Sobhasaria Engineering Collrge

SEMINAR REPORT

“INTERNET PROTOCOL TELEPHONY”

SUBMITTED IN PARTIAL FULFILLMENT OF THE

REQUIREMENT FOR THE AWARD OF THE DEGREE

bachelor of engineering

electronics and communication engineering

GUIDED BY: - SUBMITTED BY: -

MR. TOUSIF KAMAAL NARENDRA BAGORIA

LECTURER (E.C.E DEPTT.) FINAL YEAR (E.C.E.)

Department of electronics & communication engineering

Sobhasaria engineering college, sikar

University of Rajasthan

2007-2008

Sobhasaria engineering college, sikar

Department of electronics & communication engineering

CERTIFICATE

THIS IS TO CERTIFY THAT THE WORK, WHICH IS BEING PRESENTED IN THE SEMINAR ENTITLED “INTERNET PROTOCOL TELEPHONY” SUBMITTED BY MR. NARENDRA BAGORIA, A STUDENT OF FINAL YEAR B.E. IN ELECTRONICS & COMMUNICATION ENGINEERING AS A PARTIAL FULFILLMENT FOR THE AWARD OF DEGREE OF BACHELOR OF ENGINEERING IS A RECORD OF STUDENT’S WORK CARRIED OUT UNDER MY GUIDANCE AND SUPERVISION.

THIS WORK HAS NOT BEEN SUBMITTED ELSEWHERE FOR THE AWARD OF ANY OTHER DEGREE.

DATE: (MR TOUSIF KAMAAL)

PLACE: S.E.C., SIKAR (SEMINAR GUIDE)

(MR PRADEEP SHARMA) (PROF. K. B.SINGH)

(SEMINAR INCHARGE) (H.O.D OF E.C.E. DEPTT.)

Candidate’S declaration

THIS IS TO CERTIFY THAT WORK, WHICH IS BEING PRESENTED IN THE SEMINAR ENTITLED “INTERNET PROTOCOL TELEPHONY” SUBMITTED BY UNDERSIGNED STUDENT OF FINAL YEAR B.E. IN ELECTRONICS & COMMUNICATION ENGINEERING IN PARTIAL FULFILLMENT FOR AWARD OF DEGREE OF BACHELOR OF ENGINEERING IS A RECORD OF MY OWN WORK CARRIED OUT UNDER THE GUIDANCE AND SUPERVISION OF MR.TOUSIF KAMAAL, LECTURER, DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING.

THIS WORK HAS NOT BEEN SUBMITTED ELSEWHERE FOR THE AWARD OF ANY OTHER DEGREE.

DATE:- / /2008 NARENDRA BAGORIA

PLACE: S.E.C.,SIKAR FINAL YEAR, E.C.E.

Acknowledgement

The work written in this report is an outcome of the precious guidance cooperation of some persons. It is moment of great pleasure to acknowledge their help and encouragement.

The work has been performed under the guidance of Mr. Tousif Kamaal, Lecturer(E.C.E. deptt.). Words are incapable to formulate my deep sense of gratitude to him for his keen interest and encouragement. He was more willing to share his treasure of knowledge with me. I appreciably acknowledge his helpful comment for the improvement of the work.

I am cordially thankful to Mr. Pradeep Sharma for providing me the opportunity to present my seminar on a topic of my area of interest.

Last but not the least I am also grateful to Prof. K.B.Singh, HOD, E.C.E. deptt., for his guidance and kind support throughout.

NARENDRA BAGORIA

Final year,E.C.E.

ABSTRACT

A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S. and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee.

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user. Much of IP telephony focuses on that challenge.

IP telephony service providers include or soon will include local telephone companies, long distance providers such as AT&T, cable TV companies, Internet service providers (ISPs), and fixed service wireless operators. IP telephony services also affect vendors of traditional handheld devices.

INDEX

Page No.

Chapter 1 Introduction 1

1.1 Internet Protocol Telephony 1

1.1.1 Principles of IP telephony 2

1.2 VoIP 3

1.2.1 Resources from around the Web 4

Chapter 2 Implementation 7

2.1 Implementation 7

2.2 Reliability 8

2.3 Quality of service 8

2.4 Difficulty with sending faxes 9

2.5 Emergency calls 10

2.6 IP telephony scenarios 10

2.6.1 PC to PC 11

2.6.2 PC to Telephone 11

2.6.3 Telephone to PC 12

2.6.4 Telephone to Telephone 12

2.7 Integration into global telephone number system 13

2.8 VoIP phone accessibility and portability 13

2.9 Mobile phones & Hand held Devices 14

2.10 Security 15

Chapter 3 H.323 Protocols 16

3.1 H.323 Protocols 16

3.2 H.323 Architecture components 17

Chapter 4 Adoption 19

4.1 Mass-market telephony 19

4.2 Corporate and Telco use 20

4.3 Uses in Amateur Radio 21

4.4 Click to call 22

Chapter 5 Legal issues in different countries 23

5.1 IP telephony in Japan 24

5.2 Technical details 25

5.3 Applications 27

Conclusion 29

References 30

Appendix: List of figures 32

Chapter: 1

Introduction

1.1 Internet Protocol Telephony

Telephony is the technology associated with the electronic transmission of voice, fax, or other information between distant parties using systems historically associated with the telephone, a handheld device containing both a speaker or transmitter and a receiver. With the arrival of computers and the transmittal of digital information over telephone systems and the use of radio to transmit telephone signals, the distinction between telephony and telecommunication has become difficult to make.

