STUDY OF LOAD BALANCING EFFECTIVENESS

BET IN DATA COMMUNICATIONS

IN SIP NETWORK

SITI NABILA BINTI MOHAMAD FAUZI

51213116029

UNIVERSITI KUALA LUMPUR

2019

JANUARY 2019
 STUDY OF LOAD BALANCING EFFECTIVENESS
IN SIP NETWORK

SITI NABILA BINTI MOHAMAD FAUZI

51213116029

Report Submitted to Fullfill the Partial Requirements

For the Bachelor of Engineering Technology (Hons.) in
Data Communications

Universiti Kuala Lumpur

JANUARY 2019

i
 DECLARATION

I declare that this report is my original work and all references have been
cited adequately as required by the university.

Date: ___________ Signature: …………………….

(Hardcover submission date) Full Name: SITI NABILA BINTI
MOHAMAD FAUZI
ID Number: 51213116029

ii
 APPROVAL PAGE

I/We have supervised and examined this report and verify that it meets the
program and university’s requirements for the Bachelor of Engineering
Technology (Hons.) in Data Communications.

Date: ___________ Signature: …………………….

(Hardcover submission date) Supervisor: SIR MOHD ZAIN BIN
ISMAIL
Official Stamp:

Date: ___________ Signature: …………………….

(Hardcover submission date) Co-Supervisor: (if any)
Official Stamp:

iii
 ACKNOWLEDGEMENT

First and foremost, I would like to express my deepest appreciation to my

supervisor for this research study, Sir Mohd Zain Bin Ismail for the valuable
guidance and advice. He inspired me to continue to work on this research.
Without his guidance, this final year project report would not have been
presented.

I also would like to thanks to all UniKL BMI lecturers that thought me
from the first until my final semester. The knowledge learnt throughout my
entire studies here proofed to be valuable especially during conducted this
study. Not to forget to all technicians and staff that have been extremely helpful
to me while doing this research.

An honorable mention goes towards my family members for their

understanding and supports for me throughout my studies in UniKL BMI. My
sincere appreciation extends to my best friends for their supports and help
through the completion of this report.

iv
 ABSTRACT

Load balancing is the ability of the system to distribute network traffic

efficiently. Due to time-sensitive nature of Voice over Internet Protocol, the
system needs to be able to serve with almost excellent quality. Therefore,
implementation of load balancing protocol is crucial in VoIP environment. In
this study, Gateway Load Balancing Protocol (GLBP) is configured in Session
Initiation Protocol (SIP) network which is one of VoIP signaling protocol. SIP
functions to establish the connections between VoIP users as well as manages
and maintains the VoIP session. This research is conducted in simulation
environment by utilizing Graphical Network Simulator 3 (GNS3) software. Two
scenarios is constructed; load balancing scenario and non-load balancing
scenario in GNS3 topology where each of the scenarios is simulated to monitor
SIP behavior in the network and also the differences in term of network
performances and voice quality performances. The data traffics for both
scenarios were captured by using jPerf tools and Omnipeek packet analyzer.
jPerf is used to monitor the network throughput while Omnipeek packet
analyzer is used to monitor jitter, packet loss, delay, Mean Opinion Score
(MOS) and R-Factor during call session. Each of the monitored parameters
were presented and the results of the simulation shown the implementation of
GLBP to provide redundancy and load balancing in the system for smooth call
connection.

v
 TABLE OF CONTENTS

Contents Page No
DECLARATION ii
APPROVAL PAGE iii
ACKNOWLEDGEMENT iv
ABSTRACT v
TABLE OF CONTENTS vi
LIST OF FIGURES ix
LIST OF TABLES xii
LIST OF ABBREVIATIONS xiii

CHAPTER 1: INTRODUCTION
1.1 Background of the Project 1
1.2 Problem Statement 2
1.3 Objectives of the Project 2
1.4 Scopes and Limitations of the Project 3
1.5 Chapter Summary 3

CHAPTER 2: LITERATURE REVIEW

2.1 Introduction 4
2.2 Load Balancing 4
2.2.1 Per-Destination & Per-Packet Load Balancing 5
2.2.2 Hot Standby Router Protocol (HSRP) 6
2.2.3 Gateway Load Balancing Protocol (GLBP) 8
2.3 Voice Over Internet Protocol (VoIP) 9
2.3.1 VoIP Requirements 10
2.3.2 VoIP Protocols 10
2.4 Network Performance 12

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 2.4.1 Delay 12
2.4.2 Throughput 13
2.5 Voice Quality Performance 13
2.5.1 Rating Factor (R-Factor) 13
2.5.2 Mean Opinion Score (MOS) 14
2.5.3 Jitter 15
2.5.4 Packet Loss 15
2.6 Review from previous work/paper 16
2.6.1 Performance of VoIP 16
2.6.2 Load Balancing Operations 18
2.7 Chapter Summary 19

CHAPTER 3: METHODOLOGY
3.1 Introduction 20
3.2 Proposed Methodologies 20
3.2.1 Software and Hardware Requirement 21
3.2.2 Network Topology 22
3.2.3 Data Collection 24
3.3 Flowchart 25
3.4 VoIP Requirements 26
3.4.1 AsteriskNOW 26
3.4.2 AsteriskNOW Installation 27
3.4.3 AsteriskNOW Configuration 34
3.4.4 Zoiper Softphones 35
3.4.5 Zoiper Softphones Installation & Configuration 35
3.5 Simulation Software 38
3.5.1 Graphical Network Simulator 3 (GNS3) 38
3.5.2 GNS3 Installation 38
3.5.3 Configuration of Router Images in GNS3 41
3.5.4 Integration of GNS3 With Virtual Hypervisor 45
(VMware)
3.5.5 Configuration of GLBP On Router in GNS3 46

vii
3.6 Network Monitoring Tools 47
3.6.1 Omnipeek Packet Analyzer 47
3.6.2 jPerf Tools Application 49
3.7 Gantt Chart 50
3.8 Chapter Summary 50

CHAPTER 4: RESULTS AND DISCUSSIONS

4.1 Introduction 51
4.2 Analysis of Voice Packet Capture 51
4.3 Load Balancing Analysis 53
4.4 Network Performance Result 55
4.4.1 Throughput 55
4.4.2 Delay 55
4.5 Voice Quality Result 57
4.5.1 Jitter 57
4.5.2 Packet Loss 58
4.5.3 Mean of Opinion Score (MOS) 59
4.5.4 R-Factor 60
4.6 Chapter Summary 60

CHAPTER 5: CONCLUSIONS AND RECOMMENDATIONS

5.1 Introduction 61
5.2 Conclusions 61
5.3 Recommendations 62

REFERENCES 63

APPENDICES
Appendix A – Presentation Poster 69
Appendix B – Technical Journal 70
Appendix C – Gantt Chart 76
Appendix D – Router Configuration 78

viii
 LIST OF FIGURES

Figure No Descriptions Page No

Figure 2.1 HSRP operation 6
Figure 2.2 GLBP operation 8
Figure 2.3 VoIP network topology 10
Figure 2.4 H.323 architecture 11
Figure 2.5 SIP architecture 12
Figure 2.6 R-value calculations 14
Figure 2.7 Block diagram of load balancing of group of 18
SIP server.
Figure 3.1 Physical and logical topology of VoIP network 22
Figure 3.2 Flowchart 25
Figure 3.3 AsteriskNOW logo 26
Figure 3.4 VMware Workstation Pro setup wizard 27
Figure 3.5 VMware user license agreement 27
Figure 3.6 Installation progress 28
Figure 3.7 Completion of VMware installation 28
Figure 3.8 Creating new virtual machine wizard 29
Figure 3.9 Selecting guest operating system 29
Figure 3.10 Configuration of processor for VoIP Server 30
Figure 3.11 Allocating memory for VoIP server. 30
Figure 3.12 Selecting IOS Image to boot VoIP Server 31
Figure 3.13 Powering on VoIP server 31
Figure 3.14 AsteriskNOW installation wizard 32
Figure 3.15 Selecting Express Installation 32
Figure 3.16 Configuring static IP address and default 33
gateway
Figure 3.17 Installation of packages for VoIP server 33
Figure 3.18 Completion of server installation 34

