INTERNET PROTOCOL PRIVATE BRANCH EXCHANGE

A PROJECT THESIS SUBMITTED IN PARTIAL FULFILLMENT OF THE REQUIREMENT FOR THE DEGREE OF

Bachelors of Engineering Electrical (Telecommunication) Engineering

By

Nisar Ahmed Memon Muhammad Muzzamil Shaikh Muhammad Aslam Dall

Under The Supervision of

Engineer Ghulam Abbas Electrical Engineering Department

SUKKUR INSTITUTE OF BUSSINESS ADMINISTRATION 2013

Dedication

This thesis is dedicated to our parents,

For their endless love, support and encouragement

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Certificate

This project thesis is written by Nisar Ahmed Memon, Muhammad Muzzamil Shaikh and Muhammad Aslam D a l l under the direction of their supervisor and approved by all the members of thesis committee, has been presented and accepted by the Head of Department of Faculty of Electrical Engineering Department in partial fulfillment of the requirements of the degree of BACHELORS OF ELECTRICAL (specialization in Telecommunication) ENGINEERING.

H.O.D
(Project Supervisor) Electrical Engineering

Internal Examiner

External Examiner

Director Sukkur IBA

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Acknowledgement

From the very beginning, we are very grateful to Almighty Allah, Who gave us the opportunity, strength, determination and wisdom to achieve our goal. We would like to thank Engineer Ghulam Abbas (Sukkur IBA), who not only served as our supervisor but also encouraged and challenged us throughout our research project. He patiently guided us through the process, never accepting less than our best efforts. We would like to thanks Bilal Ahmed Shaikh (Sukkur IBA) for their insightful suggestions and guidance. Many of our colleagues in academics have made significant contributions to the working on this project. Our special thanks go to Professor Dr Madad Ali Shah for his vital encouragement and generous support throughout the working and experimenting the project, we would also like to acknowledge and extend our heartfelt gratitude to worthy Director Nisar Ahmed Siddiqui for providing us financial support for completing this project. The most important is to express our gratitude to our parents for all the sacrifices. They have been fully supported on this project. Their blessings and prayers have been a great inspiration for us to finish this project.

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Abstract
Unified Communication is the latest research topic and many organizations are working on it in all over the world. Every organization is trying to push and extend the boundaries of unified communication. In unified communication system the latest software is Elastix, based on Asterisk Server, which serve as the local exchange for placing voice and video calls within a private Wi-Fi cloud and legacy networks. The work proposed in this project added features for placing the voice and video calls and mobile phones (smart phones) hence increasing the mobility of the users. The model is successful in carrying out voice and video calls on android supported handhelds connected with the wireless network and PCs connected with both wired LAN and wireless LAN. Every user is provided with his own extension number, the communication devices can make voice call, video call, voice mail, Instant messaging and Interactive voice response, that can be used to connect within organization. We use here Elastix for the successful completion of this project; Elastix is an open source software platform which uses Asterisk PBX (Private Branch Exchange) as the kernel to build unified communications system. It can choose the combination of different communication components to achieve customized solutions. This project defines the structure and functions of Elastix. It implemented the functions of VOIP (Voice over Internet Protocol) like voice call, video call, chat and voice mail. This Project provides great portability, flexibility and cost effective solution to organization. This project is the integration of hardware and software .We have Asterisk based Elastix server that provide Unified Communication to clients. The different types of communication devices like android, IP telephone, Laptops, Desktops, and Hard telephone are connected to server. PCMU is one of the transport protocol used in VoIP communications. Bandwidth required by the active channel which is determined by the codec used, the server and client codec used is PCMU/G.711. G.711 requires a minimum bandwidth for each channel is 64Kbps. So with bandwidth-owned (2MB) VoIP server can serve a maximum of 20 VoIP calls to 24 calls simultaneously. Application is very crucial section as our whole business (providing IP-PBX services) depends upon it means the target market which wills actually the people, responsible for generating money or increasing our sales .Our target market includes: Corporate organizations, Institutes, Universities, Health care, Airports, Hotels, Page | iv

Banks and many more places. This project is economic, cost effective, have full control to the administrator, provide mobility throughout the world. Feasible, Web based administration modified, 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.

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Table of Contents

Topics

Page Numbers

Dedication ...........................................................................................................i Certificate................................................................................................................ii Acknowledgement..............................................................................................iii Abstract ................................................................................................................ iv List of Figures........................................................................................................x List of Tables........................................................................................................xi

CHAPTER 1 Introduction ........................................................................................ 1 1.1 Introduction to IP PBX ....................................................................................... 1 1.2 Problems and Challenges .................................................................................... 2 1.3 Contribution towards Knowledge ...................................................................... 3 1.3.1 Features .............................................................................................................. 4 1.4 Aim and Objectives .............................................................................................. 4 1.4.1 Aim ..................................................................................................................... 4 1.4.2 Objectives........................................................................................................... 5 1.5 Applications of IP PBX ........................................................................................ 5 1.6 Structure of Thesis ............................................................................................... 6 CHAPTER 2 Literature Review ............................................................................... 7 2.1 Introduction to Literature Review ..................................................................... 7 2.2 Computer Networking ......................................................................................... 7 2.2.1 Understanding computer networks ................................................................. 7 2.2.1.1 Peer to peer.............................................................................................. 7 2.2.1.2 Client - Server ......................................................................................... 8 2.2.2 Benefits of Computer Networks ...................................................................... 9 2.2.2.1 Benefits for the needs of enterprise computer networks ..................... 9 2.2.2.2 The benefits of a computer network for public needs ......................... 9 Page | vi

Topics

Page Numbers

2.2.4 Reference model of DOD (Department of defense) ..................................... 10 2.2.5 Reference Model OSI (Open Systems Interconnection) .............................. 10 2.3 VoIP (Voice over Internet Protocol) ................................................................ 12 2.3.1 VoIP Protocols ................................................................................................. 12 2.3.1.1 SIP (Session Initiation Protocol) ......................................................... 12 2.3.1.2 Composition of SIP Protocol ............................................................... 13 2.3.1.3 Components of SIP ............................................................................... 13 2.3.1.4 Address on SIP ...................................................................................... 14 2.3.1.5 Messages on SIP .................................................................................... 14 2.3.1.6 SIP request ............................................................................................ 14 2.3.1.7 SIP response .......................................................................................... 15 2.3.2 Type of VoIP network configuration ............................................................ 17 2.3.2.1 Phone via the Internet .......................................................................... 17 2.3.2.2 Communication between IP-based devices ........................................ 18 2.3.3 Quality of VoIP Matrix .................................................................................. 18 2.3.3.1 Latency .................................................................................................. 19 2.3.3.2 Delay ...................................................................................................... 19 2.3.3.3 Jitter ....................................................................................................... 19 2.3.3.4 Packet loss ............................................................................................. 20 2.3.3.5 Sequence error ...................................................................................... 20 2.4 Soft Switch .......................................................................................................... 20 2.5 Summary ............................................................................................................. 21 CHAPTER 3 Hardware Implementation .............................................................. 22 3.1 Introduction to Hardware Implementation..................................................... 22 3.2 Equipments used in Project .............................................................................. 22 3.2.1. Hardware ........................................................................................................ 22 3.2.2 PC Server required as IP PBX ...................................................................... 24 3.2.3 Software ........................................................................................................... 25 3.3 Preparation Phase .............................................................................................. 25 3.3.1 Bandwidth........................................................................................................ 25 3.3.2 Network architecture ...................................................................................... 25 3.3.3 Soft switch ........................................................................................................ 26 Page | vii

Topics

Page Numbers

3.3.4 Soft phone ........................................................................................................ 26 3.3.5 Elastix ............................................................................................................... 26 3.3.6 Connection ....................................................................................................... 27 3.4 Implementation Phase ....................................................................................... 27 3.4.1 Download Elastix ............................................................................................ 27 3.4.2 Install Elastix server ....................................................................................... 27 3.4.3 3CX Phone Soft phone .................................................................................... 29 3.4.4 Grand Stream HT 502 ATA........................................................................... 29 3.5 Integration of Hard ware .................................................................................. 30 3.6 Implementing the features of Elastix server .................................................... 31 3.6.1 Voice call .......................................................................................................... 31 3.6.2 Video call.......................................................................................................... 31 3.6.3 Voice mail ........................................................................................................ 31 3.6.4 Voicemail to Email Notification ..................................................................... 32 3.7 Configuration ..................................................................................................... 33 3.7.1 Configuring VoIP user ................................................................................... 33 3.7.2 Configuration of HT502 device...................................................................... 33 3.7.3 Configuration of 3CX ..................................................................................... 35 3.7.4 Configuration of IP Telephone ...................................................................... 35 3.7.5 Configuring Ring Group ......................................................................... 37 3.8 IVR (Interactive Voice Response) .................................................................... 38 3.8.1 IVR Configuration .......................................................................................... 40 3.8.2 Configure Inbound ......................................................................................... 41 3.9 Installation .......................................................................................................... 41 3.9.1 Installing Openfire .......................................................................................... 42 3.9.2 Install Spark Client. ........................................................................................ 44 3.10 Summary ........................................................................................................... 46 CHAPTER 4 Results and Discussions.................................................................... 47 4.1 System Testing Process ...................................................................................... 47 4.1.1. Registration of VoIP user .............................................................................. 48 4.1.2 Calls fellow user VoIP .................................................................................... 49 4.1.3 Incoming calls (Inbound) ............................................................................... 50 Page | viii

Topics

Page Numbers

4.2 VoIP user capacity ............................................................................................. 51 4.2.1 VoIP server computer specs ........................................................................... 51 4.2.2 Bandwidth capacity ........................................................................................ 51 4.3 Comparison between Xlite and 3CX. ............................................................... 51 4.4 Analysis PCMU .................................................................................................. 52 4.4.1 Delay the call. .................................................................................................. 52 4.5 QoS Measurement .............................................................................................. 52 4.6 Summary ............................................................................................................. 56 CHAPTER 5 Conlusions and Future Recommendation ...................................... 57 5.1. Conclusions ........................................................................................................ 57 5.1.1. Server Capacity .............................................................................................. 57 5.1.2 User Capacity .................................................................................................. 57 5.2. Future Recommendations ................................................................................ 58 5.2.1 Integration with PSTN Network ............................................................ 58 5.2.2 Integration with GSM Network ............................................................. 58 5.2.3 Integration with others Unified Communication Systems ................... 58 5.2.4 Communication without side SIP network ........................................... 58 5.2.5 Telephony Interface Cards ..................................................................... 58 5.2.6 High security for large scale enterprise network .................................. 58 References ................................................................................................................. 59 List of Abbreviation ................................................................................................. 63 Glossary .................................................................................................................... 65

