-service
Router(config-telephony)#max-dn // Maximum directory numbers supported
Router(config-telephony)#max-ephones // Maximum phones to support
Router(config-telephony)#ip source-address ipv4 port 2000-9999 Define tcp port for Telephony
Service/CM FALLBACK
Router(config-telephony)#exit
Router(config)#ephone-dn 1
Router(config-ephone-dn)#number
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#
Router(config-ephone)#type // 7960 Cisco IP Phone 7960//CIPC Cisco IP Phone CIP// ata

ATA phone emulation for analog phone
Router(config-ephone)#button LINE button-index:dn-index pairs example 1:2 2:5
Configuration line:button
Router(config-ephone)#exit
3.4 Working principle of VLAN

To understand the working principle and advantage of VLAN first let us looks a network without
a VLAN. Take a network shown in fig in this network if someone generate a broadcast every
end device in the network hear this broadcast. This is highly reduce the network performance by
taking so many bandwidth.
One of the technologies used to improve network performance is the separation of large
broadcast domains into smaller ones. By design, routers will block broadcast traffic at an
interface. However, routers normally have a limited number of LAN interfaces. A router’s
primary role is to move information between networks, not to provide network access to end
devices.
The role of providing access into a LAN is normally reserved for an access layer switch. A
virtual local area network (VLAN) can be created on a Layer 2 switch to reduce the size of
broadcast domains, similar to a Layer 3 device. A group of devices within a VLAN communicate as
if they were attached to the same cable. VLANs are based on logical connections, instead of physical
connections. A VLAN creates a logical broadcast domain that can span multiple physical LAN segments.
We can create around 4096 VLAN on one cisco switches. In our project we create 5 VLANs namely:
students, facility, management, voice and native. As shown in the figure the red indicate student, the
yellow indicate the facility, the blue indicate the management and the phones and the Laptop are in voice
VLAN. As shown in the figure this VLAN found on different switches if we want to send data on the
same VLANs we have to vlan first configure trunk between the switches that connect these two VLANs.
VLANs would not be very useful without VLAN trunks. A trunk is a point-to-point link between two
network devices that carries more than one VLAN. A VLAN trunk extends VLANs across an entire
network. Cisco supports IEEE 802.1Q for coordinating trunks. VLAN trunks allow all VLAN traffic to
propagate between switches, so that devices which are in the same VLAN, but connected to different
switches, can communicate without the intervention of a router.
The standard Ethernet frame header does not contain information about the VLAN to which the frame
belongs; thus, when Ethernet frames are placed on a trunk, information about the VLANs to which they
belong must be added. This process, called tagging, is accomplished by using the IEEE 802.1Q header,
specified in the IEEE 802.1Q standard. The 802.1Q header includes a 4-byte tag inserted within the
original Ethernet frame header, specifying the VLAN to which the frame belongs.
Figure 18: VLAN Tag Field
When the switch receives a frame on a port configured in access mode and assigned a VLAN, the switch
inserts a VLAN tag in the frame header, recalculates the Frame Check Sequence (FCS), and sends the
tagged frame out of a trunk port.
VLAN Tag Field Details
The VLAN tag field consists of a Type field, a Priority field, a Canonical Format Identifier field,
and VLAN ID field:
 Type - A 2-byte value called the tag protocol ID (TPID) value. For Ethernet, it is set to
hexadecimal 0x8100.
 User priority - A 3-bit value that supports level or service implementation.
 Canonical Format Identifier (CFI) - A 1-bit identifier that enables Token Ring frames to
be carried across Ethernet links.
 VLAN ID (VID) - A 12-bit VLAN identification number that supports up to 4096
VLAN IDs.
After the switch inserts the Type and tag control information fields, it recalculates the FCS
values and inserts the new FCS into the frame. After the switches add the tag on the Ethernet
frame it send it in the port that is configured with native vlan. In our project all Fast Ethernet 0/1
ports of cisco 2960 switches are configured as a native vlan. All vlan are passed through this
trunked port.
The other question that need to be asked is that, what if end users want to send packet that is
found in different vlan?. The answer is that they have to use router and layer 3 switches. In this
project we use Router –on –stick using router.
Router-on-a-stick’ is a type of router configuration in which a single physical interface routes

traffic between multiple VLANs on a network. The router interface is configured to operate as a
trunk link and is connected to a switch port that is configured in trunk mode. The router performs
inter-VLAN routing by accepting VLAN-tagged traffic on the trunk interface coming from the
adjacent switch, and then, internally routing between the VLANs using sub interfaces. The router
then forwards the routed traffic, VLAN-tagged for the destination VLAN, out the same physical
interface as it used to receive the traffic. In this project we have four sub interface. Voice,
student, Facility and Management sub interface.
Sub interfaces are software-based virtual interfaces, associated with a single physical interface.
Sub interfaces are configured in software on a router and each sub interface is independently
configured with an IP address and VLAN assignment. Sub interfaces are configured for different
subnets corresponding to their VLAN assignment to facilitate logical routing. After a routing
decision is made based on the destination VLAN, the data frames are VLAN-tagged and sent
back out the physical interface. We configure this sub interface on router Fast Ethernet port 0/1
port.
3.5 Working principle of VOIP