ix
Figure 3.19 Extension number configured in the server 34
Figure 3.20 Zoiper logo 35
Figure 3.21 Zoiper setup wizard 35
Figure 3.22 License agreement of Zoiper software 36
Figure 3.23 Installation of Zoiper 36
Figure 3.24 Zoiper softphones interface 36
Figure 3.25 Configure new account type 37
Figure 3.26 Configuring account credentials for VoIP users 37
Figure 3.27 Call session between VoIP users in this study 37
Figure 3.28 GNS3 logo 38
Figure 3.29 GNS3 setup wizard 39
Figure 3.30 GNS3 license agreement 39
Figure 3.31 Selecting components for installation 40
Figure 3.32 Installation progress 40
Figure 3.33 GNS3 installation completed 41
Figure 3.34 Adding new router templates on Dynamips 41
Figure 3.35 Selecting IOS image file 42
Figure 3.36 Description of router 42
Figure 3.37 Memory allocation for router 43
Figure 3.38 Selecting ethernet network adapter for router 43
Figure 3.39 Selecting serial modules for router 44
Figure 3.40 Configuration of idle-PC value 44
Figure 3.41 Cisco router added into workspace 45
Figure 3.42 Importing new appliance template 45
Figure 3.43 Adding new VMware virtual machine 46
Figure 3.44 InstallShield wizard for Omnipeek software 47
Figure 3.45 License agreement of Omnipeek software 47
Figure 3.46 Installation progress 47
Figure 3.47 Installation of Omnipeek software completed 49
Figure 3.48 jPerf operation 49
Figure 3.49 jPerf tools application 50
Figure 4.1 Call session summary 51
Figure 4.2 Voice and video visual expert 52

x
Figure 4.3 Snapshot of GLBP implementation 53
Figure 4.4 GLBP status of R5 53
Figure 4.5 GLBP status of R6 54
Figure 4.6 tracert command from PC 1 to 192.168.11.2 55
Figure 4.7 tracepath command from PC 2 to 192.168.14.2 55
Figure 4.8 Network throughput 56
Figure 4.9 Network delay 57
Figure 4.10 Jitter 58
Figure 4.11 Packet loss 59
Figure 4.12 Mean of opinion score 59
Figure 4.13 R-Factor 60

xi
 LIST OF TABLES

Table No Descriptions Page No

Table 2.1 Standard delay in VoIP network 13
Table 2.2 R-Factor based on ITU-T G.107 standard 14
Table 2.3 MOS value for VoIP network 15
Table 2.4 Jitter value based on ITU-T G.114 standard 15
Table 2.5 Packet loss based on ITU-T G.114 standard 16
Table 2.6 Comparisons of various routing protocol. 17
Table 3.1 Software and hardware requirements 21
Table 3.2 Table address of network topology 23
Table 3.3 Extension number for VoIP clients 24
Table 3.4 Recommended requirements 38

xii
 LIST OF ABBREVIATIONS

Abbreviation Descriptions
AVF - Active Virtual Forwarder
AVG - Active Virtual Gateway
ARP - Address Resolution Protocol
DR - Designated Router
DV - Distance Vector
EIGRP - Enhanced Interior Gateway Router Protocol
GLBP - Gateway Load Balancing Protocol
GNS3 - Graphical Network Simulator 3
GUI - Graphical User Interface
HSRP - Hot Standby Router Protocol
HTTP - Hypertext Transfer Protocol
IETF - Internet Engineering Task Force
IOS - Internetworking Operating System
IP - Internet Protocol
ITU - International Telecommunication Union
LS - Link State
MAC - Media Access Control
MCU - Multipoint Control Units
MOS - Mean Opinion Score
MTU - Maximum Transmission Unit
OS - Operating System
OSPF - Open Shortest Path First
PBX - Private Branch Exchange
PC - Personal Computer
QoS - Quality of Service
RAM - Random Access Memory

xiii
RFC - Request for Comments
RIP - Routing Information Protocol
RTP - Real Time Protocol
RTCP - Real Time Control Protocol
SDD - Solid State Drive
SDP - Session Description Protocol
SIP - Session Initiation Protocol
SMTP - Simple Mail Transfer Protocol
SNR - Signal-to-Noise Ratio
TCP - Transmission Control Protocol
UDP - User Datagram Protocol
VoIP - Voice over Internet Protocol
VRRP - Virtual Router Redundancy Protocol

xiv
 CHAPTER 1

INTRODUCTION

1.1 Background of the Study

The title ‘Study of Load Balancing Effectiveness in SIP Network’ has been
chosen as the Final Year Project title for this semester after extensive research
and observation on the current trends occurred in telephony system
nowadays. There is no doubt the Voice over Internet Protocol (VoIP) is the
most preferable form of communication chosen by many due to its ability of
the system to delivered voice and data packets on the same time and
interactively through internet. Because of the huge demand of VoIP, the
system or the network has to be available to serve all the time Furthermore,
the quality of the VoIP also played important role to deliver the content without
any disruption. However, there is no definition of clear traffic in networking
environment. Therefore, implementation of load balancing method is important
for the network devices to share the traffic load and also in case of failover of
the network routes or devices.

This research project indicates the impact of load balancing method by

configuring Gateway Load Balancing Protocol (GLBP) in VoIP system. Two
scenarios; load balancing and non-load balancing will be simulated using
simulation software. Next, the performances of the network and the VoIP
quality will be monitored and evaluated.

1
1.2 Problem Statement

Quality of Service (QoS) is a fundamental of VoIP network. However, it is the

most common problems to provide QoS guarantee over VoIP system. Voice
traffic is sensitive to delayed packets, loss of packets resulting in choppy audio,
long pauses during conversation or interference.

Packet Loss occurs due to various reason such as link failure, transient
network problem or congested network. In VoIP system, the loss of several
packets which as low as 1% can makes the call unintelligible while 5% of
packet losses degrades the whole network (Latif & Malkajgiri, 2007).

Since VoIP relies solely on IP network, the loss connection may occur
due to network congestion or any interference regarding to network thus
leading to jitter. Jitter can be described as disruption of normal sequence of
data packet transmission with the variance in time delay. The time delay of 30
milliseconds or greater effecting the quality of the voice such as distortion or
calls drop (Szigeti & Hattingh, 2004).

1.3 Objectives of the Study

Objective ensure that main focus on this research were right on track. The
research intends to achieve on objectives as state below:

• To study the behaviour of VoIP system preferably SIP protocol.

• To implement various of load balancing configuration on the network

layer and observe the impacts of the method in VoIP system.

• To analyse and inspect each of load balancing schemes in term of the

network performances and voice quality parameters.

2
1.4 Scopes and Limitations of the Study

The primary focus of this research is the implementation of load balancing in

the network layer. Related load balancing configuration protocol will be
applied. The area of research focuses mainly on SIP protocol which one of the
signalling protocols used in VoIP system.

There were few challenges that need to be considered while conducting

the research for instances the limitations of hardware such as routers,
switches, VoIP devices in UniKL BMI communication lab provide with
compatibility issues of the hardware to integrates with the protocols’
configuration. Secondly, the nature of VoIP system itself which takes quite
great amount of time to set up and maintain the system. Besides that, the lack
of understanding regarding to VoIP architecture will affects the evaluation of
the network performances.