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List of Figures
Figure No. Figure 2.1: Figure 2.2: Figure 2.3: Figure 2.4: Figure 3.1: Figure 3.2: Figure 3.3: Figure 3.4: Figure 3.5: Figure 3.6: Figure 3.7: Figure 3.8: Figure 3.9: Figure 3.10: Figure 3.11: Figure 3.12: Figure 3.13: Figure 3.14: Figure 3.15: Figure 3.16: Figure3.17: Figure3.18: Figure3.19: Figure 3.20: Figure 3.21: Figure 3.22: Figure 3.23: Figure 3.24: Figure 4.1: Figure 4.2: Figure 4.3: Figure 4.4: Figure 4.5: Figure 4.6: Figure Title Page Number

Peer to Peer Network....................................................................08 Client Server...................................................................................09 Phones through the Internet...........................................................17 IP based communication................................................................18 VoIP network..........................................25 Logo Elastix...................................................................26 VoIP Networks..............................................................................27 Elastix Server.....28 3CX Phone......................................................................................29 3CX Logo........................................................................................29 Grand Stream HT 502........................................................30 Integration.......................................................................................30 Flow chart Voice mail ...............................................31 Email Notification......................................................................32 Configuration.............................................................................33 GUI of HT502...............................................................................33 GUI of HT502...............................................................................34 Configuration of Telephone.............................................................34 Configuration of soft phone .......................................................35 GUI of IP phone............................................................................36 Configuration of IP phone. .......................................................36 Configuration of Ring group........................................................37 Forward call ring group ...............................................................38 Flow chart IVR System...............................................................39 Ring Strategy...............................................................................40 Inbound Route .............................................................................41 Openfire.......................................................................................44 Spark client..................................................................................45 System testing process..............................................................47 Registration of user VoIP..............................................................48 Calls between VoIP users.............................................................50 Bandwidth VoIP Server...............................................................51 (a) Packet loss soft phone (b) Packet loss IP phone ........54 (a) Packet loss Analogue Phone (b) Packet loss Mobile..55

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List of Tables
Table No. Table Title Page Number

Table 2.1 DOD TCP/IP and the OSI reference model. .........................................11 Table 2.2 Mean response class..............................................................................15 Table 2.3 SIP response code..............................................16 Table 3.1 Specifications of the tools used.............................................................23 Table 4.1 Comparison between 3CXphone and Xlite...........................................52 Table 4.2 MOS values with G.711 codec based R factor......................................53

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CHAPTER 1

INTRODUCTION

1.1 Introduction to IP PBX

The Internet is a global system of computer networks interconnected using the standard Internet Protocol Suite (TCP / IP) to serve many users around the world. It is a network of private business, community, academic, and international or local governments are connected to each other. Such solution, among employees can communicate with the Internet that has been provided by the office. Customers can communicate with VoIP over IP networks [1]; so as to reduce communication costs can offset the cost of the customer. Customers can also directly contact the VoIP number of employees, so that the aims and objectives can be delivered direct customer. There will be no any need for extra infrastructure for the telephone network and dont need to install the costly equipments; we just need ATA telephone adapter to provide this service. When you call someone then it rout to the prescribe person through the Internet and if the person is not responding to the analogue phone, the call automatically routed to the smart phone or any other phone which must be android supported. The purpose is to improve the mobility in the network as well as on a single line performing multiple functions. The beauty of this project is that all things you can manage user, and limitations for the few employees s can also done by using Elastix server. To design a unified communication network in which communication can be possible through PC, android phone, IP telephone and standard phone under the supervision of server that monitors its registered clients effective and efficient manner. Every technology in this world moving towards new trends and changing very rapidly and becoming cost effective[6]. Each users can communicate wherever they are, even though the area is very difficult for the cable network, provided that the area can access the Internet, It is also very Cheap, because it automatically uses VoIP voice communication costs are very low, because the cost of a call is not affected from a distance or close the communication was made, but from out of the use of Internet access, No one can ignore its Mobility factor, by using WLAN network, so that users can communicate in mobile conditions. Almost every organization use PBX , but this project is more cost efficient with many extra embedded feature in a low cost that is why every organization , hospitals , educational sectors or NGOs will be demanding product like this project. Every organization which need PBX inside it is necessary to installed this project rather to installed the old concept old PBX. Page | 1

VoIP (Voice over Internet Protocol) telephone network , the Internet is a network that uses the Internet as a communication medium, so the client can use for VoIP everywhere can connect to the Internet or TCP / IP network[12]. Unified communication is the integration of real time communication services such as Instant messaging(chat), presence information, telephony(including IP telephony),video conferencing, data sharing(interactive white board), call control and speech recognition, with non real time communication services such as unified messaging(integrated voice mail, email, SMS and fax)[8]. Asterisk is Linux based IPBX application developed by Mark Spencer, Elastix evolved from the core Asterisk. Elastix is an open source unified computing Server software to establish Unified Communications that brings together IP PBX, IM and collaboration functionality[4]. Its goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution. It was released as a Linux distribution with asterisk and it has web interface that gives all its customization option to user. Elastix server has database to store all information of its clients such as voicemail, live active and non active calls and recording voices for announcements and IVR(Interactive Voice Response).All clients must be registered by entering its Local IP and extension number along with secret code (the will be unique for all clients). Elastix has a good support for telephony hardware. Elastix also support other phone brands thanks to the SIP and these protocols are based on public available standards. For this reason any Manufacturer can build a product that supports them. In addition to these, the report also contains the details regarding the different type of communication problems which people facing these days. Above all, this report gives a detailed description of Internet Protocol Private Branch Exchange System. This description is empowered with the experimental analysis of the system and the observed practical calculations. This report will be of help for those who wish to understand and diagnosed traffic on Internet and want to introduce tax free platform of communication.

1.2 Problems and Challenges

Phone for an office is not an odd Again, since the phone was first introduced in the world, offices is the main target of the most maximum phone usage. Ranging from the use of a phone for business, local, long distance, and international offices contributed high numbers the overall use of the phone for telephone operators at world. In the office, the phone also became burden for monthly expenses. High costs first this time because the phone calls made to mobile numbers, International Direct Dialing (IDD) and Direct Connection Long Distance (DLD) which adds to the monthly telephone charges swollen for office. Another big problem is cost for the separate infrastructure to build the PBX exchange inside the corporation. As the times spending the cost of mobile operating increasing, Page | 2

specially for the mobile users have much problems they are facing expense in terms of sells tax. These all problems can be handled by this technology. This project provide our businesss services at a demandable (presentable) price to meet the customers needs. People face the limited scalability and extensibility in the existing systems, there is no database maintaining facility available so we will provide that. Time wastages is also the another big issue in which no other person focusing but this is the top concerned of this project is to use effectively and efficiently. It required dedicated line to complete a call and also limited mobility of users It is obvious to having the problems in every project therefore in making this final boundary problem is made as follows. Configuration of clients soft phone, hard phone and network through server is big task to complete. VoIP client using an IP Phone, It is impossible to all have the mobile phone which is android supported and the last thing is design of voice communication systems using the Phone Handful. Making this project available throughout the world is difficult task rather it is also not easy in smart organization. We have to make sure the availability of Internet in the organization for the successful completion of all calls. Higher the charges of calling with respect to distance.

1.3 Contribution towards Knowledge

This project performs great contribution in proprietary PBX (Private Branch Exchange). The proposed solution not only solves (burden for monthly expenses of offices, separate infrastructure to build the PBX exchange inside the corporation, limited mobility of users dedicated call lines) problems of PBX but also adds numerous novel features. User is provided ease of access to PBX and mobility as well. The solution is built on IP PBX server. Some of the main contributions of the proposed solution are as followed: i) Scalability. This solution has the great advantage of being able to easily add new phone numbers or extensions without the need for extra costs and setup time associated with traditional telephony. ii) Portability. A great advantage of VoIP is that it is Internet based; meaning any Internet-enabled device that has communication functionality can be used to send and receive calls via your VoIP telephone network. This makes phone calls as convenient as easy as plug-and-play in most cases. iii) One Wiring system. Instead of separate wiring for telephones and separate wiring for data, all data and voice are on the LAN. There is usually plenty of bandwidth available on a well designed LAN. Page | 3

iv) Web based administration. Through this project, all system administration functions are performed on the network usually through a browser based administration program. This means that the system can be modified from anywhere if required. v) Integration. This solution is integrated different communication devices like soft phone, IP phone android even hard phone for easiness of user.

1.3.1 Features
i) Peer to Peer phone calls All calls are Peer to Peer. This is a big advantage over the traditional PBX. The call is set up by the VoIP server then the call flows between the two endpoints. All of the voice or video traffic is direct between the two endpoints reducing the congestion at the server. So the optimum bandwidth is used. ii) Peer to Peer Video. Video sessions can be set up between endpoints. iii) Private Instant messaging. This solution also provides Instant Messaging. With a IP PBX system, Instant Messaging can be limited to corporate business eliminating some of the security issues associated with public Instant Messaging sites and provides complete control to management. iv)Voice mail. The great feature this solution is Voice Mail that allows you to receive user voice messages even when user phone is switched off user phone is busy. user can retrieve these messages easily. v) Interactive Voice Response. This solution has used pre-recorded voice prompts and menus to present information and options to callers, and touch-tone telephone keypad entry to gather responses. IVR solutions enable users to retrieve information including bank balances, flight schedules, product details, order status, movie show times, and more from any telephone. Additionally, IVR solutions are increasingly used to place outbound calls to deliver or gather information for appointments, past due bills, and other time critical events and activities.

1.4 Aim and Objectives

1.4.1 Aim
The purpose of this project is to build real and non real time (unified communications) applications by using open source software platform, Elastix (Server), which uses Asterisk PBX as the kernel.