Phone registration: Before a VoIP endpoint can make a call, it must first register with the
call server. This process makes the call server aware of the phone and provides information for
the interface on the phone. With a new installation or registration, a user will log into a phone
using the phone number assigned to him or her. At this point, the MAC addresses of the phone
are now tied to that particular phone or dial number.
Registration occurs via the signaling protocol, and each signaling protocol The phones perform
this task before any other.
Call setup and connection
In a traditional network, lifting the receiver closes a circuit in preparation for the voice signal.
Users dial numbers, creating tones that are sent to the telephone switch. The switch converts the
tones to digital information via the codec. The switches must establish an end-to-end circuit to
the destination. None of this is packetized, meaning it is not IP-based on protocols. For VoIP,
this process must now be changed from Signaling System 7 messages and telephone frequencies
(such as those coming from a dual-tone multifrequency, or Dual Tone Multi-Frequency (DTMF),
endpoint) to messages encapsulated in protocols. The VoIP signaling protocol (H.323, Skinny,
SIP) sends messages to the call server, indicating the number dialed, and the call server must
contact the destination. While the protocols have different methodologies, and in fact vendors
may create additional differences, these messages typically appear just before the start of the
RTP stream.
RTP conversation: RTP is used to convey voice data. Once the RTP packets are flowing, the
call has been established. However, RTP can also be used to convey samples created for other
sounds. For example, a dial tone can be placed in RTP packets sent from the call server, and
these packets will occur before the voice data for the call—so keep an eye on the IP addresses.
The RTP packet contains a payload ID indicating the codec used. When the end user speaks into
the handset, the codec takes the analog voice and creates the voice packets sent in the RTP
stream. We will see with some details in chapter four with the packet we are capture using wire
shark.
3.6 ON GNS3
We use two CIPC softphone on the virtual machine and in the real pc. To create a connection we
use cisco 3745 with 12.4(9) T4 IOS image on GNS3. After we create the following topology
Figure 19: GNS3 topology
We configure the syntax shown in appendix on the router. After we finished the configuration we
open the two phones and try to make call from one end to another. We successfully able to make
a connection between the two phone as show in the fig
Figure 20: CIPC Phones
Finally we use Wireshark to capture the packet that transmitted between them. We analysis the
packet we get in chapter 4.
Chapter Four
4. Software Simulation Results and Discussions
4.1 Result we get from Packet Tracer

After we finished all our configuration we use different method to check what we have done so
far is write. To check the result we take two steps in the first steps we try to verify what we did
so far is correct. To do that, we use different “show “command on privileged mode on router
and switches CLI. In the next steps we try to show our network working in the way we want and
the impact of our configuration on the network. To do that we use ping command and call using
the IP phones to show the connection is successful. Especially we use it to show the impact of
VLAN in our network.
We use Wireshark to grasp and show packet that are transmitted in voice communication.First
let us verify VLAN configuration.
We use show vlan brief command on CLI of each cisco switches. It display one line for each
VLAN with the name, status and its ports.
Figure 21: verify VLAN configuration.
we use show interfaces interface-ID switchport command. this command show that port F0/1
has its administrative mode set to trunk. The port is in trunking mode. Also it verifies that the
native VLAN is VLAN 8.
Figure 22: Verify vlan Trucking
The command we use to show the DHCP Configuration is running-config | begin dhcp. This
command help us to show which IP address are excluded, pool name, network and default-router
IP address.
Figure 23: Verify DHCP
On the router we try to verify the inter-vlan routing configuration and VOIP configuration To
verify inter vlan routing configuration. we use the following command, show IP-route
Figure 24: Verify routing table
Show vlan e-phone
Figure 25:Verify e-phone
After verification done we check if the phones get its number from the server. To check end to
end connection, we use ping command on prompt of each pcs and we turn the simulate
a)In the same VLAN b)In different VLAN
Figure 26: Test end to end connection between a) In the same vlan b) in different VLAN.
on in to real time to observe the packet movement. PC 10 and PC 15 ping to PC 3 with an IP

