DESIGN OF AN INTERNET PROTOCOL-PRIVATE BRANCH EXCHANGE SYSTEM BASED ON OPEN-SOURCE SOFTWARE


Content

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)           Notepad++

 

 

 

 

 

 

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