ABSTRACT
Communication
currently now plays an important role in our daily life. Without communication
in this new era, nothing seems to be easily done, managers, executives, even
presidents of nations need to be able to communicate in order to transmit
information to the world and the people inside it. Communication is basically the transmitting
and receiving of information from one individual to another individual.
This
Project is aimed at enhancing communication of the society either over the
internet, via mobile phones, gadgets, pagers be it in the Department of
Computer science or even the environment we find ourselves in. These forms of
communication need the use of internet protocol and also the use of Internet to
effectively work. With this, we can get a hold of information one-time wherever
we are, whenever we want, and however we want it.
The
IP-PBX system will run on a linux based software, which will run an open source
package, Asterisk, which then establishes calls between users, and also
offering other features which includes, voicemail, call parking, etc.
TABLE OF CONTENTS
CHAPTER1: INTRODUCTION
1.0. . Introduction on Public Switched Telephone Network
1.1
Brief
Background on IP-Private Branch Exchange
1.2
Advantages
1.3
Motivation
and Significance of Study
1.4
Statement of Problem
1.5
Aim and Objective
1.6
Equipment’s and Software’s
CHAPTER 2:
LITERATURE REVIEW
2.1
Private Branch Exchange (PBX)
2.1.1 Functionality
2.2 Internet Protocol-Private
Branch Exchange
2.3 Functionality
of Internet Protocol Private Branch Exchange
2.4 Local Area
Network (LAN)
2.5 Telephony
Adapter (ATA)
2.6 Network
2.6.1 Circuit Switched Network
2.6.2 Packet Swicthed Network
2.6.2.1 Connection Oriented Packet Switched Network
2.6.2.2 Connectionless Packet Switched Network
2.7 Communication
Over Internet Protocol
2.8 Voice Over Internet
Protocol
2.9 Protocol Used For
VoIP
2.9.1 H.323 Standard
2.9.1.1 Components of H.323
2.9.2 Session Initiation Protocol (SIP)
2.9.2.1 Components in Sip Call
2.9.2.2 SIP Messages
2.9.2.3 Overview of Sip Operations
2.9.2.4 SIP Addressing
2.9.2.5 Locating a Sip Server
2.9.2.6 SIP Transaction
2.9.2.7 SIP Invitation
2.9.2.8 Locating a User
2.9.2.9 Changing an Existing Session
2.9.3.0 Sample of a SIP Operation
2.9.4 Session Description Protocol (SDP)
2.9.5 Real-TimeTransport Protocol
2.10 Asterisk
2.10.1 Functionality
2.10.2 Architecture
2.10.3 Asterisk API
2.10.4 Dial Plan
2.10.5 Core
2.10.5.1 Channel
2.10.5.2 Technology Driver for SIP
2.10.5.3 Reading Process in Details
2.10.5.4 Writing Process in Details
2.10.6 Call Setup with SIP
in Asterisk
2.10.6.1 Translating Audio Data
CHAPTER3: DESIGN AND IMPLEMENTATION
3.1 Introduction
3.2 System Architecture
3.3 Client to Server
Communication
3.4 System Design
3.5 Dial Plan
3.6 Voicemail
3.7 Audio Conferencing
3.8 Directory Listing
3.9 Bandwidth Requirement
and Calculations
3.9.1 Selection
3.10 Implementation
CHAPTER 4: RESULT AND TESTING
4.0 Introduction
4.1 Test and Result
4.2 Summary
CHAPTER5: CONCLUSION AND
RECOMMENDATION
5.0 Conclusion
5.2 The Impact of This
Project
5.3 The Recommendation and
Further Research
REFERENCES
APPENDIX: Program Codes
LIST OF TABLES
Table
3.1:
Tabular comparison of different CODECS and their bandwidth consumption
Table
4.1:
Scenario 1 – Making a call
Table
4.2:
Scenario 2 – Joining a conference
Table
4.3:
Scenario 3 – Retrieving Voicemail.
LIST
OF FIGURES
Figure
1.0:
IP-PBX Design using SIP
Figure
2.0: Architecture
of SIP Signaling
Figure
2.1 Shows
the diagram of a SIP Operation
Figure
3.0: Asterisk Core And Its Four API
Figure
3.0: IP-PBX Design Using
Client – Server Communication
Figure
3.1: The
Counter Path X-Lite Soft Phone
Without any SIP Account Configured
Figure
3.2: X-Lite Sip Account Settings Menu Clicked From the
Downward
Pointing Arrow
Figure
3.3.: X-Lite Sip Account Settings “Popup Window” For Adding Clients
Figure
3.4: X-Lite
Client(1001) Properties “Popup Window” With IP-PBX Settings
Figure
3.5 Extension 1001 is now registered
Figure
3.6:
Extension 1001 Dials
Extension 1000, and Extension 1000 starts ringing
Figure
3.7:
Call
Establishes Between The Two Clients
Figure
3.8:
Client With Extension 1000
Ends The Call And Hangs Up
CHAPTER
ONE
INTRODUCTION
1.0 BRIEF
INTRODUCTION ON PUBLIC SWITCHED TELEPHONE NETWORK
The
main objective of a Public Switched Telephone Network (PSTN) is to create and
maintain audio connections between two recipients in order to carry information
(in form of data and voice) [1]. Humans can sense sound vibrations in the range
of 20–20,000 Hz, most of the sounds humans make when speaking tend to be in the
range of 250–3,000 Hz [2]. Since the purpose of the telephone network is to
transmit the sound wave (i.e. voice) of people speaking, it was designed with a
bandwidth in the range of 300–3,500 Hz. This limited bandwidth means that some
sound quality will be lost, especially in the higher frequencies [3].