1.5 Chapter Summary

This research aimed to investigate the applied load balancing method on SIP
network. The research background, problem statement, scopes and limitations
of the project has been discussed briefly on this chapter.

3
 CHAPTER 2

LITERATURE REVIEW

2.1 Introduction

This chapter briefly discussed about the elements applied for this research in
both theoretical and practical aspects. The discussed matter is important to be
used as a guideline in doing this research. Furthermore, this chapter were
included with few past studies related to the research for better understanding.

2.2 Load Balancing

It is a common practice to implement redundancy protocol over network.

However, instead of focusing on redundant routers or layer 3 switches to route
the network traffic when fail over or network congestion occurs it is a best
practice to implement load balancing protocol. Load sharing or often referred
as load balancing allows two or more interfaces to share the traffic load
(Oppenheimer, 2011).

The concept of load balancing is the ability of the network to transmit

data in more efficient way. The goals for load balancing are to obtain optimal
resource utilization, minimizing response time and maximize throughput rates.
(Ruhann, 2009)

The implementation of load balancing can be done in several ways such

as configuring of network protocol associated with load balancing parameters
on WAN environment for instances Enhanced Interior Gateway Router
Protocol (EIGRP) and Open Shortest Path First (OSPF). Core layer serves as
the backbone of the network as every traffic occurred in the network depends

4
on it. The design of the network on this environment as implementation of
triangle-connection instead of square-connection optimize the network
utilization.

The implementation can also be done in the distribution layer where

critical operation is performed in this area. The load balancing can be facilitated
by configuring channel aggregation where the router able to bring up multiple
channels as the bandwidth requirements increase (Oppenheimer, 2011).
Configuring load balancing parameters such as Gateway Load Balancing
Protocol (GLBP), Hot Standby Router Protocol (HSRP) or Virtual Router
Redundancy Protocol (VRRP) can be implemented on this layer (Cisco, 2008).

2.2.1 Per-Destination & Per-Packet Load Balancing

Per-destination load balancing is the default method enabled on router. The

router distributes the packets by referring to destination address (IP with ease,
2016). For examples, packets from destination A were transmitted through first
router while packets from destination B transmitted through the other routes.
In this method, both route’s bandwidth can be utilized efficiently. Due to the
nature of routers to carry numerous traffic, this method required a lot of
memory, the processing and maintaining the cache (Cisco, 2015).

Per-packet load balancing each packet were distributed along the

provided path. The router transmits one packets through the first route and
another packet through the second routes where the destination of the packets
is same. This method is used to avoid congested network traffic provide equal
utilization among the provided path. Because of the transmission schemes,
the packets may arrive out-of-order due to delay occurred in the network link
(Cisco, 2015).

5
2.2.2 Hot Standby Router Protocol (HSRP)

Hot Standby Router Protocol (HSRP) is one of Cisco’s proprietary protocol

developed as a fault-tolerance mechanism in network. With HSRP, a group of
routers representing the single virtual router to the LAN’s host or known as
standby group. From this group, a single router (active router) is elected as the
responsible router to forwards incoming packets on the virtual router. Another
routers in the group act as the standby router if the active router fail to operate.
(T.Li, B.Cole, P.Morton, & D.Li, 1998)

Figure 2.1: HSRP operation.

HSRP is convenient for host devices that do not compatible with router
discovery protocol. The virtual router was provided with MAC address and IP
address that were shared among the groups. The active router plays primary
role in HSRP operation in processing packets and frames. While the standby
routers check the status of active router whether is operable or vice versa.
Each router in the groups propagate Hello packets to inform their status. The
Hello packets were transmitted using the actual IP address of the router
instead of virtual routers. (Eisazadeh & Espahbodi, 2010)

6
 A state machine is implemented by each of the participating routers in
this group. The state machine as described by RFC 2281 is as follows:

• Initial State – is the first step when there is changes in configuration or

status of interfaces were up. The HSRP is not running on this state.
• Learn – in this state, the router has not yet been provided with virtual IP
address as well as no transmission of Hello packets from active router
to be seen.
• Listen – the routers are not in neither active nor standby routers. The
participated routers listen to the Hello packets.
• Speak – In this state, the transmission of Hello packets was ongoing,
and the routers is actively participating to become active or standby
routers.
• Standby – The candidate of next active router (standby router) sends
periodic Hello packets.
• Active – The router (active router) were operating in forwarding and
processing data packets.

HSRP were provided with features such as pre-emption, interface

tracking and authentication. The pre-emption feature determines the router
with high priority value capability to operates as active router. The interface
tracking is used in HSRP to detect failure interfaces thus decreasing the priority
value. (Cisco, 2011)

7
2.2.3 Gateway Load Balancing Protocol (GLBP)

GLBP is a Cisco’s proprietary standard designed to protect network traffic from

failover while allowing load sharing between alternative routers. Similar to
HSRP/VRRP operation, this standard creates a virtual gateway. (Cisco, 2011)

Figure 2.2: GLBP operation.

One of the routers in GLBP will be elected as gateway to become as

the active virtual gateway (AVG) one router acts as standby for AVG while the
remaining placed in a listen state. In the event of one AVF or known as primary
virtual forwarder fails to operate, it will be replaced by the secondary virtual
forwarder were chosen from the listen state. (Cisco, 2011)

AVG assign virtual MAC address to the backup devices in the group or
known as Active Virtual Forwarder (AVF). AVF learns about AVG status
through the hello messages. The transmission of hello messages was sent
every 3 seconds to multicast address of 224.0.0.102 and UDP port of 3222.
AVG is responsible in answering Address Resolution Protocol (ARP) requests.
(Eisazadeh & Espahbodi, 2010).

8
The benefits of GLBP can be seen as follows (Conlan, 2009):

• Load sharing – GLBP can be configured in a way that traffic inside the
LAN can be shared among multiple routers.
• Multiple Virtual Routers – This standard supports up until 1024 virtual
routers on each router interface and maximum of 4 VF per group.
• Preemption – AVG which were configured with higher priority value can
be pre-empted.
• Efficient Resource Utilization – Since the role in GLBP has been
assigned to each router in the group, this will eliminate the need for a
dedicated backup router since all of the router contribute in handling
network traffic.

2.3 Voice Over Internet Protocol (VoIP)

Voice over Internet Protocol (VoIP) technology is a form of communications

that were solely based on internet protocol instead of typical analog telephone
lines. (Hernandez, 2017)

As the system were purely packet-switched based network where the

data (voice) were split into packets containing destination identities that will be
routed through the internet. Unlike the analog lines, VoIP system can be
implemented in flexible ways where the system can be implemented over
traditional analog lines to reduce the cost of the implementation. It can also be
implemented where the voice data were routed to the call center before
reaching its destination (Mehta & Udani, 2001).

9
2.3.1 VoIP Requirements

Figure 2.1 below shows simple VoIP network topology along with the required
equipment in the system (Wu, 2008):

Figure 2.3: VoIP network topology.

• VoIP Gateway – Functions as converter to convert VoIP calls to/from

traditional telephone lines and to connect between the Private Branch
Exchange (PBX) and IP network.
• Gatekeeper – Acts as routing and central manager by managing the
nodes connected in the network. It also handles all call connections,
terminals, gateways and multipoint control units (MCU). This equipment
is optional in VoIP network.
• Clients – The device used by the clients can be IP phone, softphones,
VoIP software installed in workstation and much more.