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1.4.2 Objectives
To design IP based PBX network architectures To customize efficient and effective Soft Switches To implement real time and non-real time applications Design Global network by joining small IP based PBX Integrate Analog Phone and configure these phone Installation of server and troubleshoot all problems Registered SIP account and give all features of system

1.5 Applications of IP PBX

This project has been designed for PBX, besides this can be used of wide range of application. These include the following sample applications. 1.5.1 Education Sector This project that completely replaces proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. This project is far less expensive than a traditional PBX and can reduce call costs substantially. Its webbased administration makes phone system management easy. soft phone System eliminates the phone wiring network and allows users to easily work .Students can easily communicate with teachers without any cost by using verities of features like voice mail. Voice call, video call. Students can have group chat through which they can discuss their subjects issues. 1.5.2 Business Sector/ Hotels Professional service, rich guest experience and systematic hotel activities are the building blocks for success in the hotel industry. Guest centric hotels require specialized communication solution to automate hotel operations and help their staff to respond from anywhere in the hotel premise. This project are scalable as per the hotel requirements. This project Boost Staff Efficiency and Productivity and also reduce Operate cost. 1.5.3 Hospitals This project is great support for health care centers to save human lives and save time of doctors to monitor more and more patients. It will be better if we know in advance which doctor is free and which doctor is busy so that patient will be provided quick treatment and to save his/her life by using features .This system is implemented with

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the help of Elastix software having features IVR (Interactive Voice Response), call recording, remote extension, intercom, conference call, Voice mail. 1.5.4 Corporate Organization A flexible telephone system capable of many hundreds of extensions if necessary with full voicemail and IVR functionality for automated attendants This project provide Advanced functionality for automated appointment reminder phone calls and automated laboratory result messaging for employees to obtain information using a secure and automated telephone system. Interactive functionality for employees to confirm appointments and schedule new appointments. Reduced overall cost of the telephone system in general and telephony costs on a monthly basis. 1.5.5 Banks In todays banks, more and more banks are deploying open-source IP-PBXs, such as Asterisk, and other SIP-based communications servers in their networks. Developers and resellers of such systems need to be able to complement the central IP-PBX with other network elements that will provide their customers with a full solution.

1.6 Structure of Thesis

The thesis comprises five chapters, the details of the subsequent chapters is given as under: CHAPTER II: LETRATURE REVIEW This chapter describes the theory - the basic theory that would used in designing and building a network system VoIP-based, Asterisk and Elastix. CHAPTER III: HARDWARE IMPLEMENTATION This chapter contains about In designing this system, Tools and materials used in the system design will be discus in this chapter. These things will be used in this project will be discussed in this chapter, Software Elastix 2.3.0, device, Soft phone and IP Phone. CHAPTER IV: RESULTS AND DISCUSSION In this chapter contains a discussion and analysis of the topic. Final assignments are made. CHAPTER V: CONCLUSIONS AND FUTURE RECOMMENDATIONS Chapter five concludes the present work & shows future recommendation of the undertaken research Page | 6

CHAPTER 2

LITERATURE REVIEW

2.1 Introduction to Literature Review

VoIP protocol is used in VoIP transport so that voice data can be sent properly, SIP protocol is used, the following explanation of the SIP. SIP protocol is supported by some protocols, such as RSVP to make a reservation on the network, RTP and RTCP media for transmitting and know the quality of service, as well as media SDP to describe the session [23]. SIP network is used, there are two types of network servers, namely: Proxy server is a server that receives the request, processes it, and forwards the requests it receives to the next hop server after changing some headers in the request message[12]. The configuration will require a form of gateway interfaces that connect VoIP networks to the Internet network.

2.2 Computer Networking

The computer network is the set of "interconnection" between two or more computers connected to the transmission media cable or wireless (wireless).

2.2.1 Understanding computer networks

Two computers can be said to be connected if they exchange data / information, a variety of owned resource, such as files, printers, storage media (hard disk, CD-room, and flash disk). Data in the form of text, audio, and video media moving through wires or wirelessly enabling computer users in computer networks to exchange files / data, print on the same printer and using hardware / software that is connected in a network together. In computer networking system known connections between computers, namely: 2.2.1.1 Peer to peer Peer to peer network is a computer network consisting of multiple computers. Peer to peer is a model in which each PC can use the resource on another PC or give its resource to use another PC. In other words, can serve as a client and a server in the same period. The peer to peer is the system known as workgroup windows, in which Page | 7

each computer in a network are grouped in a working group. For example, there are several computers in one department group named according to the department concerned. Each computer assigned an IP address from the IP of the same class to be able to share with each other to exchange data or resource owned by each computer, such as printers, CD room, and fie. Figure 2.1 is an illustration of peer to peer.

Figure 2.1 Peer to peer Network 2.2.1.2 Client - Server Client system - the server can be applied to the local network and can also be applied to Internet technology, where there is a computer unit that serves as a server that only provides services to other computers, and a client who also just request a service from a server. Client can only use the resources provided by a server in accordance with the authority granted by the administrator. Applications that run on the client side is a resource available on the server, or application that is installed on the client side but can only be run after connecting to the server. Figure 2.2 is an illustration of the client server with a server that serves the general. Page | 8

Figure 2.2 Client server

2.2.2 Benefits of Computer Networks

Benefits for the user computer network can be grouped into two, namely to the needs of the company and to the public network. 2.2.2.1 Benefits for the needs of enterprise computer networks Resource sharing that aims to make the whole program, particularly the equipment data, can be used by everyone on the network without being influenced by the location of resources and users. High reliability obtained because of the availability of alternative resources. For example, all the files can be copied to all machines so that if one machine dies, then the file can still be accessed from other machines that are still active. 2.2.2.2 The benefits of a computer network for public needs Access to information residing elsewhere can be directly updated, such as today's news info, e-government, e-commerce or e-business. Page | 9

Person-to-person communication, such as chat, email, video conference, as well as voice over Internet protocol (VoIP) Interactive entertainment, just as watching TV shows online, streaming radio, downloads and browsing.

2.2.4 Reference model of DOD (Department of defense)

DOD model is important because of its role in making known the basics of Internet connection in use today. TCP / IP is the protocol type of the first DOD reference model used in relationship / connection between computers in a global computer network (the Internet). Many of the terms and concepts used in the Internet connection from the terms and concepts used by the TCP / IP protocol.

2.2.5 Reference Model OSI (Open Systems Interconnection)

This model is intended to be an open system, developed by the ISO (International Organization for Standardization). Open systems can be interpreted as an open system to communicate with other systems. To sum up, this model is referred to as the OSI model only. Table 2.1 represent the DOD, TCP / IP and the OSI Reference model

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Table 2.1 DOD, TCP / IP and the OSI Reference model Model OSI N Layer o Model DOD Protocol TCP/IP Name Protocol Usefulness

Applic ation

Proces s/ Applic ation

Presen tation

Protocols for IP distribution DHCP (Dynamic Host network with a limited Configuration Protocol) number of IP Database engine domain DNS (Domain Name Server) name IP address FTP(File Transfer Protocol) Protocol for file transfer HTTP (Hyper Text Transfer Protocol to transfer HTML Protocol) files and Web MIME(Multipurpose Internet Protocol for sending binary MailExtension) files in text form NNTP (Network News Protocol to receive and send Transfer Protocol) newsgroups Protocol to retrieve mail from POP (Post Office Protocol) the server Protocol to transfer various SMB(Server Message Block) DOS and Windows file servers SMTP (Simple Mail Transfer The protocol for the exchange Protocol) of mail SNMP (Simple Network Protocol for network Management Protocol) management Telnet Protocol to remotely access TFTP (Trivial FTP) Protocol for file transfer NETBIOS (Network Basic BIOS standard network Input Output System) RPC(Remote Procedure Call)

Sessio n

Remote procedure calls Input Output for BSD-UNIX SOCKET network types TCP (Transmission Control Oriented data exchange Host Protocol) protocol (connection oriented) Transp to ort UDP (User Datagram Data exchange protocol nonHost Protocol) orientation (connectionless) IP (Internet Protocol) Routing protocol to set Netwo Interne RIP (Routing Information rk t Routing protocol to select Protocol) Page | 11

2.3 VoIP (Voice over Internet Protocol)

Voice over Internet Protocol known as IP Phones. In general, VoIP is defined as a system that uses the Internet to transmit voice data packets from one place to other using IP protocol intermediaries. In fact, VoIP is more focused on using the Internet compared with traditional phone infrastructure built earlier[29]. VoIP (voice over Internet protocol) telephone network, the Internet is a network that uses the Internet as a communication medium, so the client can use for VoIP everywhere can connect to the Internet or TCP / IP network. VoIP systems employ session control and signaling protocols to control the signaling, set-up, and tear-down of calls. They transport audio streams over IP networks using special media delivery protocols that encode voice, audio, video with audio codecs and video codecs as Digital audio by streaming media. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs. Some popular codecs include law and a-law versions of G.711, G.722 which is a high-fidelity codec marketed as HD Voice by Polycom, a popular open source voice codec known as iLBC, a codec that only uses 8 Kbit/s each way called G.729, and many others. VoIP is available on many smart phones, personal computers, and on Internet access devices[29]. With VoIP technology, it is expected the three types of public communications services following has the same quality as the previous technology (which bitabene more expensive): Service with a normal voice communication Voice mail service that can be left on the number dialed Service delivery fax transmission at a reasonable cost

2.3.1 VoIP Protocols

Protocol VoIP protocol is used in VoIP transport so that voice data can be sent properly, SIP protocol is used, the following explanation of the SIP. 2.3.1.1 SIP (Session Initiation Protocol) SIP is a protocol multimedia issued by the group incorporated in Multiparty Session Control (MMUSIC) within the organization Internet Engineering Task Force (IETF) as documented in a Request For Command document (RFC)[15].SIP is a protocol that is at the application layer that defines the initial, modification, and termination Page | 12

(termination) of a multimedia communication session. Multimedia communications sessions include relationship, distance learning, and other applications. Characterized SIP client-server, this means that the request is given by the client and the request is sent to the server. Then, the server processes the request and provide a response to the client. Request and response to the request is called a SIP transaction. 2.3.1.2 Composition of SIP Protocol SIP protocol is supported by some protocols, such as RSVP to make a reservation on the network, RTP and RTCP media for transmitting and know the quality of service, as well as media SDP to describe the session.[17] By default, SIP uses UDP protocol, but in some cases may also use TCP as the transport protocol. 2.3.1.3 Components of SIP In connection with the IP phone, there are two components in SIP systems, namely: User agent User agents are end systems that are used to communicate. User agent consists of two parts, namely: User agent client (UAC) UAC is designed application on the client to initiate SIP requests User agent server (UAS) UAS is an application server that tells the user if it receives the request and provides a response to the request. The response can be either to accept or reject the request. Network server In order for SIP users on the network can initiate a call and can also call, the user is first doing register in order to know its location. Registers can be done by sending a REGISTER message to the SIP server. User location can vary so as to get the actual location of the user required a server location. In SIP networks, there are two types of network servers, namely: Proxy server Proxy server is a server that receives the request, processes it, and forwards the requests it receives to the next hop server after changing some headers in the request message. Next hop SIP server can form or another server where the proxy server does not need to know. Proxy servers can function as a client and a server as a proxy server can provide response and request.

