DESIGN AND PROTOTYPE IMPLEMENTATION OF VOICE OVER DATA

NETWORKS FOR UNIFIED COMMUNICATION (UC)

(A CASE STUDY OF RADFORD UNIVERSITY COLLEGE)

BY

MICHAEL AYISI.

(B080311004)

i
 Project Report

On

DESIGN AND PROTOTYPE IMPLEMENTATION OF VOICE OVER DATA

NETWORKS FOR UNIFIED COMMUNICATION (UC)

Submitted by

MICHAEL AYISI.

In partial fulfilment of the requirements for the Degree of

Bachelor of Science

In

INFORMATION COMMUNICATION TECHNOLOGY

(ICT)

RADFORD UNIVERSITY COLLEGE

Affiliated to

Kwame Nkrumah University of Science and Technology (KNUST).

November 2015

©2015

ii
DECLARATION

I hereby declare that the work submitted in this research is the results of my own designed

work and investigation, except where otherwise stated. It has not already been accepted for

any degree, and is not being currently submitted for any degree.

However, all aspects of this study have been discussed with and approved by my supervisor.

Michael Ayisi Signature ______________ Date……………

(B080311004)

Certified by

Mr. Collins Collinson Abgesi Signature _______________ Date …………..

Supervisor

Certified by

Bishop Emmanuel Anim Yeboah Signature _______________ Date ………….

Head of Department

i
DEDICATION

This work is dedicated to Almighty God for the knowledge, strength and ability he imparted

to me throughout this project work.

To my lovely Mom and Dad (Mr. and Mrs Ayisi) and my two sisters for their care throughout

my study. And to the Lecturers at the department I say thank you for your support. May the

almighty God richly bless you all.

ii
ACKNOWLEDGEMENT

I would like to take this opportunity to thank the Almighty God for his faithfulness. I would

as well like to take this chance to express my sincere gratitude to my well respected project

supervisor in the person of Mr. Collinson Agbesi for his meticulous and expert guidance,

constructive criticism, patient hearing and benevolent behaviour throughout my ordeal of the

present research.

I shall remain grateful to him for his cordial, cooperative attitude, wise and knowledgeable

counsel that acted as an impetus in the successful completion of my project work.

I will also take this opportunity to express a deep sense of gratitude to our project coordinator

Mr. Isaac Afoakwa for his cordial support, valuable suggestions and guidance. I extend my

sincere thanks to our respected Head of department Bishop Emmanuel Anim Yeboah, for

giving me guidance and inspiration during my study in the department.

It however, not possible for me to forget the kind of help provided by the faculty members,

Madam Sally Ann, Mr. Kofi Sarpong, Mr.Frank K. Banaseka, Mr. Stephen Dotse and Mr.

Emmanuel Sam. I also convey my thanks to IT Department Staff for extending their support

in my research on this project.

Finally, I am indebted to all those who in various ways helped to make this research work a

success.

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Table of Contents
DECLARATION ..........................................................................................................................................i
DEDICATION ............................................................................................................................................ ii
ACKNOWLEDGEMENT ............................................................................................................................ iii
Table of Contents ................................................................................................................................... iv
List of Tables ........................................................................................................................................ viii
ABSTRACT............................................................................................................................................... ix
1 CHAPTER ONE ..................................................................................................... 1
1.1 Introduction .............................................................................................................1
1.2 Background of Study ................................................................................................1
1.3 Problem Statement ..................................................................................................3
1.4 Study Objectives ......................................................................................................4
1.4.1 Global Objectives ..............................................................................................4
1.4.2 Specific Objectives ............................................................................................4
1.5 Significance of Study ................................................................................................5
1.6 Scope and Limitation of Study .................................................................................7
1.6.1 Scope ................................................................................................................7
1.6.2 Limitations ........................................................................................................7
1.7 Research Methodology ............................................................................................8
1.8 Organization of Study ..............................................................................................9
2 CHAPTER TWO .................................................................................................. 10
2.1 Literature Review ...................................................................................................10
2.2 History of Telephony ..............................................................................................11
2.3 Data Networks .......................................................................................................12
2.4 Voice over Internet Protocol (VoIP) .......................................................................12
2.5 VoIP components ...................................................................................................14
2.5.1 VoIP Protocols.................................................................................................16
2.5.2 Real-Time Protocol (RTP) ................................................................................17
2.5.3 Session Initiation Protocol (SIP) ......................................................................18
2.6 Review of existing system(s): .................................................................................23
2.6.1 3CX adoption by university of North Carolina ................................................24
2.6.2 Problems with this system..............................................................................25
2.7 Existing System ......................................................................................................26
2.7.1 Components of Existing System .....................................................................28
2.7.2 Process of the System.....................................................................................29
2.7.3 Problems of the Existing System ....................................................................30
3 CHAPTER THREE ................................................................................................ 31

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3.1 Proposed System ................................................................................................... 31
3.2 Introduction to Hardware Implementation .......................................................... 32
3.3 Development Tools and technologies used .......................................................... 32
3.3.1 Hardware ........................................................................................................ 32
3.3.2 Software ......................................................................................................... 35
3.4 Quality of Services (QoS) ....................................................................................... 35
3.4.1 QoS requirements for Voice application ........................................................ 37
3.5 Preparation Phase ................................................................................................. 37
3.5.1 Bandwidth ...................................................................................................... 37
3.5.2 Network architecture ..................................................................................... 38
3.5.3 Soft switch ...................................................................................................... 38
3.5.4 Connection ..................................................................................................... 42
3.5.5 Soft phones .................................................................................................... 42
3.5.6 3CX Phone Soft Phone .................................................................................... 43
3.5.7 Zoiper Soft Phone ........................................................................................... 44
3.5.8 Grand Stream HT 502 ATA ............................................................................. 45
4 CHAPTER FOUR................................................................................................. 46
4.1 Implementation and testing of the proposed system. .......................................... 46
4.1.1 Download and Installation of Ubuntu 14.04 LTS Server. ............................... 47
4.1.2 Download and Installation of Certified Asterisk 11.7.1 LTS (long term support).
49
4.2 Integration of Hardware ........................................................................................ 51
4.3 Implementing the features of Asterisk Server ...................................................... 52
4.3.1 Voice Call ........................................................................................................ 52
4.3.2 Voice mail ....................................................................................................... 54
4.4 Configuration ......................................................................................................... 55
4.4.1 Configuring LAN-VoIP user ............................................................................. 55
4.4.2 Configuration of E1200 Router ...................................................................... 56
4.4.3 Configuration of 3CX ...................................................................................... 57
4.4.5 Configuration of IP Telephone ....................................................................... 59
4.4.6 IVR (Interactive Voice Response) ................................................................... 61
4.5 Testing of RUC VoIP System .................................................................................. 62
4.6 Verification of VoIP metrics ................................................................................... 64
4.6.1 Security Issues ................................................................................................ 66
4.6.2 Change user main password and root default password .............................. 66
4.6.3 Avoiding SIP authentication requests from all IP addresses .......................... 67
4.6.4 Wireless phones require advanced wireless security .................................... 67
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4.6.5 Physical security .............................................................................................67
5 CHAPTER FIVE ................................................................................................... 68
Conclusion and Recommendations ................................................................................. 68
5.1 Conclusion ..............................................................................................................68
5.1.1 Server Capacity ...............................................................................................69
5.1.2 User Capacity ..................................................................................................69
5.2 Future Recommendations .....................................................................................69
REFERENCES ................................................................................................................... 71
APPENDIX....................................................................................................................... 74

vi
List of Figures

Figure 1.1 Cisco PPDIOO Life Cycle ............................................................................................8

Figure 2.1 Evolution of Telephones .........................................................................................11

Figure 2.2 VoIP Components ...................................................................................................15

Figure 2.3 VoIP Protocol Architecture .....................................................................................16

Figure 2.4 SIP Distribution Architecture (Butt Muhammad Faisal Nazir, 2006). .....................18

Figure 2.5 3CX Logo..................................................................................................................23

Figure 2.6 Radford Existing Data Network ...............................................................................27

Figure 2.7 Scenario...................................................................................................................29

Figure 3.1 Proposed System Network architecture .................................................................31

Figure 3.2 Network Architecture .............................................................................................38

Figure 3.3 Asterisk Architecture ..............................................................................................41

Figure 3.4 LAN-VoIP Network ..................................................................................................42

Figure 3.5 3CX Phone and 3CX Logo respectively. ...................................................................43

Figure 3.6 represents the Zoiper Dial pad interface and Logo respectively. ...........................44

Figure 3.7 Grand Stream HT 502..............................................................................................45

Figure 4.1 Virtual Box Version 4.3.12 r93733, 2014 ................................................................46