2.3.2 VoIP Protocols

There are two standard protocol used in VoIP; H.323 and Session Initiation
Protocol (SIP). H.323 is a standard developed by International
Telecommunication Union (ITU) to support media communication such as
video conferencing (Karim, 1999).

10
 H.323 is designed to be applied above on the transport layer where the
standard was based on data packet for instances Real-Time Protocol (RTP)
and Real-Time Control Protocol (RTCP). This standard specifies a few
protocols which are Q.931, H.225, H.245 and ASN.1 for real-time between to
nodes to communicate. The implementation of this standard required
gateways, MCU and gatekeepers (Mehta & Udani, 2001).

Figure 2.4: H.323 architecture.

Another protocol used in VoIP system is Session Initiation Protocol

(SIP). SIP is signaling protocol developed by Internet Engineering Task Force
(IETF), published as RFC 3261 (Cisco). This protocol is a lightweight protocol
used to initiate, maintain and VoIP session. The initial focus of this protocol
was voice data, later it was expanded to support rich-media communications
(Wu, 2008).

This protocol works similar as Hypertext Transfer Protocol (HTTP) or

Simple Mail Transfer Protocol (SMTP) where these protocols located on
application layer protocol. The protocol defines the user/subscriber location
along with its capabilities to the services. (Mehta & Udani, 2001)

SIP integrates with other layer protocol such as User Datagram Protocol
(UDP), Transmission Control Protocol (TCP), RTP and many more to carry the
real-time data over the network. SIP is extensible as this protocol were
important in implementing Quality of Service in VoIP network. Besides the
protocol is way simpler as SIP requires few steps to establish sessions while
H.323 use lots of steps to establish and manage the session.

11
 Figure 2.5: SIP architecture.

2.4 Network Performance

Network performance can be defined as the quality of the service provided in

a network and is measured via various parameters such as bandwidth,
throughput and end-to-end network delay (Rogier, 2016).

2.4.1 Delay

According to ITU, one-way or end-to-end telephony communication should be

less than 150ms. Delay caused by various factors such as VoIP devices,
circumstances of network environment etc. (Zheng, Zhang, & Xu, 2001).

Some paper also states that delay of network up to 200ms is still

acceptable while the delay more than 200ms can be considered as bad voice
quality (James, Chen, & Garrison, 2004). The acquire time delay in VoIP
system or often called latency can be measured from the moment the user
speaks until the other end side hears it (Mehta & Udani, 2001).

Interactive of voice communications depends on the value of latency,

the lower the latency the conversation on both sides can be assumed as
natural conversations. Table 2.1 below shows the standard delay in VoIP
network.

12
 Table 2.1: Standard delay in VoIP network. (Cahyadi, Santoso, & Zahra,
2013)

Delay (ms) Quality

0 – 150 Good
150 – 400 Fair
>400 Bad

2.4.2 Throughput

Throughput is total numbers of successful delivered packets over the provided

network capacity bandwidth and is measured in bits per seconds. (Rohal,
Dahiya, & Dahiya, 2013)

2.5 Voice Quality Performance

There are several parameters to be measured when it comes to the quality of

the voice data which is jitter, packet loss, Mean Opinion Score (MOS) and
Rating Factor (R-Factor). ITU-T provided two methods in measuring voice
quality which is subjective method by evaluating speech quality according to
the Mean Opinion Score (MOS). Another method is objective method by
evaluating the Rating Value or Rating Factor (R-Factor). (Assem, Malone,
Dunne, & Sullivan, 2013)

2.5.1 Rating Factor (R-Factor)

R-Factor is a transmission-rating factor the quality of voice packets in VoIP. It

is a scalar prediction with the range between 0 to 100 for voice network
(Miloucheva, Nassri, & Anzaloni, 2004). Table 2.2 below shows R-Factor
values on VoIP network

13
 Table 2.2: R-Factor value based on ITU-T G.107 standard. (Chochol, 2009)

Range Value Quality

90 – 100 Very satisfied
80 – 90 Satisfied
70 – 80 Some users satisfied
60 – 70 Many users dissatisfied
50 – 60 Nearly all users dissatisfied

The end devices, network disruption, noises, delay, packet losses and
compression algorithm contributed to the value of R-Factor. The value can be
calculated as shown below.

Figure 2.6: R-value calculations.

R0 is the signal-to-noise ratio (SNR) which the sources of noise from the
circuitry, the surrounding and the subscriber line. Is indicates the impairment
that simultaneously occur with the voice signals. Id indicates the impairments
cause by delay which is the delay presented in the network and listener echo
loudness rating. Ie-eff is the equipment impairment factor due to distortion. The
major of this impairment are the voice compression codec and end-to-end
packet impediments. Lastly, A indicates the advantage user which the client
tolerance to the degradation of the voice. (ITU-T, 2005)

2.5.2 Mean Opinion Score (MOS)

MOS is a system grading for the voice quality of the telephone connection that
implementing codec algorithm. The scores were displayed in numerical form
indicating the perceived quality of the voice after the compression and
transmission. The single number is expressed with the range between 1 to 5
which is from the lowest to the highest . (Ismail, 2009)

14
 Table 2.3: MOS value for VoIP network. (Ismail, 2009)

MOS Quality
5 Excellent
4 Good
3 Fair
2 Poor
1 Bad

2.5.3 Jitter

Jitter is the variation of the delay of received packet. On the source, the
packets were sent continuously with constant space. Transmission of the
packet in congested traffic environment, improper queuing causing the space
between each packet become bigger (Cisco, 2006). According to Cisco, the
acceptable value jitter should be below 30ms.

Table 2.4: Jitter value based on ITU-T G.114 standard. (Cahyadi, Santoso, &
Zahra, 2013)

Jitter (ms) Quality

0 – 20 Good
20 – 50 Acceptable
>50 Poor

2.5.4 Packet Loss

Packet loss happens when the transmitted packets fail to reach its destination
(Rouse, 2007). As VoIP are time-sensitive, packet loss undoubtedly effects the
quality of voice. UDP cannot provide the guarantee that all of the packets will
be transmitted orderly while TCP on the other hand required a lot of process.
Packet loss happens due to numerous reasons such as high latency as the
round-trip time of the packet took much more time, interference during call
session, or dropped session (Vouzis, 2016).

The rate of 1 – 3% makes the call session becomes unintelligible

however the value can still be considered as tolerable value in networking.

15
While the loss rates of 5% degrades the whole performance of the network
(Latif & Malkajgiri, 2007) .

Table 2.5: Packet loss based on ITU-T G.114 standard. (Cahyadi, Santoso, &
Zahra, 2013)

Packet Loss (%) Quality

0–1 Good
1–5 Acceptable
>10 Poor

2.6 Review from previous work/paper

Several academic sources include papers, journals, article were reviewed and
evaluate to gain better understanding regarding the research project. There
were many approaches conducted to study the behavior of VoIP with regards
to its performances and the architecture. Most of the reviewed papers states
how popular this technology and the higher demands to implement this
technology in organization.

2.6.1 Performance of VoIP

Referring to Assem et. al paper, the authors describes the objective and
subjective methodology to measure the quality of VoIP packets. The authors
proposed the improvisation of the existing E-model which was designed to
provide estimation of the network quality and the performance quality of VoIP.
The authors focused on a few parameters that played an important role to
determine the VoIP system quality which is Signal-to-Noise Ratio (SNR),
codec impairment, packet loss and delay impairment. The obtain R-value from
the proposed system mapped to MOS rating.