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2.3.1.4 Address on SIP The SIP network has the address given attribute SIP URL (SIP Uniform Resource Locator) to be easily recognizable. SIP URLs are used in SIP networks are shaped like an email address 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. 2.3.1.5 Messages on SIP Overall, the SIP message consists of two parts, the request and the response. When a client sends a request message, the server will respond to the message with the response message. Request and response messages consist of a start-line, one or more configurable headers or commonly called the message header, an empty line end of the header fields and message body that defines the communication session. SIP message format can be seen below, Generic message = start-line (in message request), Status-line (in message response), Message header, Empty line and Message body. 2.3.1.6 SIP request INVITE This message is used to initiate a communication. Message body INVITE message description of media that can be used to communicate. ACK This message serves notify the client has received a final response to the INVITE. Message body in an ACK message can read the description of the media that will be used by the user who invoked (call). If the message body is blank means call agree with the message body contained in the INVITE message. CANCEL CANCEL message request is sent to deliver a message that has been sent previously, before the server sends a final message response. BYE This message is sent by the client to terminate the communication

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OPTIONS This message is sent by the client to the server to determine its capabilities. REGISTER Client can register its location by sending a REGISTER message to the SIP server where the server can receive SIP REGISTER called registers.

2.3.1.7 SIP response Response message is sent after receiving a request message indicating the success status of the server. Response message is defined by three numbers, the first number is the class of the response. The second and third numbers indicate the meaning of the response. Table 2.2 shows the value of the class is on SIP response.

Table 2.2 Mean response class. Class Response 1xx 2xx 3xx 4xx 5xx 6xx Type Response Informational Success Redirection Client error Server error Global error Category Response Provisional Final Final Final Final Final

Response messages are divided into two categories, namely: Provisional The response is a response sent by the server to indicate the process is ongoing, but not end the call. Final Response was given that terminate SIP response code transaction SIP. See Table 2.3 for the SIP response [53].

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Table 2.3 SIP Response Code. Class Type Response Code 100 Informational 180 Request is accepted and followed by 181 processing the request 182 Success 200 Message received and understood 300 Redirection 301 Further action needs to be done to 302 complete the request 380 401 402 403 404 405 406 407 Client error 408 409 Request cannot be processed by the 410 server 411 413 414 415 420 480 481 482 483 484 485 500 501 502 503 Command Trying Ringing Call is being forward Queued OK Multiple choices Moved permanently Moved temporarily Alternative service Unauthorized Payment required Forbidden Not found Method not allowed Not acceptable Proxy authorized Request time out Conflict Gone Length required Request message too large Request URL too large Unsupported media type Bad extensions Not available Call log Loop detected Too many hops address Incomplete Ambiguous Internal server error Not implemented Bad gateway Service unavailable Page | 16

1xx

2xx

3xx

4xx

Client error 4xx Request cannot be processed by the server

Server error 5xx Request cannot be processed server

504 505 600 603 604 605

6xx

Global error

Gateway time out SIP version not support Busy everywhere Decline Doesnt exist Not acceptable

2.3.2 Type of VoIP network configuration

Some kind of combination of the subsystems will form some VoIP configuration, but with additional supporting systems. Generally, VoIP network configuration there is two types, namely: 2.3.2.1 Phone via the Internet This configuration uses PSTN or PABX facilities on both sides of the terminal subsystem. This configuration will require a form of gateway interfaces that connect VoIP networks to the Internet network. For this configuration takes an additional system that can map a telephone dialing code IP better known as the call manager. Illustration of the configuration can be seen in Figure 2.3

Figure 2.3 Phones through the Internet

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2.3.2.2 Communication between IP-based devices

Basically, this type of configuration as much on the field of software development (software) multimedia alone, have not noticed a problem setting the transmission medium. This configuration requires a signaling system that is not too complicated, so it is only in certain circumstances be required signaling management software. The system also requires a minimum of a gatekeeper. Illustration of an IP-based communication between devices is given Figure 2.4

Figure 2.4 IP-based communications between devices

2.3.3 Quality of VoIP Matrix

Understanding QoS (Quality of Service) is the ability of a network to provide better service to the traffic data. QoS is not obtained directly from the existing infrastructure, but obtained by implementing the network in question. VoIP application is a real-time application, so it cannot tolerate delay (within certain limits) and packet loss[29]. Delay Internet is huge, exceeding even delay that occurred in mobile. Fatherly reduce this delay, many ways to go, one of which is to optimize the use of bandwidth, set the queuing method used, and using management protocols to manage data packets being passed. In other words, set up QoS on a VoIP network. For the purposes of VoIP, there are requirements that must be met by an Internet network infrastructure, namely: The network must have a clear policy settings Page | 18

Network bandwidth must meet the minimum standards of application There is an order of priority data packets on the network

Without these three, the administrator cannot guarantee QoS network and will result in decreased quality of sound received by the terminal. QoS in IP phones are the parameters that indicate the quality of network data packets. Some declared QoS parameters for IP telephony include latency, delay, jitter, packet loss and sequence errors on the Internet. 2.3.3.1 Latency Latency is the time required by a device of asking for the right of access to the network to gain access rights. There are two types of latency, namely real and induced. Real latency associated with the physical network and switching characteristics of the transport media. Induced latency is the delay caused by queuing delay in the network equipment (such as Ethernet cards, routers), delay the process on the other end system and network congestion between the source and destination. 2.3.3.2 Delay One of the design considerations in implementing voice transmission is one-way delay minimization or end-to-end delay. Delay is the most critical parameter in the Quality of Service. There are several causes of delay include: Congestion Lack of traffic shaping on method Data packets with different sizes Change the speed of the network between WAN Compaction bandwidth suddenly

Voice traffic is real-time traffic so that if the delay in the delivery of voice packets is too big, given utterance cannot be recognized. Maximum delay that can be tolerated in accordance with the ITU G.114 standard is less than or equal to 150 ms. 2.3.3.3 Jitter Jitter caused by variations in time of receipt of the data packets from the sender to the receiver. This parameter can be handled by adjusting the method of queuing at the current router is congested or when a change in speed occurs. However, jitter may not be eliminated, but can be minimized by seeking ways each and TIPA data packets via the same pathways. Page | 19

2.3.3.4 Packet loss Packet loss in IP telephony network has a major effect, where if there is a certain amount of packet loss will cause TCP slow interconnect happen. Typically 10% packet loss cannot be tolerated. 2.3.3.5 Sequence error Congestion in the network may cause packets take different routes to achieve the same goal. As a result the package up in a different order.

2.4 Soft Switch

Soft switch is a generic term for a new approach to switching technology, the terms therein regarding call control, call processing. Because soft switch is a generic term that comes the understanding that some defined though some vendors and standardization bodies. Here below are some of the different definitions of soft switch vendors and some of the international consortium, which are: a) ISC (International Soft switch Consortium) proposes a model of soft switch as an intelligent system which performs the function call control in a VoIP network. ISC describes the soft switch as a system that covers all things related to NGN communication system that uses open standards to create integrated networks by combining the intelligence service capabilities in handling voice traffic, data and multimedia services more efficiently and with potential value-added services are much greater than the PSTN. b) I-Link and Dialup Audio is a company engaged field Internet and security network. Experience moving Internet world produce a product such as soft switch, better known as IPPBX. Soft switch here focuses on the technology that connects the gateway between networks. c) According to Sun Microsystems, Soft switch is a collection of products, protocols, and applications that allows any device to access the Internet and telecommunications services over IP networks. When viewed closely, soft switch is a set of technologies that perform switching functions by establishing end-to-end communication. Soft switch constitute future communication concept developed from the approach PSTN, VoIP and data networks. Communication system designed to deliver voice, data and multimedia services as well as well designed to penetrate the PSTN to migrate to the data network.

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2.5 Summary
This chapter presents the related research per formed in the computer networking and also the understanding and benefits about this. There is the comparison between Reference Model of DOD (Department of Defense) and Reference Model OSI (Open System Interconnection) which are the types of communication system. TCP / IP is the protocol type of the first DOD reference model used in relationship / connection between computers in a global computer network (the Internet). SIP URLs are used in SIP networks are shaped like an email address user @ host where user can be any user name, phone number, or the name of the agency. Voice over Internet Protocol (VoIP) is defined as a system that uses the Internet to transmit voice data packets from one place to other using IP protocol intermediaries. Discuss VoIP protocols including the SIP (Session Initiation Protocol) protocol and the composition, components, messages and response of it. Quality, Latency, Jitter and Packet loss of the VoIP is also the part of this chapter. SIP message format like Generic message = start-line (in message request), Status-line (in message response), Message header, Empty line and Message body Client can register its location by sending a REGISTER message to the SIP server where the server can receive SIP REGISTER called registers. Soft switch is a collection of products, protocols, and applications that allows any device to access the Internet and telecommunications services over IP networks . This chapter presented the detailed discussion relating to the VoIP and its related technologies and their development. The next presented the detail discussion of the system design and architecture, system components, software requirements and its specifications, solution overview and more important the implementation phase..

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CHAPTER 3

HARDWARE IMPLEMENTATION

3.1 Introduction to Hardware Implementation

Implementation is the one of the important part of our thesis. In this we will discuss about the implementation of the Elastix, Openfire and Spark and also lights on the hardware used in this project configuration of all software. We will describe the features of the project. To configure eth0 or Ethernet card that has been installed on the server can be configured, and then select enable IP4 support, and finally enter the IP Address and enter the IP DNS 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 Elastix server whether it is IP telephone or analogue telephone adapter by creating SIP account for them.

3.2 Equipments used in Project

There is the list of equipments listed here which we are going to use in this project for the completion of project. There is combination of software, hardware and the open sources libraries. As for the equipment used software is an open source program that is free program.

3.2.1. 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. Overall the hardware used in building a IP PBX server is listed below.