Figure 4.2 Creating a Virtual Machine for Ubuntu 64bit Server using Virtual Box. .................47

Figure 4.3 Ubuntu install Options ............................................................................................47

Figure 4.4 Software Selection ..................................................................................................48

Figure 4.5 Installation completed screen ................................................................................48

Figure 4.6 Installation completed screen ................................................................................49

Figure 4.7 Server login Details .................................................................................................49

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Figure 4.8 Integrating HT502 ...................................................................................................51

Figure 4.9 Flow Chart Voicemail ..............................................................................................54

Figure 4.10 Configuration of Lan- VoIP user ............................................................................55

Figure 4.11 GUI of Cisco E1200 Router ....................................................................................56

Figure 4.12 Configuration of 3CX Soft phone ..........................................................................57

Figure 4.13 Create a new account and account Type..............................................................58

Figure 4.14 Account credentials ..............................................................................................58

Figure 4.15 Configuring Polycom IP Phone ..............................................................................59

Figure 4.16 Configuration of IP phone .....................................................................................60

Figure 4.17 Sample IVR scenario 1 ...........................................................................................61

Figure 4.18 Sample RUC network Architecture .......................................................................63

Figure 4.19 Incoming Call from Accounts Office <2001> to Registry <2000> .........................63

Figure 4.20 Call established between the Registry Department and the Account Office.......64

Figure 4.21 Wireshark screenshot when RTP packets are filtered ..........................................65

Figure 4.22 Delay and jitter Vs packet sequence when only voice is transmitted ..................66

List of Tables
Table 2.1 VoIP Advantages and Disadvantages ..................................................................13

Table 2.2 3CX Products and Pricing ....................................................................................25

Table 3.1 Hardware ...............................................................................................................33

Table 4.1 Credentials .............................................................................................................53

viii
ABSTRACT

This study entails the simulation and implementation of voice over a data network, a telephony

system using an IP PBX solution. A technology called voice over Internet Protocol (VoIP), or

Internet telephony means that voice is carried over an IP network. Voice, which is an analogue

signal, is converted to digital data, which is then disassembled and transmitted through a network

only to be reconverted back to an analogue signal on the other end using a Linux based IP-PBX

solution called Asterisk. This service can be properly managed and deployed over a network with

less stress and expenses. The IP PBX main server also has integrated in it other communication

services such as Voice mails, IVR’s, all embedded in the IP PBX SYSTEM.

This technology promises an evolutionary leap beyond the standard telephone service we

have been accustomed to, as well as a host of benefits. The new technology transmits voice

signals the same way email is sent, using the Internet’s data-transfer protocols to break

conversations into digital packets that can be sent on lower-cost, more efficient “packet-

switched” networks. This project was able to address the persistent communication problem

which existed in the departments by allowing users to communicate with the services the

solution provided with less stress and comfort.

Target market includes: Corporate organizations, Universities, Health care, Airports, Hotels,

Banks etc. This project is economic, cost effective, gives full control to the administrator and

provides mobility, feasible, Peer-to-Peer phone calls. The contents of IP PBX System,

supplemented by a good number of necessary and descriptive drawings which makes this

project report very easy to understand

ix
 1 CHAPTER ONE

1.1 Introduction

Voice over Data Networks is to transmit voice over Packet Switched Data Networks by

converting it into packets while keeping reliability and voice quality as in circuit switched

telephone networks, and gaining cost savings (L. Sun, 2004). In fact, the convergence of

voice and data networks is rapidly gaining grounds across the globe. 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)

1.2 Background of Study

In today’s converged networks and telephony industry, the catchphrase is IP (Internet

Protocol) telephony. Many see it as a major evolutionary step in telecommunications. 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).

1
 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. (Lawrence Harte, 2003).

I want to focus on the impact Internet Protocol (IP) telephony solution will have on the

existing data network at Radford University College. 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.

“Embarking on an initiative to provide telephony and other voice related services over the

campus data network is piloting an open source solution (Asterisk by Degium) that provides

cost-effective high function enhanced media services, such as departmental voice systems,

voice mails, automatic call distribution, and IVRs”. (Deke Kasabian, 2005)

Students and staffs will have access to this solution and use different services such that they can

make voice calls, and use different features like voicemail, call group, queuing, interactive voice

responses and conference calls are features that a configured Asterisk server can offer.

2
This solution will fundamentally transform the way in which students and the university

administration communicate – from decreased carrier costs to increased response times, thus

the benefits of Unified Communications greatly outweigh the investment.

1.3 Problem Statement

Academic institutions like Radford University College are often challenged by the high cost

and lack of flexibility of ordinary telephone systems. Often, there are many communication

costs related to the management and implementations of programs for academic institutions.

Due to the global financial crises, these costs become an additional burden on the already

overstretched budgets.

Radford University College currently has a large campus wide data network, spanning

approximately 6 floors, and comprising of approximately 100 wired network ports, 7 CCTV

Cameras, 15 Switches and in excess of 6 wireless access points.

It’s rather unfortunate with such a detailed network infrastructure; there is no voice network.

The university employs the services of various telephone service providers e.g.: MTN,

Vodafone etc. In fact the issue of making calls internally through these service providers

imposes a high monthly cost. The current system at Radford University leaves little or no

room for customization and offers fewer features.

The modern student is increasingly technology savvy, the Pew Research Centre findings as at

October 2014 states “65% of those between 18 and 24 own a smartphone and 23% have a

tablet. As such, these young people want campus administrators to further use tools like

unified communications (UC) solutions that cater for their digital demands”.

Therefore the aim of this project is to develop a communication technology to interconnect those

hosts that are in the various departments at Radford University. This solution will permit

3
the university’s IT department to deliver free telephone calls, voicemail, ring groups, call

transfer, conferencing, IVR and other telephony services to both students and staffs internally.

1.4 Study Objectives

1.4.1 Global 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.

1.4.2 Specific Objectives

The specific objectives of this project are to achieve the following:


To develop an operational IP telephony solution for Radford University College, based on

 a software implementation of a telephone Public Branch Exchange (PBX) running

 a Linux distribution server Asterisk on the University’s data network.


Installation and
configuration of an operational Linux server based on Asterisk using
 SIP Protocol.


Design a VoIP network to be utilized by the above server in which hosts can call each
 other with softphones, IP phones and also offering features such as voicemail and

Interactive voice Response (IVR).

4
1.5 Significance of Study

The solution contributes significantly to the development and adoption of Unified

Communications solutions. IP telephony could be applied to almost any voice

communications requirement, ranging from a simple inter office intercom to a complex multi-

point teleconferencing environment.

Listed below are the main benefits of using IP telephony:

 
Cost Reduction

The first measure of success for IP telephony is the cost savings for long distance calls. Today

flat rate long-distance pricing is available with the Internet and can result in considerable savings

for both voice and fax. Large organizations with offices around the world save even more on

long-distance calls by using local Internet gateways. IP telephony provides a competitive threat to

providers of traditional telephone services that will clearly stimulate improvements in cost and

function throughout the industry. Parker, M., and D. Van Doren

 
Scalable

Proprietary systems are easy to outgrow. Adding more phone lines or extensions often

requires expensive hardware modules. In some cases you need an entirely new phone system.

Not so with an IP PBX. A standard computer can easily handle a large number of phone lines

and extension, just add more phones to your network to expand! This solution is integrated

different communication devices like soft phone, IP phones and even hard phones.

 
Network Efficiency

The sharing of equipment and operations costs across both data and voice users can also improve

network efficiency since a packet switched IP network can handle more calls with the same

transmission infrastructure than the PSTN can with its circuit switched TDM approach.

5
  
Integration

Universal use of the IP protocols for both data and voice applications holds out the promise

of reduced complexity and more flexibility. This provides an opportunity to share facilities

such as directory services and security services, and eliminate points of failure.

 
Advanced Applications

In addition to basic telephony and fax, the long-term benefits are expected to be derived from

multimedia and point of-service applications such as directory services that enable conference

calls to be set up from Web based directories, and wireless unified messaging, which will let

users retrieve their voice and e-mail messages via their cellular phones. Combining voice and

data features into new applications will also provide the greatest returns over the longer term.

 
Better customer service & productivity!

With an IP PBX you can deliver better customer service and better productivity. Since the IP

telephone system is now computer-based, you can integrate phone functions with business

applications. For example, bring up the customer record of the caller automatically when you

receive his/her call, dramatically improving customer service and cutting cost by reducing time.

 
Eliminate vendor lock in

IP PBXs are based on the open SIP standard. You can mix and match any SIP hardware or

software phone with any SIP-based IP PBX or PSTN Gateway. In contrast, a proprietary

phone system often requires proprietary phones to use advanced features, and proprietary

extension modules to add features.

6
1.6 Scope and Limitation of Study

1.6.1 Scope

The research highlights converged networks with Radford University’s existing data network.