In
the PSTN, the famous Last Mile is the final remaining piece of the telephone
network still using technology pioneered well over a hundred years ago. One of
the primary challenges when transmitting analog signals is that systems
transmitting these signals generate random unwanted sounds (noise) which would
interfere with those signals, the effects of noise gives rise to signal lost
and distortion. Instead of trying to preserve an analog waveform over distances
that may span thousands of miles, why not simply measure the characteristics of
the original sound and send that information to the receiver?, the original
waveform would not get there, but all the information needed to reconstruct it
would. One major way of opposing the noise effect is by sampling the
characteristics of the source waveform, store the measured information, and
send that data to the receiver. Then, at the receiver’s end, use the
transmitted information to generate a completely new audio signal that has the
same characteristics as the original [4].
1.1 BRIEF BACKGROUND ON INTERNET
PROTOCOL-PRIVATE BRANCH EXCHANGE (IP-PBX)
An
IP-PBX is a private branch exchange (telephone switching system within an
enterprise) that switches calls between VoIP (Voice over Internet Protocol or
IP) users [5].
A
typical IP-PBX can switch calls between a VoIP user and a traditional telephone
user (an user with an analog phone), or between two traditional telephone users
in the same way that a conventional PBX does [6].
The
abbreviation may appear in various texts as IP-PBX, IP/PBX, or IPPBX [7]
A
conventional PBX uses separate networks for voice and data communication, but
in the case of an IP PBX, it uses a single network for both voice and data
simultaneously, this gives IP PBX an advantage over PBX by using converged
networks for both voice and data communication [8].
This
means that Internet access, VOIP and traditional telephone communications, are
all possible using a single line/channel to each user, thereby providing
flexibility and also reducing maintenance costs.
Voice over IP (VoIP, or voice over Internet Protocol) can be defined
as the transmission of voice signals via internet protocol networks such as the
internet [9].
The service uses Internet service to connect an
internet protocol telephone device (or soft phone) to a similar device, or to
the public switched telephone network, in order to connect to any telephone in
the world.
This technology has been in use for decades now, by
businesses (multinational IT companies), basically to reduce the cost
long-distance calls (both international and local). Its application also allows
free computer to computer calls.
The basic premise of IP PBX using SIP as a VoIP
protocol is the packetization of audio streams for transport over Internet
Protocol-based networks [10]. The challenges to accomplishing this relate to
the manner in which humans communicate. Not only must the signal arrive in
essentially the same form that it was transmitted in, but it needs to do so in
less than 150 milliseconds [11].
If packets are lost or delayed, there will be
degradation to the quality of the communications experience, meaning that two
people will have difficulty in carrying on a conversation. The main purpose of
a telephone is to allow people to communicate. It is a simple goal, and it
should be possible for communication to happen in far more flexible and
creative ways than are currently available to us [12].
1.2
ADVANTAGES
The IP PBX system becomes less expensive to setup as
most of the Hardware functionality such as Echo cancellation, Digital signal
processing (DSP) is being ported to software (Asterisk) [13].
1.3
MOTIVATION AND SIGNIFICANCE OF STUDY
This study is to show the importance of
communication in our daily lives. Communication is a very important factor in
this 21st century. Without communication, activities will not be
easily done. The Motivation for this Project arose as the need for a
communication system became necessary for the day to day office communication in
the Department of Electrical and Information Engineering. The combination of an
open source operating system, telephony software and hardware will decrease the
cost of telephony in the department. This project has given me the opportunity
to work with professionals in the networking industry.
1.4 STATEMENT
OF THE PROBLEM
The challenge is to design an IP PBX system using an
open source operating system, PBX hardware and software and also using the
Session Initiation Protocol (SIP) for the transfer of voice and data signal
between users.
1.5 AIM
AND OBJECTIVE OF THE STUDY
The objective of this project is to develop an IP
PBX server which will manage calls between each office in the department with
audio conferencing capability and providing the following features: Call
Processing, Voicemail, Interactive Voice Response (IVR), Call forwarding and
Call Conferencing. Etc.
1.6
EQUIPMENT’S AND SOFTWARE’S
Hardware:
(a)
Pentium 4, 400MHz, 2GHz RAM, 80GB HDD,
CD-Rom
(b)
CAT-5E RJ 45
Software(s):
(a)
Ubuntu Linux
(b)
Asterisk
(c)
Asterisk Add-on
(d)
X-Lite Soft phone
(e)
DHCP 3
(f)
PuTTy
(g)
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