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Table 3.1 Specifications and Description of the Tools. No Tool IP PHONE GXP 1400 2 line keys with dual-color LED (2 SIP account and up to 2 call appearances). 3 XML programmable contextsensitive soft keys, 3-way conference. HD wideband handset, hands-free speakerphone with advanced acoustic echo cancellation. Phonebook with up to 500 contacts and call history with up to 200 records ANALOGE ADAPTER 2 Model: GrandstreamHT502 Features: 2FXS Port +2 RJ 45(LAN/WAN) Ethernet Port +Router TELEPHONE Specification

ANALOG TELEPHON

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

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

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No Tool LAPTOP

Specification

Dell N5520 5 Intel Core i3,2.4GHz / Memory 6Gb / HDD 750 Gb / 15.6

HANDPHNE

Android based

3.2.2 PC Server required as IP PBX

A PC or an Elastix Appliance to run the IP PBX. If you have chosen the PC route, it is recommended that you run a dedicated PC for this purpose. The PC described below (minimum) will be sufficient to power the IP PBX in a small office or home environment. Therefore dont throw away that old Pentium III clunker you have in the attic. 800 MHz Pentium III PC or better (P4 will give extra comfort). 312MB RAM the more the better 8GB minimum hard disk space (dependant on your usage of MOH, announcements, voice recording etc). 10/100 NIC CD-ROM Drive 10/100 4 or 8 ports Ethernet hub/switch (not required if your router has spare ports. This is dependent on how many extensions you are planning). Naturally if you are running Elastix in a heavy environment, you will need heavier duty and better specification system. When you install Elastix on this old computer, it will take it over it starts by formatting all the hard disks (if you happen to have more than one), so make sure there is nothing on the machine that you want to keep. Page | 24

3.2.3 Software
Software used is an open source software application as a soft phone 3CXPhone and VoIP Elastix as a server, because the use of the application program does not require an activation fee. Programs that used only two, namely: A. Linux Elastix-2.3.0-i386 as soft switch B. Soft phone 3CXPhone

3.3 Preparation Phase

Preparation needs to do is prepare the VoIP network in general, we are going to implement the project in Sukkur IBA while using all resources of Sukkur IBA.

3.3.1 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.

3.3.2 Network architecture

The network architecture is shown here. The Figure 3.1 showing the interconnection of the hardware components between different devices.

Figure 3.1 VoIP networks built drawings Page | 25

3.3.3 Soft switch

Asterisk TM is a Linux based IPBX application developed by Mark Spencer of Digium, the company behind Asterisk. Elastix evolved from the core Asterisk [13]. It is made up of several major components. These were developed under GPL supported relatively by users themselves. It consists of applications, a provisioning system, an installer, and an operating system that, together, make a complete package ready for use as an out of the box PBX. Within this document, Elastix and Asterisk will be referred to frequently and they are interchangeable as Elastix is in essence a superset of the Asterisk. Elastix chosen because it is easier to configure, has better graphics display, and a lot of forums that have discussed about the Elastix so as to facilitate the installation and configuration process.

3.3.4 Soft phone

Soft phone used is 3CX Phone. 3CX chosen because it has call forward features required by the company. In other soft phone call forward facility existed, but some require advance registration, and display less user friendly. From some soft phone that has been used, eventually the various considerations soft phone 3CX Phone chosen as call forward facility and also in terms of appearance that is easy to understand.

3.3.5 Elastix
Elastix is an operating system made by Asterisk and CentOS[35]. Elastix is open source software create a media platform unified communications or "Unified Communications Platform, "which consists of a component or module technologies commonly used communication media today such as: Voice call, video call ,voice mail, instant messaging, a fax server, VoIP and video conferencing. Almost all of the modules can be managed and configured through a graphical interface, where Elastix It supports advanced features such as voicemail, fax-to-email, soft phones, including the CRM system (Customer Relationship Management) and many others. This can be in the Elastix software download at www.elastix.com. Elastix logo is shown in Figure 3.2

Figure 3.2 Elastix Logo Page | 26

3.3.6 Connection
The medium used for the VoIP user can connect to server is the Internet. So the user can connect to VoIP server via the Internet wherever they may be. In this final user can connect to the server via the Internet. Connection of VoIP is shown in Figure 3.3

Figure 3.3 VoIP Network

3.4 Implementation Phase

At the implementation stage is divided into two, namely the installation and configuration. For a server installation and configuration process performed at the location of Sukkur IBA. Following the implementation of which has been implemented in building VoIP server.

3.4.1 Download Elastix

The installation process will be discussed in this report Elastix installation of the operating system on the server. This process will be explained as follows. Go to the official website of the Linux based Elastix Asterisk sever, namely www.elastix.org then download Elastix and burn it into CD [41].

3.4.2 Install Elastix server

Turn on the PC and change the boot order of drives CD / DVD room Then install Elastix operating system into a server that has been prepared. Elastix main view have two options, namely through the GUI mode and Text mode (in the discussion of the GUI mode is selected and then press enter) After waiting a while until the process is completed as preparation. After that, the dialog box appears Choose a Language, select the desired language e.g. English. Page | 27

Then Keyboard Type dialog box will appear, select the type of keyboard used Warning dialog box appears informing you of the approval to delete the data on the partition that has been created. If you want to delete then select yes. Since the HDD is used there is no data that is important, it is better to choose the option erasing ALL DATA. Then on the next option select remove All partitions to format the HDD as a whole then click OK to partition by default. To configure eth0 or Ethernet card that has been installed on the server can be configured, and then select enable IP4 support, and finally enter the IP Address and enter the IP DNS and IP Gateway. When it is to give a name for the hostname, with IPPBX. The time zone selection select zone Asia / Islamabad, and then enter the password after that process will begin. Wait until the files have been copied, after which the installation is complete. Then pass before the system reboot to complete the installation, the system will install a boot loader. Then enter the password for MySQL is available on the server. Next is to enter the admin password. This password will be used when configuring the server through a browser application, such as Mozilla or Google chrome. It have finished installed Elastix server and can be configured via the web with the IP address 192.168.208.160 Now when we write the IP address of Elastix server in Mozilla web browser we have following GUI based Elastix server This is the Main Manu of Elastix server telling about status of server.

The main menu of Elastix server in shown in Figure 3.5

Figure 3.4 Elastic Server Page | 28

The first thing that we need to do is to give static IP to this sever otherwise the DHCP server will change the IP after certain duration.

3.4.3 3CX Phone Soft phone

3CX is a soft phone that is used as a connector between one phone call to another phone call under the supervision of Elastix server. As Soft phone 3CX is chosen because 3CX have call forwarding features and call transfer required by any organization to connect either their employees or customers and As the 3CX soft phone uses SIP protocol and we have Elastix SIP server that can easily handle SIP protocol based soft phone. 3CX can be installed on a laptop, PC and Android based Mobile phone. 3CX can be downloaded from www.3cx.com. Install the program and once installation is complete open the 3CX Phone application program. Figure 3.6(a) and Figure 3.6 (b) is representing the 3CX Phone and 3CX Logo respectively.

Figure 3.5 3CX Phone

Figure 3.6 3CX Logo

3.4.4 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 analog 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 Page | 29

IP connectivity for any phone and fax Web management for easy configuration and installation Offers traditional and advanced telephony features Portable and compact for use at home or on the road

Figure 3.7 Grand Stream HT 502

3.5 Integration of Hard ware

Integration is the next step after installation of Elastix server and soft phone now we are going to integrate ATA with our IP PBX server. Here we have following steps to integrate. Refer Figure 3.8 for the hardware connection of ATA Connect a standard touch-tone analog 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 analog 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 3.8 Integration

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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.

3.6 Implementing the features of Elastix server

3.6.1 Voice call
The voice call is the basic property of unified communication system, voice call is based on sip protocol. The communication allowed for those who are registered in sip server. The all communication devices can work on voice feature. the quality of sound is good.

3.6.2 Video call

Elastix server give great flexibility we can change the code of asterisk by pressing Asterisk File Editor in tools bar Manu. It is necessary to enable video calls we need to configure /etc/asterisk/ sip_general_custom.conf to: video support=yes, allow=h264, allow=h263 and allow-h263p.

3.6.3 Voice mail

Voicemail is configured to handle calls that can not be answered. Voicemail is generally made is to call user group. Flow chart of voicemail can be seen in Figure 3.9

Figure 3.9 Flow Chart Voicemail Page | 31

Personalized voicemail is a feature that allows callers to leave messages on phone. Voicemail permits to record user 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 so user know when user caller contacted. This is the great feature of Elastix server that is voice attachment to particular user to enable voice mail go to the extension profile to the user then enable status of voice mail.

3.6.4 Voicemail to Email Notification

As an optional feature you can to receive email notification. This can also be a text message to a cell phone or both. To enable email notification enters the email address in the extensions module on the line for email address. Voicemail password: the password of your voicemail Email Address: Email of a person who has that extension (it is recommended if you want to be informed through emails) Email Attachment: yes (attach your voice mail in email) Play CID: yes Enable Envelop: yes There are two features of voice mail first, when user want access the voice mail through user phone, Press *97 for accessing the user voice mail menu in which the operator tell user new and old voice mails. Second, when user want to access web based account then enter user extension and voice mail password. Now user have following figure for Email Notification in Figure 3.10

Figure 3.10 Email Notification Page | 32

3.7 Configuration
Elastix server configuration can be done via the web interface, it is very easy to configure. Configuration is carried out also in accordance with the purposes of the Sukkur IBA. The following is a configuration that has been done:

3.7.1 Configuring VoIP user

Configuring all VoIP users through Elastix 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 VoIP users for Elastix server. Figure 3.11 shows this scenario of configuration.

Figure 3.11 Configuration

3.7.2 Configuration of HT502 device

The ATA HT 502 is also web based So we need to give IP address to this device.

Figure 3.12 GUI of HT 502 Page | 33

Now to integrate telephone sets we have to give Elastix Sip server IP address 121.52.154.75. The results can be seen from the snapshot given in Figure 3.12 and Figure 3.13. And name and its telephone number

Figure 3.13 GUI HT 502 Same thing occur with FXS port2 but having different name and number Now we have to configure through Elastix server having same name and phone number so that we can access these telephone set through our server and implements features of server in telephone sets. After successful entry, view of the Elastix server can be seen in Figure, to create a user, select the PBX as Figure 3.14 then select SIP device and click submit. As our telephone set work with Sip protocol so we have to create sip based extension.

Figure 3.14 Configuration of Telephone Page | 34

Likewise we create for second telephone set. This process for creating extension will be same for 3cx soft phone, IP telephone and android cell phone.

3.7.3 Configuration of 3CX

After creating extension in Elastix server now our turn to create sip profile in 3cx when we click on create profile. After that we need to do account setting. The last stage is shown in Figure 3.15

Figure 3.15 Configuration of Soft phone Same process will be for android cell phone in which we have 3cx too.

3.7.4 Configuration of IP Telephone

GXP1400 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP account, a 128x40 graphical LCD, , and 3-way conference. The GXP1400 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 3.16 and Figure 3.17 showing the GUI and Configuration of IP phone respectively. To set up the GXP1400,follow the below step: There are slots at the back side of phone Connect the handset and main phone case with the phone cord Page | 35

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 5V DC output plug to the power jack on the phone ;plug the power adopter into an electrical outlet. The LCD will display update information about the IP address . Now use your keyboard and make configuration through GUI while entering the ip address in web browser

The device can be configured through given IP address in manual that is 192.168.208.158, when we enter IP address in web browser we have following figure.