address 192.168.3.11.In each case the ping is successful. PC 10 found in different vlan with PC
3,but the ping is successful this is because of inter vlan configuration.
Voice phone also communicate each other.
We have 8 phone number start from 0921 – 0928.
We have three different kind of phone. The first is analog phone with ATA adapter, Softphone
and IP phone as shown in the figure below. As show in the figure with the read mark they get
their number from the server.
a) Analog phone b) IP phone c) Softphone
Figure 27: Different kind of VOIP phones
Next we check if all phone can make a connection with any other phone they want. In figure
below we showed that, when IP phone with a number 0923 make a call to analog phone with
phone number 0928.
Figure 28:phone 0923 make a call to phone 0928
As show in the figure, phone 0923 make a call to phone 0928, and if somebody answers the call
on phone 0928, the two ends can make voice conversation using IP network.
In cisco packet tracer, it is not possible to capture actuall packet because of that we use GNS3.
And we capture actual packet by using wire shark.
On the wire shark we are going to see the sampling rate of the voice, the coding it use, and the
signaling and transport protocol of voip.
First let as look the transport protocol packet. From different transport protocol, we use SIP
protocol.
In this figure we observed that it have 5 columns. The first column talks about time. It indicates
when the packet is captured. Second and third column shows the source and destination IP
address. The fourth is Protocol column. This column tells us which protocol is in use. As show
in the figure first the SIP protocols and next RTP protocol are shown. The last column show
more information about the capture packet.
4.2 Result we get from GNS3
We use Wireshark to capture the packet that is transmitted between the two end CIPC
phones.We get the result as show in the figure, we use filter future in Wireshark only to look
transport and signaling protocols since these two protocols are our focus.
As shown in the fig SIP protocols occurred first as we discussed in chapter two, to facilitate end
to end communication and the RTP follow to carry real time data.
Figure 29: VOIP protocols
Signaling protocol Sip
Figure 30: Signaling protocol Sip
The above fig show SIP messages. This messages transmitted between the server and the end
device. Transport protocol
RTP
Here we can observe all RTP header, that we have been discuses in chapter two.
Figure 31: RTP header
Figure 32: analog speech
This final simulation result show more information. The graph is shown is make by the analog
speech we make. The highlighted part of the figure contains source and destination port and IP
address, sample rate used and payload.
Chapter Five
5. Conclusion and Recommendations for Future Work
5.1 Conclusion
Global VoIP adoption has increased exponentially over the last few years (see graph below). In
some countries, VoIP is more common than the POTS. This trend is expected to continue and
may mean that VoIP will soon become the default standard for telephony.
Figure 33: Global VOIP Market
But in our country we use our network for limited purposes even in our university LAN. Because
of this we limited the potential of IP network. It is possible to implement with small amount of
money.
As shown the simulation it is possible to implement voice network in a data network, and get its
advantage. When we implement we have to give high priority to QOS to get good real time
conversation result.
5.2 Recommendations for Future Work
It is crystal clear, the leading communication methods now a day and will be in the future is IP
network, there for understanding it deeply is required to use it with its full potential. In this paper
we focused on VOIP protocols, configuration and equipment used.
Some of the topics of IP network that need to be study and improvement are
QOS of VOIP: The Quality of Service is hard to ensure for VoIP especially due to the fact that
the voice packets are sent in the same network as the data packets. In order to maintain the
same level of quality as traditional telephone networks, a VoIP network must limit the packet
loss and the delay.
VOIP security: The most important aspect that cooperate executives should certainly not
overlook when deciding whether to implement VoIP is the security threats.
Wireless VOIP: wires voip provide many advantages than wired one the same as the one we did
in this project. But it is not easy to implement we have to consider many tings including noise.
IP TV: IPTV is defined as multimedia services (Television, Video, Audio, Text, Graphics, data)
delivered over IP based networks on TV; managed to provide the required level of QoS, security
and reliability:
IOT: The Internet of Things (IoT) is a system of interrelated computing devices, mechanical and
digital machines, objects, animals or people that are provided with unique identifiers and the
ability to transfer data over a network without requiring human-to-human or human-to computer
interaction.
Appendix
On packet tracer
DHCP Protocol
Router(config)#
Router(config)#ip dhcp excluded-add 192.168.3.1 192.168.3.10
Router(config)#ip dhcp pool FACL3

Router(dhcp-config)#network 192.168.3.0 255.255.255.0
Router(dhcp-config)#default-router 192.168.3.1
Router(dhcp-config)#exit
Router(config)#ip dhcp pool MANA6