More specifically the study would detail an

 
 Installation and configuration of an operational Linux server based on Asterisk.


Design of a VoIP network to be utilized by the above server in which hosts

(users) in different departments of the University can call each other internally

with their audio enabled personal computers, softphones, IP phones and with


traditional phones.

 
Design of a system offering voicemail and Interactive voice Response (IVR).

1.6.2 Limitations

There are still considerable technical challenges and limitation in implementing Voice over

Wi-Fi, including concerns about security, battery life in wireless handsets and call quality.

Wireless networks allocate bandwidth according to which devices are nearest to the WLAN

access points, which can cause problems for voice call quality although some suppliers are

developing systems that allocate bandwidth equally from the access points and can prioritize

voice traffic using Quality of Service (QoS).

7
1.7 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 required for a

network (Stephen J. Occhiogrossp, 2011). 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.

Figure 1.1 Cisco PPDIOO Life Cycle

8
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.

1.8 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.

9
 2 CHAPTER TWO

2.1 Literature Review

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.

The type of equipment used in making and receiving a phone call classifies IP Telephony

usage set-ups or scenarios. The call can be initiated or terminated either by a PSTN (Public

Switched Telephone Network) device or a computer (PC or laptop) on each side of the call.

(Nick Galea, 2009).

10
2.2 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 2.1 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. It is commonly used for residential use.
 POTS is an analogue system and is controlled by electrical loops.


ISDN (Integrated Services Digital Network): This is a faster and more feature-filled

connection (also more expensive). This gained some popularity within small to
11
 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

(Barrie Dempster, 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).

2.3 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) which covers a country for example or Local Area Networks (LANs) which connect

workstations in a building for example.

An IP network is based on the "best effort" principle which means that the network makes no

guarantees about packet loss rates, delays and jitter. For voice traffic, the perceived voice

quality will suffer from these impairments (e.g. loss, jitter and delay). I will focus on

transferring voice over the IP network which is abbreviated, VoIP.

2.4 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

12
(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.

Table 2.1 VoIP Advantages and Disadvantages

Advantages Disadvantages
 
Low cost Users cannot make calls during power

Flexibility. outages.
 
Provides voice mail and call forwarding. Connection limitation to emergency

Easy to implement and install services.
 
Free services gained usually when Depends on Internet connection condition.

connecting from PC to PC (G. Samrat, B. IP network that does not guarantee Quality
Sudeept, 2008) of Service for voice communication (J.M.

Network Capacity utilization Lozano-Gendreau, A.Z. Halabi, M.

Users can make VoIP calls from anywhere Choueiri and V. Besong, 2006).
for long distance or international calls.

Integration with other available services
over the Internet.

13
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.

2.5 VoIP components

VoIP consists of three essential components: CODEC (Coder/Decoder), packetizer and

playout 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 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 2.2.

Figure 2.2 VoIP Components

Besides, there are signalling protocols of VoIP namely Session Initiation Protocol (SIP) and
H.323. These signalling 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 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.

15
 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 2.3 VoIP Protocol Architecture

2.5.1 VoIP Protocols

Voice over IP (VoIP) is the transmission of voice over network using the Internet Protocol.

Here, I will introduce briefly be outlined the various VoIP Protocols that aided my project.

The Protocols that provide basic transport (RTP), call-setup signalling (H.323, SIP was used)

and QoS feedback (RTCP) (Clifford Stoll, 2009).

16
2.5.2 Real-Time Protocol (RTP)

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video

over IP networks. RTP is used extensively in communication and entertainment systems that

involve streaming media, such as telephony, video teleconference applications, television

services and web-based push-to-talk features. This protocol is a user session protocol which

relies on User Datagram Protocol (UDP), hence, make use of the checksums and

multiplexing services to allow data handling for programs in real time unicast or multicast

transmissions. RTP does not in itself guarantee real-time delivering of multimedia data. The

tool that RTP uses to achieve real time transmissions is the Real Time Control Protocol

(RTCP), which provides a feedback about some control information. With this, it is possible

to monitor the quality of the transmission and also possible to diagnose network problems.

RTP consists of four main fields; the detail of these fields is described below (Butt

Muhammad Faisal Nazir, 2006).

RTP Payload type Indicates the specific media encoding and which codec to use. The codec

conveys the type of the data (such as voice, audio or video) and how it is encoded. It can be

changed if it has to adapt the variation in bandwidth, frame indication, which marks the

beginning and end of each frame.

Sequence number helps the receiving end to reassemble the data and detect lost, out-of-

order and duplicate packets.

Time Stamp: It is used to reconstruct the timing of the original audio and video. It also helps

the receiving side determine variations in packet arrival times, known as jitter. It is the time

17
stamp that brings real value to RTP. At the receiving node each packet is compared with a

time stamp to make RTP transmission possible.

Source id: It is used to distinguish among multiple, incoming streams by the software at the

receiving side.

In this project work, RTP is used as voice streaming protocol to send real-time traffic.

To place a call on the data network, VoIP involves two types of protocol; call setup protocols

and voice streaming protocols. Call setup protocols are available to serve as the VoIP

signalling protocol, SIP, H.323 and IAX (Inter Asterisk exchange) are most common choices.

The next paragraphs focus on these signalling protocols.

2.5.3 Session Initiation Protocol (SIP)

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, modifying 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 2.4 SIP Distribution Architecture (Butt Muhammad Faisal Nazir, 2006).
18
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 port numbers 5060 and/or 5061 to connect to SIP servers and other SIP

endpoints. Port 5060 is commonly used for non-encrypted signalling 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.

2.5.3.1 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.

2.5.3.2 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,

19
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 redirect 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.

20
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).

2.5.3.3 SIP Addresses

The SIP network has the address attribute: SIP URL (SIP Uniform Resource Locator) to be easily

recognizable. SIP URLs used in SIP networks 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.

2.5.3.4 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. Appendix C-1 presents details on these methods;

21
2.5.3.5 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. These 6 classes of responses are

explained in Appendix C-2.

2.5.3.6 H.323 protocol

H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-

T) that defines the protocols to provide audio-visual communication sessions on any packet

network. The H.323 standard addresses call signaling and control, multimedia transport and

control, and bandwidth control for point-to-point and multi-point conferences.

It is widely implemented by voice and videoconferencing equipment manufacturers, is used

within various Internet real-time applications and is widely deployed worldwide by service

providers and enterprises for both voice and video services over IP networks.

Within the context of H.323, an IP-based PBX might be a gatekeeper or other call control

element which provides service to telephones or videophones. Such a device may provide or

facilitate both basic services and supplementary services, such as call transfer, park, pick-up,

and hold (Jim Van Meggelen, Leif Madsen, and Jared Smith, 2007).

N.B: SIP has pretty much dethroned the once-mighty H.323 as the VoIP protocol of choice

certainly at the endpoints of the network. The premise of SIP is that each end of a connection is a

peer; the protocol negotiates capabilities between them. What makes SIP compelling is that

22
it is a relatively simple protocol, with syntax similar to that of other familiar protocols such as

HTTP and SMTP (Simple Mail Transfer Protocol) [12].

2.6 Review of existing system(s):

Figure 2.5 3CX Logo

Developer(s) 3CX

Stable release Version 12 – Build 32127 /September 4, 2013

Written in C++

Operating system Microsoft Windows

Type Voice over Internet Protocol

License Proprietary

Website: http://www.3CX.com

3CX Phone System is software based private branch exchange (PBX) based on the SIP

(Session Initiation Protocol) standard. It enables extensions to make calls via the public

switched telephone network (PSTN) or via Voice over Internet Protocol (VoIP) services.

3CX Phone System for Windows is an IP business phone system that supports standard SIP

soft/hard phones, VoIP services and traditional PSTN phone lines.

3CX Phone System for Windows was developed by 3CX and first published as a free IP PBX

product in 2006. The product was intended to provide a VoIP solution for use in a Microsoft

Windows environment. The first commercial edition of the product was launched in 2007.

Reviews of the product have noted its easy configuration, management, and hardware

23
compatibility. Smith on VoIP commented in a blog post about 3CX that it was very easy to

use, but did not have all the features its competitors provide.

2.6.1 3CX adoption by university of North Carolina

In January 2007 Craig Hyatt, Information Technology Director for Campus Services at the

University of North Carolina (http://www.csit.unc.edu), decided that it was time to get rid of

a department’s traditional PBX which offered limited functionality and was very costly. He

opted for 3CX Phone System for Windows and as a result, monthly phone bills have been

reduced by a whopping 81%, and staffs has become more mobile and productive.

“Campus Services Information Technology is in charge of providing superior, innovative and

responsive technology management to departments within the Campus Services Division to

support the university’s mission of teaching, research and public service. This means that when

we decide to adopt a new technology in our own premises, it has to be the best”, says Hyatt.