Figure 3.16 GUI of IP phone The default password is admin when we hit login we have following figure

Figure 3.17 Configuration of IP phone Page | 36

In this IP telephone we have two Sip account first we make configuration for first account and same process will occur for 2nd account only name and number will be changed. Account Name: SaleemIPTEL SIP Server : 121.52.154.75 (Elastix server IP address) SIP user ID :3007 (same number will be given while creating extension for IP telephone)

3.7.5 Configuring Ring Group Ring group made was 3, which groups Technician, Marketing and Administration. From the ring group will later be connected to the IVR. So from IVR to continue input will do call the group made. Here is the configuration that has been done on the ring group. This can be seen in Figure 3.18.

Figure 3.18 Ring configuration group From image configuration can be seen that the group Technicians with extension number 100 has a member with the extension number 101, 103, 104, 105, 106, 107, 108, 110, and 999. And at the end of the configuration is the arrangement of the group technician if no one answers, the call will be transferred to the operator extension number 114 as shown in Figure 3.19. Page | 37

Figure 3.19 Forwarded call rings group For group marketing and administration can be made. While the configuration of call forward if not answered

3.8 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 IPPBX server, it automatically calls will be serviced by the IVR, then the user of IVR caller can make call, then the call will be transferred to the division or VoIP users in accordance with user needs. For more details can be seen in Figure 3.20

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End Figure 3.20 Flow Chart IVR system work (IVR) interactive voice response is said to 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. When user registered extensions, Elastix 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 didnt reply. And we should do as following: Call center, configuration of telephony system, follow me. We faced with this window in Figure 3.21. Page | 39

Figure 3.21 Ring Strategy Choose the extension user want to define this features. When user registered extensions, Elastix can be set to meet our needs. It is possible that user want the system automatically connected to the extension user already defined if user extension didnt reply. And user should do as following: Call center, configuration of telephony system, follow me Ring Strategy: dial the main number first and then the others. Extension list: 1102 is the deputy director and 1103 is the office assistant. Ring Time: 20 second Destination If no answer: terminated call-hang up Whenever dialed to the manager, dials the Asterisk extension number of the manager. If no one replied, the contact is with the extension of 11 and 22. And if no one answered again, Asterisk terminates the call. After finishing, choose the submit changes key and Apply Configuration changes here Note: There is main difference between call forward and follow me option that in call forward we have only one extension while in follow me option we have more than one extension available for attend call.

3.8.1 IVR Configuration

Before you configure the IVR menu, there is a need to do is upload a voice that will be used for the IVR announcement, here are ways that have been implemented: Click on the recording system menu, and then select the file to be used as a voice on the IVR, then click upload to upload files to the server and click Save to save the file with the name.

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3.8.2 Configure Inbound

Inbound used when an incoming call on line 1, line 2 and line 3. Means of inbound route incoming calls to the server. While the existing route incoming calls to the IVR server by hand. So for inbound only need to configure the line in as well as some sort of connection IVR automated answering machine on the server that the user can forward calls to the department / person of interest. Refer Figure 3.22 for configure Inbound Route.

Figure 3:22 Inbound route

3.9 Installation
Instant messaging with Openfire is a popular chat program and use Jabber/XMPP protocol for exchanging data. After installing this program you can have services such as Google talk, yahoo messenger and etc. name of client program installed in staff computers is SPARK which they will have these features by the configuration you did: Chat Exchanging the file Calling an extension by pressing a key You can send you current screen work Spark client has built in language translator

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3.9.1 Installing Openfire

After clicking the instant messaging tab (openfire) user will see this message, because it is not installed on Elastix as a default. Click the button click here and installation progress will start In first stage you should select the language.In this section you should fill the domain part with the name of Host or user IP server which is recommended dont change the name system recognized! You can change the console ports if you want but it is better to use default one. In next stage you should select how you want to be connected to database. The first one asks you a lot of question about connecting to the database which waste your time!! Select the second one and continue. In this stage it is asked that where do you want to store users item 2 and 3 is used when you want to store them in a directory server or clear space otherwise select Default. Determining an email for admin user and select a password for admin user (this password has nothing to do with your email), try to choose a password you can remember!! Because its retrieving is very difficult. Congratulation!! Your Openfire installed now by clicking on Login to the admin console go to the page of management. Enter with admin user and password you have selected in previous stage. Dont be sad! You are not supposed to change all the setting! And never try to update an Openfire, this program will be update with any new versions of Elastix. Updating manually may cause many problems so dont risk. Now we go to plug-in to install some add-ons (in this stage you need Internet, if you dont access, you need to download the add-ons and upload here). After clicking on available plug-in, lists of add-ones will appear. For installing the add-ons (Asterisk-IM Openfire plug-in) click on green sign (+), after installation, these add-ons will be added to the list of plug-in. Install these add-ons either: SIP Phone Plug-in, Presence Service, and IM gateway Enable Asterisk-IM and in change the Asterisk queue presence and drop-down device selection to Yes and save it. click on the IM tab to bring up OPENFIRE and then click on the ASTERISK-IM Tab Page | 42

and then click on Add Server hyperlink which will take us to the following screen. Click on Create Server and if everything is successful. If the green dot is actually grey, then you have correctly edited the file, but it appears that for some reason you have not correctly connected to the Elastix Server. This may be the result of the user and password not being set correctly for the Asterisk Management Interface. The ones that we have provided in this chapter are ones that are setup by default by Elastix. If you have changed manager config under /etc/asterisk, you will need to correct the login and password to suit. So if you have the green dot, you now have a working Openfire Server connected to your Elastix Server. All we need to do now is add users and install the client on the desktops. Click on users and groups tab at the top and the following screen will appear. Only the admin user will appear. We now need to create users for your system. Click on create new user on your system. And the following screen will appear. Fill in the details for each user you want to connect to Openfire. Keep the usernames in lower case (makes it easier), fill in their proper name, their current email address, and provide them with a password. This password does not have to match anything, it will be used by the client that resides on their desktop to connect to the Openfire server. Now you have a screen for creating user, Click on CREATE USER (or Create and Create another if you want to keep adding more). After you have done this, you should see the screen (we have only done one user) like this We have now setup one user on the system. For the system to recognize when we are on the phone, we need to map the user to an extension we do this by clicking again on the Asterisk-IM Plug-in Tab and then clicking on Phone Mappings on the left menu. We are now going to setup a phone mapping for the user that we just setup under Users and Groups. Username is the username you setup, in this case it was bob (remember I said to setup in lowercase, it just makes it easier, as the system will not recognize if you use an uppercase char, it sees it has a different login). The device is the actual phone, and you should be able to drop down the box and it will show extensions that you have in your Elastix System. If it doesnt show, then you can enter it manually (e.g. for our one user we would add SIP/301). Then add the Page | 43

extension number which is the same, without the SIP/, so we would enter 301 in here, and then a caller ID. I normally enter 301 in here as well. You can click on the primary field as well if you like, but it is not crucial. This does have a purpose, but it is for more complex systems, which are beyond the scope of this document.

Figure 3.23 Openfire You have now successfully mapped a user and phone together. Refer Figure 3.23

3.9.2 Install Spark Client.

As mentioned the SPARK Client from the same people that developed Openfire, is a good starting point. You can always change the client later, whenever user want, and by then, you will know what you are looking for in a client. user can download it from http://www.igniterealtime.org/projects/spark/index.jsp Install it as per default install instructions. And you should end up with the following screen.

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Figure 3.24 Spark Client Client has now successfully connected to the Openfire server. If we had more users installed, user would see the users listed, showing their presence status, whether they are offline, on the phone, away from keyboard etc. Initial window can be seen in Figure 3.24. If this is not what user want, and user want all the people that are on user local Network to be immediately available to each SPARK user, then user can set them into Groups (a subject we did not broach). If user go back to Open fire Users and Groups Tab, create a Group Name and add the selected users to the Group and they will be immediately available to communicate with if they are members of that group. There are many more features, and functions within Open fire. It deserves a book all by itself, which again is not the purpose of this document. We hopefully have provided enough to get you started, so that you can explore Elastix and the integrated Open fire server.

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3.10 Summary
In this chapter we discuss the architecture of project and its working principle which is comprised of two parts hardware design and software design. It based on server software and its client VoIP to complete the project. The complete and brief introduction of hardware components used in this project, specially Grand Stream HT 502 ATA, and for the phase selection we have two options like soft phone and soft switch, 3CX is the example of soft switch. . Elastix is open source software create a media platform unified communications or "Unified Communications Platform, "which consists of a component or module technologies commonly used communication media today such as: Voice call, video call ,voice mail, instant messaging, a fax server, VoIP and video conferencing. As Soft phone 3CX is chosen because 3CX have call forwarding features and call transfer required by any organization to connect either their employees or customers and As the 3CX soft phone uses SIP protocol and we have Elastix SIP server that can easily handle SIP protocol based soft phone. This process for creating extension will be same for 3cx soft phone, IP telephone and android cell phone. After creating extension in Elastix server now our turn to create sip profile in 3cx when we click on create profile. So for inbound only need to configure the line in as well as some sort of connection IVR automated answering machine on the server that the user can forward calls to the department / person of interest. For the system to recognize when we are on the phone, user need to map the user to an extension we do this by clicking again on the Asterisk-IM Plug-in Tab and then clicking on Phone Mappings on the left menu. If you go back to Open fire Users and Groups Tab, create a Group Name and add the selected users to the Group and they will be immediately available to communicate with if they are members of that group.

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CHAPTER 4

RESULTS AND DISCUSSION

4.1 System Testing Process

Steps taken in the testing of the system is user registration VoIP, VoIP calls fellow user, incoming calls from the laptops, analogue telephone , IP telephone Android based to a IP PBX server , call out of the user VoIP to analogue telephone, laptops ,IP phone and Mobile phone having the application of Android. For more details can be seen in Figure 4.1

Figure 4.1 Systems testing process Page | 47

4.1.1. Registration of VoIP user

In the process of using a soft phone VoIP user registration often have many problems. Problems can be identified through the report display on the screen soft phone. Here are some of the problems that have been encountered and the settlement that has been done. Figure 4.2 is an illustration of a VoIP user registration

Figure 4.2 Registration of user VoIP

Registration VoIP user using a PC / Laptop, analogue telephone Android HP, IP Phone. Registration VoIP user using a PC / Laptop is adding phone functionality on user PC / laptop that is by installing a soft phone application. 3CXPhone Soft phone is used. In the testing that has been done with some application soft phone, 3CX has major advantages compared to other soft phone programs, namely: Can do call forward and call transfer It has a great view Easy to understand the operation

In addition to these advantages, the process registration VoIP users there are some problems, this can occur for many reasons, here is a summary of the various problems registration common VoIP user. a) 400 Bad Request Requests cannot be understood by the server, the blame lies on the Soft phone SIP profile, complete reconfiguration on SIP soft phone.