Router(config)#ip dhcp pool STUDE4

Router(config)#ip dhcp pool VOICE10

Router(dhcp-config)#option 150 ip 192.168.10.1
Router(config)#
E-phone configuration
Router#conf t
Router(config)#telephony-service
Router(config-telephony)#max
Router(config-telephony)#max-dn
Router(config-telephony)#max-dn 8
Router(config-telephony)#max-ephones 8
Router(config-telephony)# ip source-address 192.168.10.1 port 2000
Router(config-telephony)#ephone-dn 1
Router(config-ephone-dn)#number 0921
Router(config-ephone-dn)#ephone-dn 2
Router(config-ephone-dn)#exit
Router(config)#ephone 1
Router(config-ephone)#type 7960
Router(config-ephone)#button 1:1
Router(config-ephone)#mac-address
Router(config-ephone)#ephone 2
Router(config-ephone)#type CIPC
Router(config-ephone)#type ata
Router(config-ephone)#
Router#
Inter vlan Configuration

interface FastEthernet0/0.3
encapsulation dot1Q 3
ip address 192.168.3.1 255.255.255.0
Router(config)#interface FastEthernet0/0.3
Router(config-subif)#encapsulation dot1Q 3
Router(config-subif)#ip address 192.168.3.1 255.255.255.0
Router(config-subif)#interface FastEthernet0/0.4
Router(config-subif)#
Router(config-subif)#encapsulation dot1Q 8 native
Router#
On GNS3
R1(config)#interface FastEthernet0/0
R1(config-if)# ip address 10.0.0.1 255.0.0.0
R1(config-if)# duplex auto
R1(config-if)# speed auto
R1(config-if)#no shutdown
R1(config)#ip dhcp pool voice

R1(dhcp-config)# network 10.0.0.0 255.0.0.0
R1(dhcp-config)# default-router 10.0.0.1
R1(dhcp-config)# option 150 ip 10.0.0.1
R1(config)#telephony-service
R1(config-telephony)#max-dn 2
R1(config-telephony)#max-ephones 2
R1(config-telephony)#ip source-address 10.0.0.1 port 2000
Reference
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JIGJIGA UNIVERSITY

Department Of Electronics And Communication

ADVISOR NAME: Mr. Dr. Anbarasu

In partial fulfillment of the requirements for the Degree of Bachelor of Science

Figure 1: Cisco PPDIOO Life Cycle ............................................................................................. 5

JIGJIGA UNIVERSITY III

Table 1: VoIP Advantages and Disadvantages ............................................................................ 10

JIGJIGA UNIVERSITY VII

Voice telephony (highly cost sensitive; needs rock-solid reliability).

1.2 Statement of the Problem

1.3 Study Objectives

1.3.2 Specific Objectives

 To Analysis VoIP system and Protocols.

1.5 Research Methodology

Figure 1: Cisco PPDIOO Life Cycle

1.6 Organization of Study

2.1 History of Telephony

Figure 2: Evolution of Telephones

2.2 Data Networks

2.3 Voice over Internet Protocol (VoIP)

 Low cost  Users cannot make calls during

2.4 VoIP components

Figure 3: VOIP components

Figure 4:VoIP Protocol Architecture

2.5 VoIP compression Algorithms

2.6 VoIP Protocols

2.6.1 Signaling Protocols

2.6.1.1 Session Initiation Protocol

Figure 5: SIP Distribution Architecture

SIP distributed architecture

Figure 6: H.323 protocol header

2.8 Transport Protocol (Real-Time Protocol (RTP))

Figure 7: over all VOIP header

Figure 8: RTP header

Ver, Version. 2 bits: RTP version number. Always set to 2.

CSRC, Contributing source. 32 bits: An array of 0 to 15 CSRC elements identifying the

2.9 Quality Of Service QOS:

Services and Protocols

A great example of a streamlined technique used on larger networks is DiffServ. DiffServ

2.10 Why Change to VoIP?

2.11 Comparison of voip vs. PSTN

Table 3: Comparison of voip vs. PSTN

3.1 System Design and Analysis

3.1.1 Network simulation tools

3.1.2 Network protocol analyzer

3.1.3 Virtual machine

3.2 System components and operation

Figure 9: Hardware components of router

Figure 10: Router interface

Figure 11: Front panel of the Cisco 2960-24TT-L switch.

(1) LED Area

(2) 24 x Ethernet 10/100 ports

(3) 2 x 10/100/1000 TX uplinks

Figure 12: Back panel of the Cisco 2960-24TT-L switch.

(1) RJ45 Console Port (3) RPS Connector

(2) Fan Exhaust (4) AC power connector

Integrated wireless controller capability with:

◦ Up to 40G of wireless capacity per switch (48-port models)

cabinets in enterprise, retail, and branch-office environments

Figure 13: Cisco IP Phone cable connections

1 DC adapter port (DC48V) 5 Access port (10/100 PC)

2 Power supply with AC adapter port plug 6 Handset port

3 Power cable with wall socket plug 7 Headset port