Telephone bills at UNC’s Campus Services Information Technology have been dramatically

reduced since the adoption of 3CX, “With our old traditional PBX the cost per handset was

$48 per month, with a total monthly bill mounting to almost $400 for eight handsets. These

days, the share of the phone bill for eight handsets – out of the 25 we have installed – is only

$75 per month”, highlights Hyatt. This is an astonishing 81% decrease in cost!

The University of North Carolina at Chapel Hill Campus Services IT is using the Pro Edition

of 3CX Phone System for Windows and a total of 25 extensions with Grand stream GXP-

2000 phones. They are planning to add more extensions in the near future.

“We are delighted to see that the Information Technology department of an established

educational institution such as the University of North Carolina at Chapel Hill chose 3CX

Phone System for Windows to reap the benefits of IP Telephony; including considerable cost
24
savings in phone bills and increase in staff productivity and mobility through the use of

unified messaging”, says Nick Galea, CEO at 3CX.

2.6.2 Problems with this system

There is no doubt 3CX helps improve business processes, reduce costs and increase customer

satisfaction, but there likely isn’t one VoIP system without its ups and downs. I would like to

elaborate the drawbacks the above solution has.

Compatibility: As a Windows-based software, 3CX works great. Unless you’re not using

Windows, in which case you’ll likely have to rethink your 3CX strategy for the future.

Proprietary: As a developer, I am limited as to what customization I can make to 3CX. The

number of simultaneous calls only limits 3CX Phone Systems. 3CX licenses are available in

simultaneous call counts ranging from 4 to 1024 simultaneous calls.

Pricing: 3CX Phone System is licensed based on the number of simultaneous calls (internal

&external) that a company makes.

All editions support a limited number of extensions.

Table 2.2 3CX Products and Pricing

PACKAGES SIMULTANEOUS EXTENSIONS PRICE MAINTENANCE

CALLS COST
1 3CX Mini 1-4 10 $495 $99
Edition
2 3CX Small 2-8 30 $795 $199
Business
Pro Edition
3 3CX 3-10 3-30 $2495 $299
Enterprise
Edition
25
2.7 Existing System

Radford University College currently has a large campus wide data network with internet,

spanning approximately 6 floors, and comprising of approximately 100 wired network ports,

7 CCTV Cameras, 15 Switches and in excess of 5 wireless access points.

It’s rather unfortunate with such a detailed network infrastructure; there is no voice network

as shown in Figure 2.6.

The university employs the services of various mobile/telephone service providers. A single

PSTN line (landline) from Vodafone as well as various mobile telephony services from other

Telco’s allows customers/students to reach the university.

Internally there isn’t any PBX system the links the reception to the various

departments/faculties/lecture rooms or entities that make up the university.

Departments and students are unable to communicate with each other without one physically

being present at one’s office, however even if communication exists it’s achieved via a

mobile phone which cost money which could have been avoided.

For example; During Enquires a front desk personnel would need to reach an

individual/department being sought for in person, thus reducing productivity and allowing

longer queues at the reception. In fact the issue of making calls through Mobile providers

imposes a high cost.

The current phone system at Radford University leaves little or no room for customization

and thus offers less functionality.

26
Figure 2.6 Radford Existing Data Network

27
2.7.1 Components of Existing System

Currently, RUC is a large university offering 7 undergraduate programs and certificate

programs with more than 2,000+ students. For more detail see www.radforduc.edu.gh/

RUC Offices and Faculties

Offices include:
 
 Reception
 
 Conference room

 
 Library

 
 Faculties

 
 Accounts

 
 Registry

 
President Office

RUC has 7 faculties:

 
 Information communication technology
 
 Business Administration

 
 Fashion Design

 
 Graphic Design

 
 Applied science

 
School of health sciences

The VoIP server will be situated (located) at RUC ICT server room. End users use IP phones,

analogue phones with ATAs and finally installation of softphones on different computers

(desktop and laptops) will be performed in order to make call by using computers. Linux and

Windows users will enjoy telephony features offered by this system.

28
2.7.2 Process of the System

The current telephony system at Radford University allows for 80% direct (verbal) human

communication between students, lecturers, continuing students and administration as well as

potential customers.

This delays information transmission that increases waiting time, thus producing longer

queues at the university’s front desk office.

Figure 2.7 Scenario

As shown in figure 2.7, enquires are done through phone calls to the front desk of the

university. One can also visit the reception to make enquiries.

Most often than not, customers are given verbal directions by the receptionist as to where to

find an office/department/individual that would best avail them with information they seek.

29
2.7.3 Problems of the Existing System

Some of the problems being faced by Radford University telephony system are as follows:

Heavy work load for the front desk personnel, which leads to inefficiency.

University public/mobile lines are always busy with high response rates leaving

customers dissatisfied.

Longer queues and overcrowded offices with new students, continuing students and

customers all seeking information.

Inability to transfer calls to Departments, faculties, teaching and non-teaching staffs,

students (where applicable).

After verbal directions given, there is no assurance that whoever is being sought for is

at his/her office at any particular point in time.

Information inaccuracy.

Huge communication gap between all entities of the University, be it students to

students, lectures, faculty, administration and vice versa.

Huge costs on making calls internally.

30
 3 CHAPTER THREE

3.1 Proposed System

The proposed solution will be implemented on Radford University’s existing campus network

to provide free voice calls between the various entities (i.e. student, staff, faculties,

departments etc.) within the university. It will also allow them use of features like Voicemail

and IVR (Interactive Voice Response).The service is secure and allows users to request for

extensions to place calls. Additionally, all the calls are placed through the Linux based

Asterisk PBX (Private Branch Exchange) which is in fact the core kernel. All calls are

encrypted thereby prevents hackers to intercept an ongoing phone calls.

Figure 3.1 Proposed System Network architecture

The model is successful in carrying out voice calls on IOS/Android supported handhelds

connected with the wireless network and PC’s connected with both wired LAN and wireless

LAN. Every entity (i.e. user/office/department/faculty etc.) is provided with a unique extension

31
ID upon request and that will be used to connect within the university`s existing data

network. It is possible to use RUC existing telephone systems and methods of calling, even if

the host to call is not directly connected to the network, hence necessitating the use of

gateways to connect a voice over IP call to the public telephone networks. The different types

of communication devices such as smartphones, IP telephone, Laptops, Desktops, and hard-

phones can be connected to this service.

3.2 Introduction to Hardware Implementation

Implementation is the one of the important part of this thesis. I will discuss about the

implementation of Asterisk and also throw more lights on the hardware used in the

configuration of all software. We will describe the features/components of the project. To

configure eth0 or Ethernet card that has been installed on the server, and then select enable

IP4 support, and finally enter the IP Address and IP Gateway. After installation of server and

soft phone now we are going to integrate ATA with our IPPBX server. Configuring all VoIP

users through the Asterisk server whether it is IP telephone or Softphone by creating SIP

account for them.

3.3 Development Tools and technologies used

There is the list of equipment’s listed here used for the simulation to a final completion of

this project. There is combination of software, hardware and the open sources libraries.

3.3.1 Hardware

The different hardware used in the system can be seen in Table 3.1 the table contains the

specifications and brief description of the tools used in this project.

32
Table 3.1 Hardware

NO TOOLS SPECIFICATION

GRANDSTREAM GXP1628 132 x 48 backlit LCD display

2 dual-color line keys (with 2 SIP
accounts and up to 2 call
1 appearances), 3 XML
programmable context-sensitive
soft keys, 8 BLF keys, 3-way
conference, multi-language support
Dual-switched Gigabit network
ports, integrated PoE, HD wideband
audio, full-duplex hands-free
speakerphone with advanced
acoustic echo cancellation.