Another paper that has been reviewed related to the study of the VoIP
performances by applying different IP configuration protocol in network layer.
Che et. al state their preferences on how each of the IP routing protocol effect
the VoIP performances. The studied routing protocol used in this research are

16
RIP version 1, EIGRP and OSPF. Each of the routing protocols were evaluated
along with commonly used VoIP performances metric, which is delay, jitter,
packet loss and MOS. From the conducted research, the authors found that
the OSPF implementation is the most effective to provide better VoIP
performances because of the resiliency and its efficiency to support numerous
VoIP protocols. The comparisons of the of various routing protocol is shown
below in table form.

Table 2.6: Comparisons of various routing protocol.

RIP OSPF EIGRP

Type DV LS Hybrid

Medium
Area Small networks Enterprise networks
networks

Classful routing
Routing Classless and
loop with counter Classless
mechanism loop-free
mechanism

Available
bandwidth,
The bandwidth of
Metrics Number of hops delay. Load,
links (inverse value)
MTU and the
link reliability

DR transmit
Broadcast Dual multicast
Discovery multicast packets
periodical incrementally
and Updates every time the
updates updates
changes is made

Failure
Slow Faster than RIP Dual algorithm
Recovery

Supports
maximum of 6
Supports maximum unequal paths
Supported only of 6 equal-cost but were
Load
on equal-cost routes however, it is ignored due to
Balancing
paths difficult to configure its complexity
and implement. and instability of
the
implementation.

17
2.6.2 Load Balancing Operations

Load balancing can be implemented in various forms. A few papers were

focused on implementation of server load balancing or server clustering
methods. The paper written by Kambourakis et. al, Manikandan and Vanilla
and Jiang et. al emphasizes the clustering of SIP server and proposed how
load balancing can be achieved.

For instances, according to Jiang et. al, the authors describe load
balancing algorithms that were based on assigning the calls to the server
(backup server) which has least amount of work assigned in the event of traffic
congestion in the network. The load balancer acts as controller which
determines and assigned the work to the associated server. All of the response
and request will go through the load balancer and it also responsible to forward
the response to the destination address. As load balancer monitors the
performances and the utilization of the group of SIP servers, it able to
determines which server has done processed the request and assigned a new
request query. Figure below shows the block diagram of how load balancing
was implemented in this paper.

Figure 2.7: Block diagram of load balancing of group of SIP server.

While a few of written papers regarding with the implementation of load

balancing in network layer also has been reviewed as the main approach in
this research is to implement the same schemes on the network layer. Bernat
conducted research with the title of ‘Redundancy and load balancing at IP layer
in access and aggregation networks’ He states that the implementation of load

18
balancing schemes is important in every part of the network mainly on the edge
network where the area have less mechanism of protection in case of any
failures. The authors indicate on the implementation of IP virtual redundancy
as these methods also allow load sharing of traffic between the actual gateway
and the backup gateway. This method proves the ability of the load balancing
in reducing the load stress on certain network devices thus allowing the
network becomes more efficient.

Another paper written by Eisazadeh et. al with title of ‘Fast fault recovery
in switched networks for carrying IP telephony traffic’ also emphasize the
implementation of the same method which is the configuration of load
balancing protocol in network layer. The targeted area is on the switched
network which provides the access to the clients. A few protocols have been
implemented and were evaluated such as HSRP, VRRP and so on. The
authors also state loop prevention method while applying the load balancing
protocol.

2.6 Chapter Summary

This section includes all of the academic sources obtained related to the
research. Each of the key elements in the research for instances load
balancing, quality of service and much more were described for better
understanding on theoretical aspects of the research. This section helps in
choosing suitable method to applied and implemented which will be elaborated
on methodology chapter.

19
 CHAPTER 3

METHODOLOGY

3.1 Introduction

Methodology is a set of guidelines in order to gather scientific knowledge on

certain issue. It can also be defined about what suitable method to be applied
in investigating a problem in terms of identifying techniques, analyzing the
information, evaluate the situation and so on. Methodology is one of the crucial
parts while doing any project or research to ensure the researches achieved
its objectives. (USC Libraries, 2018)

This chapter will explain the methods implemented while conducted the
research in detail. This chapter also includes the design of the network,
flowchart, software and hardware requirements and work plan.

3.2 Proposed Methodologies

The research focused on implementation of load balancing protocols to

monitor its effectiveness on the IP telephony network. The research is
conducted in simulation mode using Graphical Network Simulator 3 (GNS3)
software where few scenarios were being simulated. The comparison between
the scenario includes with the performance evaluation in terms of the SIP
packets performance and the voice quality were recorded.

20
3.2.1 Software and Hardware Requirement

GNS3 is used to simulate the VoIP network as the software capable to simulate
into real time environment. Two scenarios have been constructed with each of
the scenarios is configured with two different network protocol mainly to
compare the result between load balancing environment and non-load
balancing environment.

Basic VoIP network were constructed consists of one VoIP server,

routers, switches, and several VoIP clients. AsteriskNOW is installed on virtual
server as the operating system for VoIP server. For VoIP clients, PC-to-PC
type of connections is implemented by installing Zoiper softphone in local
computer and virtual computers. Cisco routers 3745 and switches are used to
forward the data and voice traffic in the network.

The call session is monitored using Omnipeek software while the

network traffic is monitored using jPerf application. Omnipeek is installed on
local computer while jPerf in installed on local computer to acts as server and
on each virtual computer as clients. Table 3.1 below shows the software and
hardware implemented for the purpose of this study.

Table 3.1: Software and hardware requirements.

Devices Operating System/Application Quantity

Graphical Network Simulator 3
Simulator 1
(GNS3)
Virtual
Computer VMWare Workstation Player 1
Simulator
Local Computer Windows 7 Professional 1
Virtual
Ubuntu 14.04 LTS 5
Computer
VoIP Server AsteriskNOW 1
Routers Cisco 3745 6
Switch Cisco 2960 1
Softphone Zoiper 6
VoIP Analyzer Omnipeek 1
Network
jPerf 6
Analyzer

21
3.2.2 Network Topology

Figure 3.1 shows logical and physical diagram of two scenarios of VoIP
network that has been mapped for this purpose of study.

Figure 3.1: Physical and logical topology of VoIP network.

The VoIP Server is situated on local network (192.168.10.0/24) with two

VoIP clients (PC1 and PC2) installed on it. The GLBP configuration is
implemented on both routers R5 and R6 with the virtual address of
192.168.10.1. Each of the external network is installed with one VoIP clients.
Figure 3.1 shows the implementation of the network for load balancing
operation. The same topology also is used for non-load balancing operation.
Table 3.2 below shows the table address of the network topology while table
3.3 shows list of VoIP extensions configured.

22
 Table 3.2: Table address of network topology.

Device IP Address Subnet Mask Interface

192.168.11.1 /24 F0/0

10.1.1.1 /30 S0/0

R1
10.1.2.1 /30 S0/1

10.1.3.1 /30 S0/2

192.168.12.1 /24 F0/0

R2 10.1.3.2 /30 S0/0

10.1.4.1 /30 S0/1

192.168.13.1 /24 F0/0

R3 10.1.4.2 /30 S0/0

10.1.5.1 /30 S0/1

192.168.14.1 /24 F0/0

R4
10.1.5.2 /30 S0/0

192.168.10.2 /24 F0/0

R5
10.1.1.2 /30 S0/0

192.168.10.3 /24 F0/0

R6
10.1.2.2 /30 S0/0

VoIP Server 192.168.10.10 /24 F0/1

PC 1 (Local
192.168.10.11 /24 F0/2
PC)

PC 2 192.168.10.12 /24 F0/3

PC 3 192.168.11.2 /24 F0/1

PC 4 192.168.12.2 /24 F0/1

PC 5 192.168.13.2 /24 F0/1

PC 6 192.168.14.2 /24 F0/1

23
 Table 3.3: Extension number for VoIP clients.

Device Extension
PC 1 (Local PC) 6000
PC 2 6001
PC 3 6002
PC 4 6003
PC 5 6004
PC 6 6005

3.2.3 Data Collection

Data collection is essential in any research-based project. This study applied

quantitative data collection. The data is presented in numerical form which are
clear and precise. Beside the recorded data can be transformed into graphical
view such as graphs and charts.