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b) 401 Unauthorized Request requires user authentication, user authentication error on SIP soft phone profile, complete reconfiguration on SIP soft phone c) 403 Forbidden Requests can be understood by the VoIP server, but not biased implemented. The blame lies on the Soft phone SIP profile, complete reconfiguration on SIP soft phone d) 404 Not Found (User not found) Registration request cannot be accepted, because the user configuration in SIP VoIP server does not have the desired information by SIP Soft phone e) 407 Proxy Authentication Required Registration request cannot be accepted, because the proxy configuration on the soft phone user cannot find the proxy in question. f) 409 Conflict Users are requested VoIP SIP soft phone is already used by others, resulting in duplicate SIP user that caused the conflict. There are many SIP registration response has not been explained, because in making the report we have just write stuff ever experienced. As for the process registration using , analogue telephone , HP Android, and IP Phone also has the same SIP response. Because the process is not affected registration of equipment used to perform registration.

4.1.2 Calls fellow user VoIP

In the process of dialing phone Internet (VoIP user) did not experience a lot of problems, but with the provision that user is active / online at the time of the call. Problems often occur in the process of extension dialing SIP is an Internet connection. VoIP Users who have low connection more often fail in doing the calling and the called. It simply cannot be avoided, unless the user adds VoIP bandwidth used. Figure 4.3 is an illustration of sesame user VoIP calls. VoIP call is made from user 3 to user VoIP 5. The red path indicates the direction line call is made.

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Figure 4.3 Calls between VoIP users In the mobile phone used in the design process has a 3G network specifications and also features wireless devices. For VoIP connections using wireless devices connected to the network hotspot Sukkur IBA not experience any problems in the process of communicating, but if you are using a public hotspot access is free and more often have problems, because the bandwidth received by mobile phones is limited and there is interference from other devices connect to free hotspots are. In addition to using wireless devices, mobile phones can also be used to connect to the Internet using the 3G network. VoIP service quality that is used is also comparable to the quality of the operators used .

4.1.3 Incoming calls (Inbound)

Incoming calls to the server or inbound calls from outside is VoIP calls to the number Sukkur IBA is 500. Calls will be received and handled by the IVR (Intelligent Voice Response), then from the call will be transferred in accordance with the purposes of the caller. Page | 50

One of the identification when the inbound problem is when there is an incoming call, the call is not handled directly by the IVR, but only hear a ringing tone on the caller.

4.2 VoIP user capacity

A VoIP server would have limited VoIP users that can be served. Limitations can be divided based on two things, namely from the VoIP server PC specifications and also in terms of the bandwidth of the VoIP server [29].

4.2.1 VoIP server computer specs

The question that often arises is about the specifications of the PC to be used as a VoIP server. Based on information from the book edition of VoIP computer info: Telkom forerunner of the people, which was written by Mr. Onno W. Purbo mention that, in general asterisk need about 30 MHz CPU capability for each channel or user SIP enabled. Therefore the PC server with 1.8 GHz CPU speed is theoretically able to handle about 60 simultaneous VoIP user. According to these data, the specification of VoIP server used is sufficient required by SUKKUR IBA.

4.2.2 Bandwidth capacity

Bandwidth required by the active channel is determined by the codec used, the server and client codec used is PCMU/G.711u/alaw/ulaw. G.711 requires a minimum bandwidth for each channel is 64Kbps. So with bandwidth-owned (2MB) VoIP server can serve a maximum of 20 VoIP calls to 24 calls simultaneously. Figure 4.4 is a bandwidth which is owned by the VoIP server.

Figure 4.4 Bandwidth VoIP Server

4.3 Comparison between Xlite and 3CX.

3CXphone soft phone used is because the application contained 3cxphone call transfer facility that can be used free of charge, while the xlite to use call transfer facility is required to update to version eyebeam first, Table 4.1 shows the comparison between 3CXphone and Xlite[35]. Page | 51

Table 4.1 comparison between 3CXphone and Xlite NO

1 2 3 4 5

FACILITIES
Can be installed on Windows, Android Supports G.711 and GSM codecs Multiple line Record the conversation to the HDD Call transfer

3CXPHONE
Yes Yes Yes Yes Yes

XLITE
Yes Yes Yes Yes No

4.4 Analysis PCMU

PCMU is one of the transport protocol used in VoIP communications. Each protocol has its own advantages respectively, the difference in sound quality, delay and jitter is a distinguishing characteristic of the protocol. Analyses were conducted with the conditions of Internet bandwidth and server 2 megabytes of user using the default codec (G.711u). The data capture from the client and from the server.

4.4.1 Delay the call.

Delay the call in question is the delay between the call setup until just before ring back tone. The observation of each device can be seen in Figure 4:13 until 4:17.

4.5 QoS Measurement

Measurement of QoS parameters used are MOS (Mean Opinion Score) with G.711 codec and frame size 20ms packet. MOS values taken by the speaker and the listener satisfaction while holding a VoIP connection. MOS value in accordance with ITU-T can be seen in Table 4.2.

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Table 4.2 MOS values with G.711 codec based R factor

NO PACKET LOSS (%) 1 0 2 1 3 2 4 3 5 4 6 5 7 6 8 7 9 8 10 9 11 10 12 15 13 20 14 25 15 30 16 40 17 50

MOS SCORE 4:4 4:2 3:9 3:7 3:4 3:2 3 2:8 2:6 2:5 2:4 1:9 1:6 1:4 1:3 1:2 1:1

R-FACTOR 93 85 77 71 66 62 58 54 51 48 46 36 30 25 22 17 14

MOS Good Good Self Self Self Self Self Ugly Ugly Ugly Ugly Poor Poor Poor Poor Poor Poor

Sound Quality Clear Clear Less Clear Less Clear Less Clear Less Clear Less Clear Not Clear Not Clear Not Clear Not Clear It is not Clear It is not Clear It is not Clear It is not Clear It is not Clear It is not Clear

MOS value measurement made global QoS of the data capture results that have been implemented. If it is found the packet loss occurs, then the data packet is immediately analyzed further to determine the percentage of packet loss results. Packet loss is the number of lost data packets per second. Packet loss can be caused by a number of factors, including a decrease in the signal network media, limit network channels, the corrupted packets cannot be transmitted, and network hardware errors. Packet Loss of softphone, IP phone, Analogue phone and Mobile softphone can be seen in Figure 4.5(a), Figure 4.5(b), Figure 4.6(a) and Figure 4.6(b) Packet loss can be calculated by the formula: Packet loss = (packet data sent - received data packet) 100% packets of data sent Of packet loss will be obtained MOS values in accordance with table 4.2, but if the data capture packet loss is not found, then the general communication that has captured the MOS value 4:4. Below is a graph of the packet loss calculation has been done.

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Packet Loss Softphone

30.00 p a c k e t 25.00 20.00 15.00 10.00 5.00 0.00 0 2 13 17 25 148 1305 3298 3411 3433 3955 4280 6479 0%

Total Packet

Figure 4.5(a) Packet loss soft phone

Packet Loss IP Phone

1 0.8

P a 0.6 c k 0.4 e t 0.2

0

0%

17

22

48

52

59

69

91

107

119

148

573

Total Packet

Figure 4.5(b) Packet loss IP Phone Page | 54

Packet Loss Analogue Phone

1 0.8 P a 0.6 c k 0.4 e t 0.2 0 18 38 42 45 58 Total Packet 77 85 88 333

0%

Figure 4.6(a) Packet loss Analog telephone

Packet Loss Softphone Mobile

1.2 1 P a 0.8 c 0.6 k e 0.4 t 0.2 0 3 14 23 38 39 46 80 85 89 92 420

0%

Total Packet

Figure 4.6(b) Packet loss mobile soft phone Based on data graphs packet loss and MOS value data in table 4.2, the overall value of MOS based packet loose is 4:4, for all the scenarios that have been implemented are not found packet loss of more than 1%. Page | 55

4.6 Summary
Steps taken in the testing of the system is user registration VoIP, VoIP calls fellow user, incoming calls from the Android based Mobile phone to a VoIP server with IVR, call out of the user VoIP to IP phone and Mobile phone having the application of Android. Registration VoIP user using a PC / Laptop, HP Nokia, Blackberry, Android HP, IP Phone. Registration VoIP user using a PC / Laptop is adding phone functionality on your PC / laptop that is by installing a soft phone application. Requests cannot be understood by the server, the blame lies on the Soft phone SIP profile, complete reconfiguration on SIP soft phone Request requires user authentication, user authentication error on SIP soft phone profile, complete reconfiguration on SIP soft phone Registration request cannot be accepted, because the user configuration in SIP VoIP server does not have the desired information by SIP Soft phone Users are requested VoIP SIP soft phone is already used by others, resulting in duplicate SIP user that caused the conflict. In the process of dialing phone Internet (VoIP user) did not experience a lot of problems, but with the provision that user is active / online at the time of the call. For VoIP connections using wireless devices connected to the network hotspot Sukkur IBA not experience any problems in the process of communicating, but if you are using a public hotspot access is free and more often have problems, because the bandwidth received by mobile phones is limited and there is interference from other devices connect to free hotspots are. Limitations can be divided based on two things, namely from the VoIP server PC specifications and also in terms of the bandwidth of the VoIP server. 3CXphone soft phone used is because the application contained 3cxphone call transfer facility that can be used free of charge, while the xlite to use call transfer facility is required to update to version eyebeam first, Table 4.1 shows the comparison between 3CXphone and Xlite.

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CHAPTER 5 CONCLUSIONS AND RECOMADATIONS 5.1. Conclusions

This model can be implemented in the university campus to provide free voice and video calls. Its a most effective way to diminish the large phone call bills. The service is secured and allows only the registered user to place calls. Moreover, all the calls placed using the Asterisk Server are encrypted thereby avoiding hackers to intercept an ongoing phone calls. Asterisk based voice exchange provide us with a much better alternative solution. Its not only cost effective but also provides us with various features which we generally dont get with the conventional circuit switched based PBX. Moreover, the system also provides for unlimited expansion and since it runs on a secure operating system like Linux. Its much less prone to viruses, worms and hackers. SIP is less complex than other protocols. 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). .

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 SUKKUR IBA is 10 MB, then for VoIP-based voices is enough to meet. The greater the bandwidth provided, the smaller delay caused. Codec Codecs 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.