2 1 FXS telephone port (RJ11), 1

GRANDSTREAM HT502
FXO PSTN line port (RJ11) with
power-outage life line support,
Up to 2 SIP account profiles, SIP
over TCP/TLS, SRTP
Dual 10/100 Mbps network ports
(RJ45) with integrated high
performance NAT router

ANALOG TELEPHONE
Electronic Handset Volume Control
(6-Step) Flash (for Hook, or use with
special telephone company services, such
3 as
call waiting) 3-Step Ringer Selector
(Off/Low/High) Switchable Tone/Pulse
Settings

33
 PC SERVER IP-PBX
Intel Dual Core E2160
1,8 Ghz
4 Memory 512Mb / HDD 3Gb / Fast
Ethernet Card

WIRELESS ROUTER/AP Cisco Linksys E1200 Wireless-N Router

Frequency Band 2.4 GHz
5 Authentication Method RADIUS
Encryption Algorithm 128-bit WEP, 64-bit
WEP, WPA, WPA2
Routing Protocol RIP, static IP routing

LAPTOP
6 Asus R510c
CPU : Intel Core i5 (3rd Gen) 3337U
Max Turbo Speed: 2.7GHz
OS: Windows 8.1
64-bit Computing: YES
RAM:6GB
HDD:750GB

CISCO CATALYST 2960 8port Layer 2 Switch

24 Port PoE Switch
Cisco 2960 PoE
7 Data Switch
Lan Switch
Cisco Small Business Switch
Cisco managed switch

CAT 5E

9 HANDHELD/SMART MOBILE APPLE IPHONE 5C (IOS)

PHONE DEVICES APPLE IPAD MINI ( IOS)
STUDIO TAB (ANDRIOD)

34
3.3.2 Software

Soft wares used includes;

 
 Ubuntu 14.04 LTS
 
 Asterisk 11.7.1 as soft switch
 
 Soft phones software: 3CXPhone and Zoiper

 
 Putty

 
 Cisco Packet Tracer

 
 Pixlr Editor
 
 Audacity software for voice formatting

 
Wireshark 1.10.1

Wireshark is a network analyzing tool; it captures the packets and displays its details,

Wireshark helps to understand what is going on in the network in real time voice call.

Some of the features of Wireshark are captures live packets, displays a detailed protocol

information for the packet captured, can save the data packets captured, can apply packet

filters, performs stream analysis, and finally it captures information like delay, jitter,

bandwidth, codec etc for packets (Phil Shade,2007).

3.4 Quality of Services (QoS)

QoS is the differentiation between types of traffic and types of services so that the different types

of service and traffic can be treated differently. This way, one type can be favoured over another.

The primary goal of QoS is to provide priority including dedicated bandwidth, controlled jitter,

and latency, and improve loss characteristics. Real time packets such as voice

35
and video packets must be prioritized such that they arrive at their destination on time. Before

showing how to improve the performance of the network to insure QoS, it is important to

describe the five fundamentals network problem for VoIP.


Bandwidth is the fundamental requirement that there be enough space in a network 
 path for all of the packets to get through unimpeded (Not slowed or prevented). This

bandwidth need is symmetric-each end will transmit and receive this amount of traffic

(Quality of Service, n.d).


Packet loss is the amount of packets that does not arrive correctly to their destination.
 This is due to insufficient bandwidth or transmission errors (Quality of Service, n.d).


Latency is the time delay between an event occurring on one site and the remote  end
 seeing it. Latency is introduced both by the encoding/decoding process, and hence

depends on the equipment used, and also by the time it takes packets to traverse the

network. A disruption in the image can cause a bad playing in the destination, but a

disruption in the voice is more important since it makes the transmission not

understandable (Quality of Service, n.d).


Jitter Packets from the source will reach the destination with different delays. A packet's

 delay varies with its position in the queues of the routers along the path between

source and destination and this position can vary unpredictably. This variation in

delay is known as jitter and can seriously affect the quality of streaming audio and/or

video (Quality of Service, n.d).

N.B: Thought this project work we consider that NUR site has enough bandwidth to

accommodate voice and video traffic. While developing, my attention will be focused on how

to minimize packet loss, delay and jitter in VoIP traffic.

36
3.4.1 QoS requirements for Voice application

VoIP deployments require the provisioning of explicit priority servicing for VoIP (bearer

stream) traffic and a guaranteed bandwidth service for Call-Signaling traffic.

Recommendations for voice Loss should be no more than 1 percent, one-way latency (mouth

to ear) should be no more than 150 ms., average one-way jitter should be targeted at less than

30 ms and a range of 21 to 320 kbps of guaranteed priority bandwidth is required per call

(depending on the sampling rate, the VoIP codec, and Layer 2 media overhead).

Voice quality directly is affected by all three QoS quality factors: loss, latency, and jitter

(Tim Szigeti & Christina Hattingh, 2004).

3.5 Preparation Phase

The prepare phase of the Cisco PPDIOO lifecycle approach to network design for a VoIP

solution at Radford University will necessitate the usage of all resources of the University.

3.5.1 Bandwidth

Bandwidth given to PC VoIP server is 1MB. With the number of VoIP users were 6 pieces,

and use codec PCMU. So generally get computations bandwidth used by 6 x 64KB = 384 Kb.

So with 1MB bandwidth is adequate.

37
3.5.2 Network architecture

The network architecture is shown here. The Figure 3.2 showing the interconnection of the

hardware components between different devices.

Figure 3.2 Network Architecture

3.5.3 Soft switch

Soft switch is a central device in a telecommunications network which connects telephone

calls from one phone line to another, across a telecommunication network or the public

Internet, entirely by means of software running on a general-purpose system. Most landline

calls are routed by purpose-built electronic hardware however, soft switches using general

purpose servers and VoIP technology are becoming more popular.

Nowadays, many telecommunications networks make use of combinations of soft switches

and more traditional purpose-built hardware. (Buckley and Sean, 2013)

38
A soft switch is also a VoIP server, providing a soft switch platform with full IP PBX call

features. The most difference from IP PBX is its enormous numbers of users. After thorough

review into platforms I could use in developing IP telephony solution for Radford University

I settled on an open source framework called Asterisk. Answers to these questions informed

my choice.

What is Asterisk?

Asterisk is an open source framework for building communications applications. Asterisk

turns an ordinary computer into a communications server. Asterisk powers IP PBX systems,

VoIP gateways, conference servers and other custom solutions. It is used by small businesses,

large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free

and open source. Asterisk is sponsored by Digium (Asterisk.org, n.d).

Today, there are more than one million Asterisk-based communications systems in use, in

more than 170 countries. Asterisk is used by almost the entire Fortune 1000 list of customers.

Most often deployed by system integrators and developers, Asterisk can become the basis for

a complete business phone system, or used to enhance or extend an existing system, or to

bridge a gap between systems.

Where did Asterisk Come from?

The Asterisk project started in 1999 when Mark Spencer released the initial code under the

GPL open source license. Since that time, it has been enhanced and tested by a global

community of thousands. Today, Asterisk is maintained by the combined efforts of Digium

and the Asterisk community.

39
What Can You Do With Asterisk?

Asterisk is a framework for building multi-protocol, real-time communications applications

and solutions. Asterisk is to real time voice and video applications as what Apache is to web

applications: the underlying platform.

Asterisk abstracts the complexities of communications protocols and technologies, allowing

you to concentrate on creating innovative products and solutions. You can use Asterisk to

build communications applications, things like business phone systems (also known as

PBXs), call distributors, VoIP gateways and conference bridges.

Asterisk includes both low and high-level components that significantly simplify the process

of building these complex applications. See the Asterisk Applications section for more

examples.

What informed my choosing Asterisk for Radford University College?

Asterisk became a preferred platform to develop Rad-VOICE for Radford University because

it is open source, which means you can get under the source code, see how it works and make

any changes or enhancements you like.

Asterisk is flexible and lets you define the solution that truly fits your requirements. Asterisk

is stable, reliable and in production on thousands of systems worldwide. Asterisk is free to

use. But the framework itself is built by developers for developers. If one want to create

applications and solutions with Asterisk you will need a working knowledge of Linux, script

programming, networking and telephony.

40
Figure 3.3 Asterisk Architecture

How Asterisk Works.

• Asterisk is a hybrid TDM and packet voice PBX

• Interfaces any piece of telephony hardware or software to any application

• Prime components: channels and Extensions.conf - the Asterisk dial plan

• Channels can be many different technologies - SIP, IAX, H323 etc.

• extensions.conf is basically a programming language controlling the flow of calls

• Applications do the work - answer a channel, ring a channel, voicemail. (Jonnie-martin,

2007).

41
3.5.4 Connection

The medium used by the Lan-VoIP user is to connect to the server via the Intranet. Users can

connect to Asterisk IP-PBX server via the LAN Intranet wherever they may be on campus.

Connection of Lan-VoIP is shown in Figure 3.4

Figure 3.4 LAN-VoIP Network

3.5.5 Soft phones

I reviewed two soft phones: namely 3cx soft phone and Zoiper soft phone. 3CX was chosen

because it had call forward features which is a major requirement by the university. This can

be installed on any personal computer and compatible with all the operating systems. As

compared to other softphones which provided free call transfer facility, they required advance

registration and less user friendly. One major drawback to using 3CX softphone was its

inability to be installed on an Android/ IOS enabled handheld/smart mobile devices without

having to pay for premium subscription. This led to looking out for another softphone. I

reviewed Zoiper, which actually allows users to install on IOS/android enabled

handheld/smart mobile devices and even on PC. Its only drawback was also user’s inability to

transfer calls if they are not a premium user.