This study intended to monitor the VoIP network in two different

scenarios which is GLBP configuration is implemented indicating load
balancing operation in the network. While another scenario is non-load
balancing operation which is implementation of HSRP configuration.

The data collected is divided into two aspects; network and voice
quality. For network, delay is measured by using ‘ping’ command and
throughput is measured by using jPerf tools application. For voice quality
parameters; jitter, MOS, R-Factor, packet loss is recorded by using Omnipeek
packet analyzer. The data is recorded by making two call session from local
network (192.168.10.0/24) to external networks and data packets from local
PC (PC 1) is captured for 60 seconds.

24
3.3 Flowchart

Figure 3.2: Flowchart.

25
Figure 3.2 above shows flowchart of the procedure taken for this project. The
project began with gathering required data related to the project such as VoIP
network architecture, software and hardware required and suitable protocols
to be implemented on the topology. Next, installation of software and hardware
and pre-configuration on each network devices. Each of the devices are linked
with correct cable type of cable and at proper interfaces, assigned with
hostname and IP address and configured with routing protocols. As for VoIP
clients, each client is assigned with extension number prior to make call
session. Then, configuration of GLBP for load balancing scenario and HSRP
for non-load balancing scenario on R5 and R6. The next phase is ensuring
connectivity status by ping end-to-end devices and initiating call session. If the
devices are not able to connect to each other, troubleshooting process is
required. The packet capture monitoring tools is executed along with jPerf
application then two call session is done from local network (192.168.10.0/24)
to external network and the call process is monitored and the required data is
collected for further analysis.

3.4 VoIP Requirements

The VoIP requirements used in this research is state as below;

3.4.1 AsteriskNOW

Figure 3.3: AsteriskNOW logo.

AsteriskNOW is an open source software implementation of private

branch exchange (PBX). AsteriskNOW is developed by Sangoma
Technologies Corporation and is commonly used in managing VoIP related
session. (Meggelen, Madsen, & Smith, 2007)

26
3.4.2 AsteriskNOW Installation

Before installing VoIP server, the hypervisor software needs to be installed to

run the server on local computer. Figures below shows step by of installing
VMware Workstation Pro. (VMware Inc., 2013)

Figure 3.4: VMware Workstation Pro setup wizard.

Figure 3.5: VMware user license agreement.

27
 Figure 3.6: Installation progress.

Figure 3.7: Completion of VMware installation.

After VMware is installed, the installation process of AsteriskNOW as

VoIP server is done on the software. Figures below shows step by step
installation process of AsteriskNOW.

28
Figure 3.8: Creating new virtual machine wizard.

Figure 3.9: Selecting guest operating system.

29
Figure 3.10: Configuration of processor for VoIP server.

Figure 3.11: Allocating memory for VoIP server.

30
Figure 3.12: Selecting IOS Image to boot VoIP Server.

Figure 3.13: Powering on VoIP server.

31
Figure 3.14: AsteriskNOW installation wizard.

Figure 3.15: Selecting Express Installation.

32
Figure 3.16: Configuring static IP address and default gateway.

Figure 3.17: Installation of packages for VoIP server.

33
 Figure 3.18: Completion of server installation.

3.4.3 AsteriskNOW Configuration

AsteriskNOW is accessible via web graphical user interface. Any configuration

of the server such as extension number, trunk, network were configured in the
web by accessing through configured IP address. Figure 3.19 below shows
the extension numbers configured for the VoIP users in this study.

Figure 3.19: Extension number configured in the server.

34
3.4.4 Zoiper Softphones

Figure 3.20: Zoiper logo.

Zoiper is free cloud-based softphones application that provides media

communication between VoIP users. This application is installed on all 6 PC
and were configured with suitable VoIP configurations to enable the VoIP
session.

3.4.5 Zoiper Softphones Installation & Configuration

Zoiper installation on both OS which is Windows 7 and Ubuntu 16.04 is quite

similar. For the configuration process, new account is created of each VoIP
users that were assigned with extension numbers in this study. Figures below
are installation and configuration process of VoIP softphones. (Zoiper, 2018)

Figure 3.21: Zoiper setup wizard.

35
Figure 3.22: License agreement of Zoiper software.

Figure 3.23: Installation of Zoiper.

Figure 3.24: Zoiper softphones interface.

36
 Figure 3.25: Configure new account type.

Figure 3.26: Configuring account credentials for VoIP users.

Figure 3.27: Call session between VoIP users in this study.

37
3.5 Simulation Software

This study mainly depends on the application of simulation software due to

limited of hardware resources and its capability to perform desired operation.
The chosen software used in this research is Graphical Network Simulator
(GNS3).

3.5.1 Graphical Network Simulator 3 (GNS3)

Figure 3.28: GNS3 logo.

This software released in 2008 allowing the combination of virtual and

real element of networking. GNS3 is a network software emulator used to
simulate complex network while monitoring the network performance. (Bombal
& Duponchelle, 2019)

3.5.2 GNS3 Installation

Installation of GNS3 software is done on Windows based OS. Table 3.4 shows
Recommended requirements for GNS3 to be able to run smoothly on PC.

Table 3.4: Recommended requirements. (Coleman, Bombal, Duponchelle, &

Ganancial, 2019)

Item Requirement
Operating
Windows 7 (64 bit) or latest
System
Processor 4 or more Logical cores
Virtualization Virtualization extensions need to be enabled
Memory 16 GB RAM

38
 Figures below show step by step of GNS3 installation done in local PC.
(Coleman, Bombal, Duponchelle, & Ganancial, 2019)

Figure 3.29: GNS3 setup wizard.

Figure 3.30: GNS3 license agreement.

39
Figure 3.31: Selecting components for installation.

Figure 3.32: Installation progress.

40
 Figure 3.33: GNS3 installation completed.

3.5.3 Configuration of Router Images in GNS3

GNS3 provide various way to emulate Internetwork Operating System (IOS)

images in the software. Since this study requires Cisco router 3745, Dynamips
emulator is used to install the Cisco IOS image. Dynamips is Cisco router
emulator to run Cisco IOS images platforms series (Bodnárová, Hátaš,
Olševičová, Soběslav, & Štefan, 2010). Figures below shows the installation
and configuration process of Cisco router image on GNS3 (Duponchelle,
2018).

Figure 3.34: Adding new router templates on Dynamips.

41
Figure 3.35: Selecting IOS image file.

Figure 3.36: Description of router.

42
 Figure 3.37: Memory allocation for router.

Figure 3.38: Selecting ethernet network adapter for router.

43
 Figure 3.39: Selecting serial modules for router.

Figure 3.40: Configuration of idle-PC value.

After the installation of Cisco router is complete. The device is dragged into
workspace.

44
 Figure 3.41: Cisco router added into workspace.

3.5.4 Integration of GNS3 With Virtual Hypervisor (VMware)

As the virtual machines which in this study is AsteriskNOW as VoIP server.
Ubuntu 16.04 as clients is already installed on VMware software, the next step
is integration of the virtual machines in GNS3 topology. Figures below show
integration process of VMware with GNS3 (Bombal & Duponchelle, 2018).

Figure 3.42: Importing new appliance template.