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5.2. Future Recommendations

5.2.1 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 . 5.2.2 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. 5.2.3 Integration with others Unified Communication Systems This project can be integrate with other unified communication systems like Cisco, astriskNow and many others by keeping same protocol. This project can be integrated with business telephone system. 5.2.4 Communication without side SIP network Compared with the general SIP server, it can be said that Asterisk is more focused on providing basic functions. But Asterisk can connect with SIP server easily, so it is possible to implement the necessary additional functions by just connecting with other outside SIP servers. 5.2.5 Telephony Interface Cards In the testing and analysis we can also use Telephony interface cards like PCI or PCI Express expansion cards that connect computers running Asterisk directly to legacy phone lines, phones and phone systems. The cards convert the legacy signaling and media into Asterisk's internal formats 5.2.6 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 .

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B.Chatras, and S.Garcin, Service drivers for selecting VoIP protocols, Proc.of Telecommunications Network Strategy and Planning Symposium, Vienna (Austria), Jun.2004, pp.131-136. Basicevic, M.Popovic, D.Kukolj, Comparison of SIP and H.323 Protocols, Proc.of The Third International Conference on Digital Telecommunications (ICDT08), Bucharest (Romania), Jul.2008, pp.162-167. L.Deri, Open Source VoIP Traffic Monitoring, Proc.of System Administration and Network Engineering (SANE 2006), Delft (The Netherlands), May.2006. Asterisk : http://www.asterisk.org/ Digium, USA, "Asterisk: The open source telephony project." [Online]Cited 2010-03-01.Available at:http: //www.asterisk.org. Telecommunication standardization sector of ITU (ITU-T), Packet based Multimedia communications systems, International Telecommunication Union (ITU), ITU-T Recommendation H.323, Jun.2006. Available: http://www.itu.int/rec/T REC-H.323/e. Elastix: http://www.elastix.org/ [Accessed 3rd July 2012] 3CX Soft phone: http://www.3cx.com/voip/3cxphone/ [Accessed 25th February 2013] Freebx: http://www.freepbx.org/ VoIP History: http://www.voipreview.org/news.details.aspx?nid=51 Extended version of this paper http://pubs.doc.ic.ac.uk/AsteriskCallManagementPolicy/AsteriskCallManage mentPol icy-extended.pdf International Telecommunications Union (ITU-T). Recommendation H.323 packet-based multimedia communications system, July 2003. http://www.itu.int/rec/T-RECH. 323-200307-I/en. Internet Engineering Task Force. RFC 3261 SIP: Session initiation protocol, 2002. http://www.ietf.org/rfc/rfc3261.txt. Internet Engineering Task Force. RFC 3550 RTP: A transport protocol for real-time applications, 2003. http://www.ietf.org/rfc/rfc3550.txt. Internet Engineering Task Force. RFC 3711 the secure real-time transport protocol (SRTP), 2004. http://www.ietf.org/rfc/rfc3711.txt. Page | 62

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[43] [44]

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Appendix-A: List of Abbreviations

Abbreviations ATA ATM Full Form A Analogue Telephone Adapter Automated teller Machine B Basic Input / Output System Berkeley Software Distribution C Chinas Chip Design Industry Customer Premise Equipments Customer Relationship Management D Direct connection Long Distance Department Of Defense Disk Operating System Digital Signaling Processing Domain Name Service Direct Inward Dial Digium/Asterisk Hardware Device Interface F Foreign Exchange Office (Port) Foreign Exchange Station (Port) Fixed Wireless Terminal G General Switched Telephone Network Graphical User Interface Global System for Mobile (Communication) General Packet Radio Service I International Direct Dialing Interactive Voice Response Internet Low Bit rate Codec Internet Engineering Task Force Page | 63

BIOS BSD

CCID CPE CRM

DLD DOD DOS DSP DNS DID DAHDI

FXO FXS FWT

GSTN GUI GSM GPRS

IDD IVR iLBC IETF

Abbreviations ICQ ISC

Full Form Internet Chat Query International Soft switch Consortium M Metropolitan Area Network Media Gateway Controller Mean Opinion Score N Next Generation Network Network Address Translation P Private Branch Exchange Public Switched Telephone Network Public Land Mobile Network Pulse Code Modulation MEO-Law Pulse Code Modulation A-law R Request for Command (document) Resource Reservation Protocol Real Time Protocol Real Time Control Protocol Remote User Multiplex S Session Initiation Protocol SIP Uniform Resource Locator Session Distribution Protocol Signaling System 7 T Transmission Control Protocol Technical Image Press Association Trivial file Transfer Protocol V Voice Over Internet Protocol

MAN MGC MOS

NGN NAT

PBX PSTN PLMN PCMU PCMA

RFC RSVP RTP RTCP RUM

SIP SIP URL SDP SS7

TCP TIPA TFTP

VoIP

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Appendix-B: Glossary
A Analog audio signals Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the pitch of the sound. The same technology is used for radio wave transmissions. Analogue Telephone Adapter ATA or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup. B Bandwidth Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths C Call A call is an informal term that refers to some communication between peers, generally set up for the purposes of a multimedia conversation. Client A client is any network element that sends SIP requests and receives SIP responses. Clients may or may not interact directly with a human user. User agent clients and proxies are clients. Codec Codec is a term that arises from the Compressor-Decompressor or enCOder/DECoder process. It is used for software or hardware devices that can convert or transform a data stream. For instance, at the transmitting end codecs can encode a data stream or data signal for easy transmission, storage or encryption. At the receiving end, they can decode the signal in the appropriate form for viewing. They are most suitable for videoconferencing and streaming media solutions. Compression This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files. Page | 65

D Domain Name Server A computer program running on a web server, translating domain names into IP addresses. In the last years special types of domain names records were added to the DNS world-wide system, which provide support to SIP/VoIP (SRV/NAPTR, ENUM)

F Find-me/follow-me A feature that allows calls to find you wherever you are, ringing multiple phones (such as your cell phone, home phone, and work phone) all at once. H H.261 It is a video codec use for wideband (>= 64Kbps). H.263 is used for narrowband (< 64-kbps). Both are widely supported. H.264 It is a newer narrowband codec that produces higher-quality results than H.263 and is recommended in its place. H.264 is also known as ISO 14496-10 and as MPEG-4 part 10 and as MPEG-4 AVC (Advanced Video Coding). I Instant Messaging IM, which stands for Instant Messenging, is a software that allows users to exchange messages in real time. However, to do so both the users must be logged on to the instant messaging service at the same time. Some of the popular IM services are: MSN Messenger, AOL Instant Messenger, Yahoo! Messenger, Google Talk and ICQ. Interactive Voice Response In computer telephony, Interactive Voice Response is a horizontal application wherein computer-based information is accessed over the phone - with a telephone versus a computer. An IVR platform uses computer telephony components to translate callers' touch-tones or voice commands into computer queries after the callers hear an audio menu. Internet Protocol IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.

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International Telecommunication Union ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications. J Jitter It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses. K Kbps Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second. L Latency Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks. M Mean Opinion Score A measurement of the subjective quality of human speech, represented as a rating index. MOS is derived by taking the average of numerical scores given by juries to rate quality and using it as a quantitative indicator of system performance. Message Data sent between SIP elements as part of the protocol. SIP messages are either requests or responses.

P Packet A logically grouped unit of data. Packets contain a payload (the information to be transmitted), originator, destination and synchronizing information. The idea with packets is to transmit them over a network so each individual packet can be sent along the most optimal route to its. Packets are assembled on one end of the communication and re-assembled on the receiving end based on the header addressing information at the front of each packet. Routers in the network will store Page | 67

and forward packets based on network delays, errors and re-transmittal requests from the receiving end. Packet loss Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data. Private Branch Exchange Private Branch Exchange or PABX (Private Automatic Branch Exchange). In telephony, a PBX system behaves as a customer's premises over trunk lines (thus the term "branch"). At first, PBXs mimicked a small telephone company switchboard. Users would use an operator to take and make telephone calls to and from the PSTN (Public Switched Telephone Network). Over time, users were able to dial directly, without the use of an operator. Today, computer telephony platforms such as automated attendants are able to route incoming calls automatically, too. Peer-to-Peer (P2P) The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-topeer network does not work on the traditional client-server model but on equal peer nodes that work both as "clients" and "servers" to other nodes on the network. Protocol It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details. Public Switched Telephone Network Public Switched Telephone Network. The combination of local, long-distance, and international R Redirect Server A redirect server is a user agent server that generates 3xx responses to requests it receives, directing the client to contact an alternate set of URIs. Request A SIP message sent from a client to a server, for the purpose of invoking a particular operation. Response A SIP message sent from a server to a client, for indicating the status of a request sent from the client to the server. Page | 68

Ring back Ring back is the signaling tone produced by the calling party's application indicating that a called party is being alerted (ringing). S Session Initiation Protocol An Internet Engineering Task Force (IETF) standard for initiating, maintaining, and terminating an interactive user session involving video, voice, chat, gaming, virtual reality, and more. SIP phone A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet (for signaling (and uses RTP for media)). The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are (normally) no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone. Server A server is a network element that receives requests in order to service them and sends back responses to those requests. Examples of servers are proxies, user agent servers, redirect servers, and registrars Skype Skype is a peer-to-peer Internet telephony company that revolutionized the way voice calls are made by using VoIP technology. The company, which has been acquired by eBay, was founded by Niklas Zennstrm and Janus Friis. Skype users can speak to other Skype users for free, but have to pay a small fee for calling or receiving calls from conventional phones. Soft phone IP telephony software that lets users send and receive calls from non-dedicated hardware, such as a PC or Pocket PC device. It is typically used with a headset and microphone. Soft switch It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks. T Transmission Control Protocol Transmission Control Protocol. The transport layer protocol developed for the ARPAnet which comprises layers 4 and 5 of the OSI model. TCP controls sequential data exchange in TCP/IP for remotely hosts in a peer-to-peer network. Page | 69

Telephony Taken from Greek root words meaning "far sound", telephony is the discipline of converting or transmitting voice or other signals over a distance, and then reconverting them to an audible sound at the far end. U UNIX A multi-user, multi-tasking operating system originally developed in 1969 by Ken Thompson of AT&T Bell Laboratories. UNIX is used in telephone company and mission critical applications. V Video encoding There are fewer video codecs (than audio codecs) associated with the H.323 and SIP protocol suites (thankfully). Voice over Internet Protocol The process of making and receiving voice transmissions over any IP network. IP networks include the Internet, office LANs, and private data networks between corporate offices. The main advantage of VoIP is that users can connect from anywhere and make phone calls without incurring typical analog telephone charges, such as for long-distance calls

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