42
3.5.6 3CX Phone Soft Phone

3CXPhone is a popular softphone that can be used on Windows, Mac, Android and iOS

operating systems. As a softphone it can be used to make and receive phone calls from your

PC, laptop, smartphone or tablet.

The advantage of using 3CXPhone as your softphone is that you can boost your company’s

productivity and mobility while at the same time slashing your telecommunications costs.

3CXPhone Benefits

Quick and simple installation

Presence

Substantial savings on telephone bills

Open Standards based next generation softphone

Easy to use, intuitive user interface with dial pad and buttons

MSI installation allows for easy network wide installation

Completely free – saving licensing costs and licensing administration

Figure 3.5 3CX Phone and 3CX Logo respectively.

43
3.5.7 Zoiper Soft Phone

Zoiper is a VoIP softphone that lets you make chat or make voice and video calls with your

friends, family, colleagues and business partners.

Unlike other software like Skype or Viber, it is open and can be used with any VoIP provider

or PBX. Allowing for much more flexibility and cheaper or better quality termination.

• SIP + IAX protocols

• Available codecs are GSM, ulaw, alaw and speex.

• STUN server per account

• Changeable number of lines

• Up to two accounts

• Echo cancellation

• Available on Android, windows, iOS

• Account password encryption

• Outbound DTMF tones sending

Figure 3.6 represents the Zoiper Dial pad interface and Logo respectively.

44
3.5.8 Grand Stream HT 502 ATA

The Grand stream HT502 Analog Telephone Adaptor is an all-in-one VoIP integrated device

designed to be a total solution for networks providing VoIP services. The HT502 VoIP

features and functions are available using a regular analogue telephone. The HT 502 is

powerful VoIP router. The product inclusion of an integrated high performance NAT router

and dual 10/100 Mbps Ethernet WAN and LAN ports enables a shared broadband connection

between multiple Ethernet devices. In addition to being SIP 2.0 standard compliant, the

product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced

telephony features. The image of the Grand Stream HT 502 is given in Figure 3.7

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

Universal Plug-in-Play (UPnP)

2 FXS ports (RJ11) w/up to 2 SIP account profiles

Dual10/100 Mbps ports (RJ45) w/integrated router

HTTP/HTTPS (pending)/Telnet/TFTP

Provisioning IP connectivity for any phone and fax

Web management for easy configuration and installation

Portable and compact for use at home or on the road

Figure 3.7 Grand Stream HT 502

45
 4 CHAPTER FOUR

4.1 Implementation and testing of the proposed system.

The implementation phase is very important for this project. A compromise between the ideal

set-up for this project and what is realisable with the available equipment must be found, but

nevertheless, the ideal project set-up must not be forgotten, to keep it as close to reality as

possible.

For the actual implementation a (Revised) step-by-step plan was chosen. The implementation

stage is divided into two sections, namely the installation and configuration. A server

installation and configuration process is initiated on an Oracle Virtual Machine software

called virtual box.

Figure 4.1 Virtual Box Version 4.3.12 r93733, 2014

46
4.1.1 Download and Installation of Ubuntu 14.04 LTS Server.

A 32/64bit version Linux based Ubuntu 14.04 LTS Server Operating System can be

downloaded at http://www.ubuntu.com/download/server. The process begins with the

installation of Linux based Ubuntu 14.04 LTS server on the Virtual Machine as shown below:

Figure 4.2 Creating a Virtual Machine for Ubuntu 64bit Server using Virtual Box.

1. Select Install Ubuntu Server and follow recommended steps. Fig 3.8

Figure 4.3 Ubuntu install Options

47
 2. You can choose one of these option at the time of install or later. I chose LAMP

(Linux, Apache, MYSQL, PHP) server and also to be able to remotely log on to the

server I checked Open SSH server for install and clicked continue. Fig 3.9

Figure 4.4 Software Selection

3. Click Continue to restart, so as to begin Asterisk installation.

Figure 4.5 Installation completed screen

48
 4. A new login screen appears after a restart for your credentials to be entered.

Figure 4.6 Installation completed screen

5. Ubuntu server details after successful login as a user.

Figure 4.7 Server login Details

6. Log in with administrator rights as root.

4.1.2 Download and Installation of Certified Asterisk 11.7.1 LTS (long term support).

For this install I used Asterisk 11.7.1 and will be compiling from source at

http://downloads.asterisk.org/pub/telephony/asterisk-11-current.tar,gz

49
Before you begin the install process you will want to be sure that you server OS is up

to date. When the update completes, the server will reboot. Logged in as root type in

the following:

50
4.2 Integration of Hardware

Integration is the next step after installation of the Asterisk server. Analogue phones are

going to be integrated using ATA with our IP PBX server. Here we have following steps to

integrate. Refer to Figure 3.15 for the hardware connection of ATA

Connect a standard touch-tone analogue telephone to the PHONE port.

Insert a standard RJ11 telephone cable into the Phone1 port and connect the other end

of the telephone cable to the analogue telephone.

Insert the Ethernet cable into the WAN port of HT502 and connect the other end of

the Ethernet cable to an uplink port a router.

Connect a PC to the LAN port of HT502 if it is being used as a router.

Insert the power adapter into the HT502 and connect it to a wall outlet.

Figure 4.8 Integrating HT502

The HT502 has two FXS port. Both FXS ports can have a separate SIP account. This is a key

feature of HT502 as it supports simultaneous calls on both FXS ports.

51
4.3 Implementing the features of Asterisk Server

4.3.1 Voice Call

The voice call is the basic property of unified communication system, voice call is based on

sip protocol. Communication is only allowed for those who are registered with the sip server.

Communication devices can work on voice feature to provide good sound quality.Now

Asterisk is installed and running so we are going to create 6 new users with their extensions

and their voicemails.

1. Creating a new sip.conf and configuring it

Type the following to create a new sip.conf

sudo vi /etc/asterisk/sip.conf

NB: A screen shot can be found at appendix A- 5

Users include:

Users / User Locations Extensions Password

B080311004(Student) 100 1234abcd

Front Desk 1001 1234abcd

Rad-Voice 1000 1234abcd

Registry 2000 1234abcd

Account 2001 1234abcd

Faculty 2002 1234abcd

52
Table 4.1 Credentials

[B080311004] [Front desk]

type=friend type=friend

secret=1234abcd secret=1234abcd

host=dynamic host=dynamic

context=phones context=phones

[Accounts] [Faculty]

type=friend type=friend

secret=1234abcd secret=1234abcd

host=dynamic host=dynamic

context=phones context=phones

[Registry]

type=friend

secret=1234abcd

host=dynamic

context=phones

2. Create the extensions for the users.

To create the extensions for the users we need to modify the file extensions.conf. If we dial

the number 1001 we should contact Radford University Front desk and if we dial the number

2000 we should contact the registry office. We also create a special extension "1000" to dial

into an IVR (Interactive Voice response). We also create a special extension "*100" to access

to the voicemail main menu for extension B080311004.

53
4.3.2 Voice mail

Voicemail is configured to handle calls that cannot be answered. Voicemail is generally made

is to call user group. Flow chart of voicemail can be seen in Figure 3.16

Figure 4.9 Flow Chart Voicemail

Personalized voicemail is a feature that allows callers to leave messages on phone. Voicemail

permits users to record outgoing message, so that when calls are routed voicemail callers will

hear greeting and have the option to leave a message. The voicemail message will also

provide a timestamp that informs when a caller contacted. This is an essential feature of

Asterisk server that allows a voice attachment activated for a particular user to enable voice

mail go to the extension profile that user, then enable status of voice mail.

54
4.4 Configuration

Asterisk server configuration will be done via a CLI console, which demands knowledge of

Linux to configure. Configuration is carried out also in accordance with the purposes of the

Radford University College. The following is a configuration that has been done:

4.4.1 Configuring LAN-VoIP user

Configuring all LAN-VoIP users through Asterisk server whether it is IP telephone or

analogue telephone adapter by creating SIP account for them. All communication devices

communicate through Sip protocol and all communication devices appear like LAN-VoIP

users for Asterisk server. Figure 3.17 shows this scenario of configuration.

Figure 4.10 Configuration of Lan- VoIP user

55
4.4.2 Configuration of E1200 Router

Cisco Linksys E1200 Wireless-N Router is also web based, so we need access it via the web

interface by using the default IP. 192.168.0.1.

Figure 4.11 GUI of Cisco E1200 Router

We need to change the Network IP address to match our Asterisk server network address since

the router will serve as a gateway.

We Assign 192.168.1.1 to the router.