45
 Figure 3.43: Adding new VMware virtual machine.

3.5.5 Configuration of GLBP On Router in GNS3

In this study, Gateway Load Balancing Protocol (GLBP) is configured on R5

and R6 based on designed topology. GLBP protocol is configured for load
balancing scenario. Both routers interface (R5 and R6) which connected to
192.168.10.0/24 network is enabled with GLBP as commands below;

R5(config)#interface f0/0

R5(config-if)#glbp 5 ip 192.168.10.1

R5(config-if)#glbp 5 priority 105

R5(config-if)#glbp 5 preempt

R6(config)#interface f0/0

R6(config-if)#glbp 5 ip 192.168.10.1

46
3.6 Network Monitoring Tools

Network monitoring describe the practice of supervise and inspection of

network operation by using specified software tools. It is important to conduct
monitoring to continuously keep track of the performances status in the
network system as well as to detects any failures in the network system (Talley,
2014).

There were two type of monitoring; active and passive. Active

monitoring generates probe packets for the to monitor how the packets travels
in the network (Venkat Mohan, 2011). In this study, active monitoring
measures is applied by installing Omnipeek packet analyzer and jPerf software
to measure the voice and network packets.

3.6.1 Omnipeek Packet Analyzer

Omnipeek is a packet analyzer developed by Savvius. For this study,

Omnipeek is used to monitor the voice network by capturing the packets of call
session of VoIP clients. Figures below show installation of Omnipeek software.
(Savvius Inc., 2016)

Figure 3.44: InstallShield wizard for Omnipeek software.

47
Figure 3.45: License agreement of Omnipeek software.

Figure 3.46: Installation progress.

48
 Figure 3.47: Installation of Omnipeek software completed.

3.6.2 jPerf Tools Application

jPerf tools is a GUI based of network monitoring tools, Iperf. It is used to

determine network throughput, packet loss jitter and other network issues. For
this study, all of PC in the both scenarios is installed with jPerf tools where PC1
which is from local network (192.168.10.0/24) runs as a server and PCs
outside of the network runs as clients. Figure 3.50 below illustrate how jPerf is
operated.

Figure 3.48: jPerf operation.

To run jPerf tools on PC, click jperf.sh files from the extracted files as
shown as figure 3.51 below.

49
 Figure 3.49: jPerf tools application.

3.7 Gantt Chart

Refer to appendix C.

3.8 Chapter Summary

Methodologies played important role to ensure all the required element needed
in research were achieved. The chosen method to conduct the research is by
simulation with integration of monitoring tools software to monitor the network
and VoIP quality performance.

50
 CHAPTER 4

RESULT AND ANALYSIS

4.1 Introduction

This chapter explained the detail about the results collected from this study
and each of the results is analysed based on the theory and practical
implementation. The results were recorded via Omnipeek packet analyser and
jPerf tools. This chapters also compared both of results from two different
scenarios; load balancing and non-load balancing scenario to review the
impact of the implementation of the load balancing protocol on SIP network.
This chapter then briefly explained about how SIP protocol and GLBP protocol
worked during conducting this study.

4.2 Analysis of Voice Packet Capture

Omnipeek packet analyser is a powerful tool to captured network packets

especially VoIP packets. The software provides analysis of the captured data
in detail. The voice or media data can be viewed on Voice and Video options
on the left pane. By clicking on the desired captured call session, a new
window tab opened Voice & Video Visual Expert tab as shown in figures below.

Figure 4.1: Call session summary.

51
 Figure 4.2: Voice and video visual expert.

Figure 4.2 above shows various of lines indicating the flow of the
network during a call session represent signalling and media streams packet.
The captured events were originated from PC 1 (192.168.10.200) with dial
extension of 6000 to PC 3 (192.168.11.2) with dial extension of 6002. Based
on the image above, the line that begin with a small diamond indicates the
initiation of call event (SIP INVITE). SIP sends these messages in UDP on port
5060. The invitation includes call setup and parameters of audio that is used
for the call events. These are included in Session Description Protocol (SDP).
After both clients agree, they started to exchange media via RTP as
represented with a grey line. Green lines indicate R-Factor values and blue
lines indicate jitter value for this call event. This call event ended when 6002
terminate the call session (BYE).

52
4.3 Load Balancing Analysis

Figure 4.3 shows the snapshot on GLBP implementation in this study.

Figure 4.3: Snapshot of GLBP implementation.

GLBP network protocol is configured on R5 and R6 and is implemented

on the third layer of OSI network model. The virtual IP address of 192.168.10.1
is assigned on both R5 and R6. R5 is assigned with IP of 192.168.10.2 and
R6 is 192.168.10.3. Figure 4.4 and 4.5 below shows the status of configured
GLBP on R5 and R6 when issued command #show glbp.

Figure 4.4: GLBP status of R5.

53
 Figure 4.5: GLBP status of R6.

When issued #show glbp command on both routers, it is shown that role
of R5 as an AVG and AVF while R6 as AVF. R5 is elected as AVG as the
priority is configured to 105 and standby router is R6 in case of router failover.

The virtual MAC address is assigned to both forwarders. Virtual MAC

address for GLBP protocol is reserved as 0007. 𝑏400. 𝑥𝑥𝑦𝑦 where 𝑥𝑥 is the
group number and 𝑦𝑦 is the AVF number (Froom & Frahim, 2015). R5 is
assigned with 0007.b400.0502 shows that configured GLBP group is 5 and the
AVF number is 2. While R6 is assigned with 0007.b400.0501 shows that
configured GLBP group is 5 and the AVF number is 1.

The issued command also shows Hello packets timer which is sent
every 3 seconds by routers to updates their status and to detect any router
failure in the network. The Hello packets were sent to multicast address
224.0.0.102. (Froom & Frahim, 2015)

For this study, round-robin algorithm is configured as load balancing

method. Network traffic is distributed evenly across the group of GLBP routers
(Conlan, 2009). ARP request from PC 1 is responded by AVG with MAC
address of forwarder 2 (192.168.10.2) while ARP request from PC 2 is
responded by AVG with MAC address of forwarder 1 (192.168.10.3). Figure

54
4.6 and 4.7 below show the network path of PC 1 and PC 2 to its desired
destination.

Figure 4.6: tracert command from PC 1 to 192.168.11.2

Figure 4.7: tracepath command from PC 2 to 192.168.14.2

Figure 4.6 shows that traffic from PC 1 (192.168.10.200) to

192.168.11.2 is forwarded via R5 while traffic from PC 2 (192.168.10.201) to
192.168.14.2 is forwarded via R6 thus achieving load balancing objective.

4.4 Network Performance Result

Monitoring network performance is essential to check the network system

functionality, detect and troubleshoot error. For this study two parameters
have been taken into consideration for monitoring process; throughput and
network delay. Network throughput is measured by using jPerf tools while
network delay is measured by using ping command on command prompt.

4.4.1 Throughput

Figure 4.8 shows network throughput on two different network protocols. For
voice data communication, it requires higher throughput value in order to
transmit voice data. Higher throughput value means the transmission of data

55
from one point to another is fast and efficient. Figure below indicates that GLBP
network protocol provides higher throughput values compare to network
protocol.

Figure 4.8: Network throughput.

4.4.2 Delay

Delay can be described as time taken for transmission of data from source to
its destination. Figure 4.9 shows the delay value comparison between GLBP
network protocol and HSRP network protocol. It can be concluded HSRP take
more time to transmit voice network because the traffic of two call session from
local network (192.168.10.0/24) were going through one router which is R5
(192.168.10.2) since R5 is elected as active router. GLBP network protocol
take less time to transmit data as the traffic were distributed equally among
two routers in GLBP group. Besides, the number of network devices used (in
this case are routers) used to transmit the data leads to high delay value.
Based on figure below, transmission of data between PC 1 (192.168.10.200)
to PC 3 (192.168.11.2) provide delay value of 13.60 milliseconds while from
PC 1 (192.168.10.200) to PC 6 (192.168.14.2) provide delay values of 22.42
milliseconds for load balancing scenario.