We proceed by disabling/ unchecking the DHCP option as the Asterisk server will be handling

the assigning of IP’s to the various devices both wirelessly and wired. You will need to set a

56
4.4.3 Configuration of 3CX

After creating extensions in the Asterisk server we start by creating sip profiles in 3cx. We

proceed by a click on create profile. After that we need to do account setting. The last stage is

shown in Figure 3.19

Figure 4.12 Configuration of 3CX Soft phone

ACCOUNT SETTINGS & SIP CREDENTIALS:

Account name: Compulsory entry which is the name of the extension given you by admin.

Caller ID: User defined detailing you ID

Extension: Unique extension number with which users can call given by Admin

ID: Same as the extension

Password: SIP authentication Password

*Settings are applicable to both Desktop and mobile/smart device.

57
4.4.4 Configuration of Zoiper

Click on the Settings menu and select “create a new account”.

Figure 4.13 Create a new account and account Type

A new page will appear to select the type of account you want to make. This will usually be

SIP. Select the type of account you want to configure and click on the “Next” – button.

Credentials

Your provider or system administrator should have provided you with a username, password

and possibly a hostname.

Figure 4.14 Account credentials

Fill in the username on the first line and the password on the second line.

If your administrator provider you with a domain, proxy, registrar, hostname, outbound proxy

or server field, please fill enter it on the last line.

58
4.4.5 Configuration of IP Telephone

Polycom Sound Point IP450 is a next generation small-to-medium business IP phone that

features 3 lines with 3 SIP account, a 128x40 graphical LCD and 3-way conference. The IP

450 delivers superior HD audio quality, rich and leading edge telephony features,

personalized information and customizable application service, automated provisioning for

easy deployment, advanced security protection for privacy, and broad interoperability with

most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for

small-to-medium businesses looking for a high quality. Figure 3.20 and Figure 3.21 showing

the GUI and Configuration of IP phone respectively.

To set up the Polycom IP450, follow the below step:

There are slots at the back side of phone , Connect the handset and main phone case

with the phone cord

Connect the LAN port of the phone to the RJ45 socket of a hub/switch or a router

(LAN side of the router) using Ethernet cable

Connect the 24V DC output plug to the power jack on the phone; plug the power

adopter into an electrical outlet.

Now use your keyboard and make configuration through GUI while entering the IP

address in web browser.

The device can be configured through a given IP address by the Asterisk DHCP server that is

192.168.1.105. When the IP address is types in the address bar of any browser we get

Figure 4.15 Configuring Polycom IP Phone

59
Once logged into Polycom Sound Point IP450 web UI using its default username and

password, Polycom and 435 respectively. Navigate to the Lines section at the top -> Line 1

and fill out the following credentials:

Figure 4.16 Configuration of IP phone

Display name: SIP Extension Number

Address: SIP Extension Number

Authentication User ID: SIP Extension Number

Authentication Password:

Label: SIP Extension Number

Address: IP address of the Asterisk server

Port: The port doing the listening

Leave all other settings as default and click the submit button

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4.4.6 IVR (Interactive Voice Response)

IVR is a useful service such as automatic answering machine before the user can connect the

caller with the desired number of VoIP. If a call comes into the IP-PBX server, pending

configuration, calls will be automatically sent to the IVR, then the user can interact with the

IVR, before the call transferred to the division or VoIP users (Front desk or Account Office)

in accordance with user needs.

For more details can be seen in Figure 4.17

Figure 4.17 Sample IVR scenario 1

61
Interactive Voice Response is said to be a digital receptionist. An IVR plays the recorded text

to the caller and ask them to press the key to connect to an organization, work group, a

person or etc. then IVR send the call to the destination.

With registered extensions, Asterisk can be set to meet our needs. It is possible that we want

the system automatically connected to the extension we already defined if our extension

didn‘t reply, thus configuration of the telephony systems.

4.5 Testing of RUC VoIP System

In this section we present ways in which testing of the deployed VoIP server has been

conducted. The IVR presented on Figure 4.17 is the RUC IVR system. Customers should dial

1000 in order to get through 2 different 2 offices. It present a path to reach to RUC front

desk, and account Office.

As a prototype system IVRs have been used to direct users to the departments they need to

enquire from. When the call is directed to a department it will be directed to the secretary of

that department, this one will perform a transfer according to the choice of the caller.

The same system will be implemented to support call that will be directed to other faculty.

The IVR also should provide options in which customer will choose in order to join

conferencing room. When calls are directed to URC ICT call department, agents will take

care of these callers. And when all agents are all occupied, callers will be held in queue.

The system should provide music on hold and announce the position to customers which are held

in queue. A sample below shows how customers (users) will be disposed in order to make

62
calls to each other. Softphones on PCs and laptops will send registration request to the VoIP

server, such that they can be registered and used while making calls.

Figure 4.18 Sample RUC network Architecture

Figure 4.19 Incoming Call from Accounts Office <2001> to Registry <2000>

63
The system has been configured to provide voice calls; Figure 4.20 shows a Zoiper softphone

with extension number <2000> from the <Registry> being called by a 3CXPhone softphone

from the <Account> Department with extension number <2001>.

Figure 4.20 Call established between the Registry Department and the Account Office

4.6 Verification of VoIP metrics

The verification of VoIP metrics consist of determining if VoIP requirement has been

established. Wireshark tool has been used in order to find the values associated to the

jitter,delay and packet loss. The values of these parameters have been used to determine the

values of MOS for Voice and Video calls. Remember that MOS is evaluated between 0(not

recommended) and 5 (excellent) signal qualities.

64
We use Wireshark to capture RTP packet that are crossing the network in order to be

analyzed. The filtering option on Wireshark permits me to isolate SIP and RTP packets so

that they can be represented graphically. Figure 4.21 displays filtered packets (RTP).

Figure 4.21 Wireshark screenshot when RTP packets are filtered

The filtered packet will be analyzed by using a utility that Wireshark provide. Open

Wireshark 1.10.8, filter interface which you are using to transmit communications then filter

the RTP protocol.

To analyze RTP stream click to Telephony menu RTP stream analysis.

The opened table provides me a number of sequence of packets, jitter, delay (Delta), packet

loss, IP bandwidth per packet. And finally it provides the maximum delay, packet loss

65
percentage and the maximum and mean jitter; their values are presented in Appendix A-3.

The following figures display jitter Vs packet sequence and delay Vs packet sequence.

Figure 4.22 Delay and jitter Vs packet sequence when only voice is transmitted

During voice call the maximum latency is 21.5 ms and maximum jitter is 0.72ms while in

video calls maximum delay 31ms and maximum jitter is 1.83 ms.

4.6.1 Security Issues

Security issues must be taken into consideration, while you are planning to deploy a system

for a large institution.

4.6.2 Change user main password and root default password

In order to grant confidentiality and authentication to the server configuration files Asterisk

default main (admin) password and root passwords must be changed from default to the ones

the admin prefers.

66
4.6.3 Avoiding SIP authentication requests from all IP addresses

This is done by using the “permit=” and “deny=” lines in sip.conf to only allow a reasonable

subset of IP addresses to reach each listed extension/user in the sip.conf file. Even if

accepting inbound calls from “anywhere” (via [default]) don’t let those users reach

authenticated elements!

4.6.4 Wireless phones require advanced wireless security

Many VoIP phone systems offer wireless handsets for mobility. These implementations often

make use of existing 802.11x wireless solutions. Weak wireless security exposes VoIP

vulnerabilities. Do not give hackers the opportunity to wirelessly access your network. The

best wireless solutions require centralized network authentication, in addition to wireless

encryption. If you are using lower end wireless access points, you should at the very least use

WPA (a form of encryption) over WEP encryption, although a novice hacker can easily

defeat WEP.

4.6.5 Physical security

All terminating equipment (such as switches, routers, and Asterisk server itself) should be

secured in an environment that can only be accessed by authorized persons.

67
 5 CHAPTER FIVE

Conclusion and Recommendations

5.1 Conclusion

The objective of this project was to provide a VoIP system that the Radford University

college campus can use to interconnect its users (student and stuffs). The provided system is

based on Asterisk; Asterisk was created in 1999 by Mark Spencer of Digium. Like any PBX,

it allows attached telephones (hard phones & softphones) to make calls to one another, and to

connect to other telephone services including the public switched telephone network and

permit to deliver other VoIP services like voicemail, video calls, queuing, call parking&

transfer and conferencing call.

To verify that the service that the system is offering are appreciated by customers

and that the meet ITU-T requirements, QoS was implemented and verification test has been

conducted on pilot basis. From these tests I found, RUC VoIP system can be established

without network problems. Different telephony features has been configured while

developing the projects. The system is capable of delivering Voice, Voicemail, and IVR has

been created and configured. The tested system provides to the administrator a way of

configuring the system at any end point (PC) RUC network using a command line interface.

Quality of Service is shown by the delay and packet loss by transferring packets from

IP PBX network and by receiving packets from IP PBX network. Delay of phone displays the

highest delay of about 2.5 seconds compared with the SIP phone. Quality of Service is not good

while communicating phone to any SIP phone, this is likely due to the noise from the wireless

network there is in the air and due to ATA (Analogue Telephone Adapter).

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5.1.1 Server Capacity

As specification of server will increase then more efficient IP PBX will be because of higher

processors that will process more call and manage data base.

5.1.2 User Capacity

IP PBX user capacity is highly dependent on factors - factors as following.

Bandwidth

Bandwidth provided by Radford University College is 5 MB, then for VoIP-based voices is

enough to meet. The greater the bandwidth provided, the smaller delay caused.

Codec

Codec’s are used in determining the capacity of any user who capable of server capacity.

Because of this codec provides a measure of Different sampling. In this final project, the

codec is used G711 (PCMU) with sampling at 64 kbps.

5.2 Future Recommendations

After research that has been conducted on Radford University College IP network, the

following recommendations have been suggested;

Implement a VoIP server which will work on the Radford university College IP

network, to support the call establishment and provide different telephony features

like Video calls, voicemail, IVR, conferencing and queuing system.

Ghanaian institutions and enterprises in general, should implement this VoIP system

since it is cost effective and can interoperate with their existing Data networks.

69
Integration with PSTN Network

Asterisk can connect with the existent PSTN by using FXO telephony card, so it is

possible to be used as the VoIP gateway this will increase portability and decrease cost.

Integration with GSM Network

This project can also be integrated with GSM network through gateways. This will increase

portability and decrease cost. This project can also be integrated with Skype gateways

through which we can call from our all communication devices to any Skype ID.

Development of Admin web interface to help manage calls

Due to limitations such as time a Command Line Interface was used to manage calls and

configure calls. I was only able to develop a prototype web interface by typing into your

browser the server IP address. It such need a lot of improvement.

High security for large scale enterprise network

When developing the large scale enterprise network by connecting multiple Asterisk

servers located in different sites based on IAX2, to realize high security is the issue

because the voice data is not encrypted. To solve this issue, VPN method could be

established by using Open VPN.

70
REFERENCES
Meisel, J.B and Needles, M. (2005). “Voice over internet protocol (VoIP) development “,

info, Vol 7 No. 3, pp.3-15.

Oliver Hersent, Jean Pierre Petit, and David Gurle, “Beyond VoIP Protocols:

Understanding Voice Technology and Networking Techniques for IP Telephony”,

1st Edition. March 4th

Theodore Wallingford, (2005). “Switching to VoIP”, O’Reilly Media, Inc June, Print

ISBN-13:978-0-596-00868-0

Tim Szigeti & Christina Hattingh, (2013). End-to-End QoS Network Design: Quality of

Service for Rich-Media & Cloud Networks, 2nd Edition. November 26

Jim Van Meggelen, Leif Madsen, and Jared Smith, (2007). Asterisk: The Future of

Telephony, 2nd Edition

Parker, M., and D. Van Doren, (2009). "Achieving Cost and Resource Savings with Unified
Communications."

L. Sun, (2004). “Speech Quality Prediction for Voice Over Internet Protocol, PhD”

University of Plymouth, School of Computing, Communications and Electronics Faculty of

Technology, January 24

Mohamed Boucadair, (2009). “Inter-Asterisk Exchange (IAX): Deployment Scenarios in

SIP Enabled Networks”. ISBN: 978-0-470-77072-6

71
Buckley, Sean, (2001). “Packet over Cable.” Telecommunications, March 4.

Lawrence Harte, (2003). Telecom Basics: Signal Processing, Signalling Control, and Call

Processing October 1st

Barrie Dempster and Kerry Garrison, (2006). Trixbox Made Easy: A Step-by-step Guide

to Installing and Running Your Home and Office VoIP System. September 1st

Stephan S. Jones, (2009). The Basics of Telecommunications August 9th

Puglia, Vincent, (2010). "Unified communications: The search for ROI through

tomorrow’s business communication solutions".

Accessed from http://scholarworks.rit.edu/theses/233

Frost & Sullivan, (2007). Advanced capabilities of VOIP

Accessed from https://www.tremcom.com/ilink/ilink_and_voip.htm

Andre Mouton and Julie Roy, (2003). IP telephony represents a structured

cabling revolution May 1.

Accessed from www.cablinginstal.com/articles/print/volume-11/issues-

5/contents/technology/ip-telephony-represents-a-structure-cabling-revolution.html

Craig Hyatt, (2007). University’s Phone Bills Reduced by 81% with 3CX Phone System

for windows January.

Accessed from http://www.3cx.com/case-studies/unc/

Nick Galea, (2007). IP PBX: Ten reasons to switch February 10th.

72
Accessed from http://searchunifiedcommunications.techtarget.com/tip/IP-PBX-Ten-
reasons-to-switch

Stephen J. Occhiogrossp, (2011). CCIE or NULL! : The Cisco PPDIOO Life Cycle May 9.

Accessed from https://ccie-or-null.net/tag/ppdioo/

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 APPENDIX
A: LIST OF ABBREVIATIONS

Abbreviations Full Form

ACK Acknowledge

ATA Analogue Telephone Adapter

CLI Command line Interface

DAHDI Digium/Asterisk Hardware Device Interface

DTMF Dual Tone Multi Frequency

FXO Foreign Exchange Office (Port)

FXS Foreign Exchange Station (Port)

GUI Graphical User Interface

GSM Global System for Mobile (Communication)

HTTP Hypertext Transfer Protocol

74
 I

IAX Inter-asterisk exchange

IP Internet Protocol

IVR Interactive Voice Response

IETF Internet Engineering Task Force

ISDN Integrated service Digital Network

LAN Local Area Network

MAN Metropolitan Area Network

PBX Private Branch Exchange

PSTN Public Switched Telephone Network

PCMU Pulse Code Modulation MEO-Law

PCMA Pulse Code Modulation A-law

QOS Quality of Service

RSVP Resource Reservation Protocol

RTP Real Time Protocol

75
 RTCP Real Time Control Protocol

RUC Radford University College

SIP Session Initiation Protocol

SIP URL SIP Uniform Resource Locator

SDP Session Distribution Protocol

SMTP Simple Mail Transfer Protocol

TCP Transmission Control Protocol

TDM Time Division Multiplexer

UC Unified Communications

UDP User Datagram Protocol

VPN Virtual Private Network

VoIP Voice over Internet Protocol

WAN Wide Area Network

WLAN Wireless Local Area Network

76
B-1 Screenshot of Sip.conf

B-2 Screenshot to proposed RUC VoIP System GUI

77
B-3 Screenshot of Extensions.conf

C-1 Mean Response table

Class Response Response Type Category Response

1xx Information Provisional

2xx Success Final

3xx Redirection Final

4xx Client Error Final

5xx Server Error Final

6xx Global Failure Final

78
C-2 SIP Response Code.

Class Response Type Code Command

1xx Informational 100 Trying
Request is accepted and followed by 180 Ringing
processing the request
181 Call in being forwarded
182 Queued
2xx Success 200 OK
Message received and understood
3xx Redirection 300 Multiple choices
301 Moved permanently
Further action needs to be done to 302 Moved temporarily
308 Alternative service
complete the request
4xx Client error 401 Unauthorized
402 Payment required
Request cannot be processed by the server 403 Forbidden
404 Not Found
405 Method not allowed
406 Not acceptable
407 Proxy authorized
408 Request time out
409 Conflict
410 Gone
411 Length required
413 Request message too large
414 Request URL too large
415 Unsupported media type
420 Bad extensions
480 Not available
481 Call log
482 Loop detected
483 Too many hops address
484 Incomplete
485 Ambiguous
5xx Server error 500 Internal server error
501 Not implemented
Request cannot be processed by the server 502 Bad gateway
503 Service unavailable
504 Gateway time out
505 SIP version not supported
6xx Global error 600 Busy everywhere
603 Decline
604 Doesn’t exist
605 Not acceptable

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C-3 SIP Requests

Request Description
name

INVITE Indicates a client is being invited to participate in a call session.

ACK Confirms that the client has received a final response to an INVITE request.

BYE Terminates a call and can be sent by either the caller or the callee.

CANCEL Cancels any pending request.

OPTIONS Queries the capabilities of servers.

REGISTER Registers the address listed in the To header field with a SIP server.

PRACK Provisional acknowledgement.

SUBSCRIBE Subscribes for an Event of Notification from the Notifier.

NOTIFY Notify the subscriber of a new Event.

PUBLISH Publishes an event to the Server.

INFO Sends mid-session information that does not modify the session state.

REFER Asks recipient to issue SIP request (call transfer.)

MESSAGE Transports instant messages using SIP.

UPDATE Modifies the state of a session without changing the state of the dialog.

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D-2 Asterisk

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