56
 However, from results collected, the delay values for both scenarios is
considered as excellent as the value is below 150 milliseconds. If the delay
values exceed 400 milliseconds based on ITU G.114 standard, the voice
quality is considered as worst. (Zheng, Zhang, & Xu, 2001)

Figure 4.9: Network delay.

4.5 Voice Quality Result

As this study is related to VoIP network system, monitoring the quality of VoIP
is important to provide maintain the quality standard and enhance service
quality for users’ experience. By using Omnipeek packet analyzer, 4
parameters are measured in this study; jitter, packet loss, MOS and R-Factor.
Voice coded used in this study is G.711 a-law.

4.5.1 Jitter

Jitter is one of common issue in VoIP. Jitter caused by store-and-forward

queueing congestion in network devices; switch and routers lead to bigger
packet spacing. The more devices used to transmit the data or the more hops
for the voice packet to travel to its destination the higher the jitter value
(Wildpackets.Inc).

57
 From figure 4.10 below, the graph indicates jitter value is increase from
0.4 milliseconds for the call session to PC 3 (192.168.11.2) to 1.32
milliseconds for the call session to PC 6 (192.168.14.2) for load balancing
scenario. While for non-load balancing scenario (HSRP), jitter value from 0.23
milliseconds to 4.3 milliseconds as the number of hops increases. This is
because to transmit voice traffic to PC 6, the traffic went through 5 routers
originating from local network (192.168.10.0/24) to 192.168.14.0/24 network.

Figure 4.10: Jitter.

4.5.2 Packet Loss

Figure 4.11 below shows plotted graph of packet loss on two scenarios. Packet
loss is measured in percentage. Packet loss is caused by dropped packets
during the transmission due to network congestion or device failure and jitter.
From graph below, the percentage of packet loss in HSRP is slightly higher
than GLBP with the value of 0.017% and 0.014%. As the higher number of
hops for the packet to travel the percentage value of 0.065% for HSRP and
0.0.35% for GLBP. These values are considered as excellent for voice quality
as the value is below than 1% as this study is conducted on simulation software
therefore dropped of packets can barely be seen in this network. If the study
is conducted on real environment, the value of packet loss can clearly be seen.

58
 Figure 4.11: Packet loss.

4.5.3 Mean of Opinion Score (MOS)

Figure 4.12 shows MOS on two scenarios plotted in a graph. MOS represents
the Quality of Service in VoIP network. As seen in figure below, as the number
of hops for the voice packet to travel, MOS value is becoming lower such as
the value for load balancing scenario is between 4.17 (call session to
192.168.11/0 network) to 4.12 (call session to 192.168.14.0/24 network). While
for non-load balancing scenario, the value is slightly lower than load balancing
configuration is implemented. Based on the collected results, MOS is
considered as good as the value between 4 – 5.

Figure 4.12: Mean of opinion score.

59
4.5.4 R-Factor

R-Factor is a derived value from parameters such as packet loss, jitter and
latency. The scores range between 50 to 120 which is from worst to excellent.
As seen in figure 4.13 below, R-Factor score for GLBP is range between 91
until 93 and is the call session is considered as excellent due to load balancing
implemented on the network. While for non-load balancing scenario (HSRP) it
can be seen the score drop to 85 during call session from PC 1
(192.168.10.200) to PC 6 (192.168.14.2) however still considered as
acceptable value and the quality of the call is fair.

Figure 4.13: R-Factor.

4.6 Chapter Summary

Based on the results recorded and the discussion for each of the results it can
be concluded that the significance of load balancing implementation in VoIP
network to maintain the quality of voice communication thus increased users’
satisfaction and experience. However, since this project is conducted on
simulation environment, the results is different from real environment due to
clean environment where there are less interferences during the simulation.
This section explained all of the collected data and related them to theory from
academic sources.

60
 CHAPTER 5

CONCLUSION

5.1 Introduction

This section discussed the summary of findings related to this research and
conclusion of this report. This section also discussed future recommendations
to improve this study research.

5.2 Conclusion

Regards to the completion of this study, the objectives are successfully

achieved. The main purpose of this study is to review the impact of load
balancing implementation in VoIP network. Nowadays, VoIP network is rapidly
deployed due its ability to deliver the network and media content over internet.
This study compared the implementation of redundancy protocol and load
balancing protocol. Load balancing plays an essential role as it enable
multidevice to share or distributes the load among them in order to transmit to
the destination instead on forwarding traffic on a single dedicated router while
redundancy provides the failover or backup system in case of any devices or
links failure in the network. GLBP proven to be the solution for VoIP network
implementation as the protocol itself provides load balancing, IP redundancy
and network failover. The presents of load balancing in VoIP network will boost
the productivity and performance of the network itself.

61
5.3 Recommendation

This research is conducted in simulation environment by using GNS3 network

simulator. The results provided is different than the results of real environment
implementation. This is because simulation environment having less to no
interferences thus the results provides on this study are based on improving
the theory on advantages of load balancing configuration. The implementation
of real environment can be done on UniKL BMI.

Furthermore, various of load balancing algorithms and methods can be

implemented and analyzed for their impacts on VoIP network such as weighted
algorithms, server load balancing and many more. The comparison of the
network and voice quality for these algorithms and methods will provide clear
decision for better VoIP system implementation.

Besides, this study can be improved by increase the size of network

topology such as numbers of VoIP clients, routers, switches and network links
to make it seem like actual scenario is being analyzed providing clear results.
Since implementation of VoIP can be quite tricky and requires spending more
time to inspect and analyzed for each parameters value.

62
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 APPENDIX A

(PRESENTATION POSTER)

69
 APPENDIX B

(TECHNICAL JOURNAL)

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71
72
73
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75
 APPENDIX C

(FYP 1 GANTT CHART)

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(FYP 2 GANTT CHART)

77
 APPENDIX D

(ROUTER CONFIGURATION)

ROUTER R5

R5#sh run
Building configuration...

Current configuration : 1390 bytes

!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname R5
!
boot-start-marker
boot-end-marker
!
!
interface FastEthernet0/0
ip address 192.168.10.2 255.255.255.0
duplex auto
speed auto
glbp 5 ip 192.168.10.1
glbp 5 priority 105
glbp 5 preempt
glbp 5 weighting 1
!
interface Serial0/0
ip address 10.1.1.2 255.255.255.252
clock rate 2000000
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/1
no ip address
shutdown
clock rate 2000000
!
router eigrp 10
passive-interface FastEthernet0/0
network 10.1.1.0 0.0.0.3
network 192.168.10.0
auto-summary
!

78
ip forward-protocol nd
!
!
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
!
!
end

ROUTER R6

R6#sh run
Building configuration...

Current configuration : 1352 bytes

!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname R6
!
boot-start-marker
boot-end-marker
!
!
interface FastEthernet0/0
ip address 192.168.10.3 255.255.255.0
duplex auto
speed auto
glbp 5 ip 192.168.10.1
glbp 5 weighting 1
!
interface Serial0/0
ip address 10.1.2.2 255.255.255.252
clock rate 128000
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/1
no ip address

79
 shutdown
clock rate 2000000
!
router eigrp 10
passive-interface FastEthernet0/0
network 10.1.2.0 0.0.0.3
network 192.168.10.0
auto-summary
!
ip forward-protocol nd
!
!
!
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
!
!
end

80