In the other words the Internet Protocol (IP) is the method or protocol by which data is sent from one computer to another on the Internet. Each computer (known as a host) on the Internet has at least one IP address that uniquely identifies it from all other computers on the Internet. When you send or receive data (for example, an e-mail note or a Web page), the message gets divided into little chunks called packets. Each of these packets contains both the sender's Internet address and the receiver's address. Any packet is sent first to a gateway computer that understands a small part of the Internet. The gateway computer reads the destination address and forwards the packet to an adjacent gateway that in turn reads the destination address and so forth across the Internet until one gateway recognizes the packet as belonging to a computer within its immediate neighborhood or domain. That gateway then forwards the packet directly to the computer whose address is specified.

On the Internet, three new services are now or will soon be available:

· The ability to make a normal voice phone call (whether or not the person called is immediately available; that is, the phone will ring at the location of the person called) through the Internet at the price of a local call.

· The ability to send fax transmissions at very low cost (at local call prices) through a gateway point on the Internet in major cities.

· The ability to send voice messages along with text e-mail.

Some companies that make products that provide or plan to provide these capabilities include: IDT Corporation (Net2Phone), Netspeak, NetXchange, Rockwell International, VocalTec, and Voxspeak. Among uses planned for Internet phone services are phone calls to customer service people while viewing a product catalog online at a Web site.

Fig.1.1 An overview of IP Telephony

A Telephony API (TAPI) is available from Microsoft and Intel that allows Windows client applications to access voice services on a server and that interconnects PC and phone systems. Both Microsoft and Netscape provide or plan to provide support for voice e-mail.

1.1.1 Principle of IP telephony

To understand IP telephony, it’s necessary to be familiar with the fundamental characteristics behind the Internet and how it compares to the Public Switched Telephone Network (PSTN). The most important of these characteristics is the data transport mode, also known as data connection type which is either a circuit switched or packet switched as explained below:

Circuit Switched Connection: A device using a circuit switched connection only connects when data is to be sent. The connection is dedicated exclusively to the sending and receiving nodes for the entire duration of the call. Because the two points are connected in both the directions, the connection is called a circuit. The connection is only present when you need it and, since bandwidth remains constant, you only pay for the duration of the connection. While connected on a circuit switched network you have exclusive use of the established connection and data can be sent continuously. This type of data transaction is typically routed through the PSTN. Although the circuit switched network pro vides a very reliable connection for voice transmissions, it makes very inefficient use of the available bandwidth.

Packet Switched Connection: While circuit switched connection is open and constant for the entire duration of the call, packet switched connection opens just long enough to send a small chunk of data, called a packet, from one system to another. A packet switched connection keeps you connected all the time but you only pay for the amount of data transferred. In this case, the data is divided into small packets and each packet contains a source and a destination address. Packets of data are sent from source to destination using the quickest route available. The network bandwidth is shared and multiple simultaneous users are allowed to access multiple locations across a network. This provides for much more efficient use of available bandwidth but can create problems for voice traffic, which is very sensitive to delay.

1.2 VoIP

VoIP (voice over IP) is an IP telephony term for a set of facilities used to manage the delivery of voice information over the Internet.VoIP involves sending voice information in digital form in discrete packets rather than by using the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service.

VoIP derives from the VoIP Forum, an effort by major equipment providers, including Cisco, Vocal Tec, 3Com, and Netspeak to promote the use of ITU-T H.323, the standard for sending voice (audio) and video using IP on the public Internet and within an intranet. The Forum also promotes the user of directory service standards so that users can locate other users and the use of touch-tone signals for automatic call distribution and voice mail.

In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure that packets get delivered in a timely way. Using public networks, it is currently difficult to guarantee Quality of Service (QoS). Better service is possible with private networks managed by an enterprise or by an Internet telephony service provider (ITSP).

A technique used by at least one equipment manufacturer, Adir Technologies (formerly Netspeak), to help ensure faster packet delivery is to use ping to contact all possible network gateway computers that have access to the public network and choose the fastest path before establishing a Transmission Control Protocol (TCP) sockets connection with the other end

Fig.1.2 A typical analog telephone adapter for connecting an ordinary phone to a VoIP network

Using VoIP, an enterprise positions a "VoIP device" at a gateway. The gateway receives packetized voice transmissions from users within the company and then routes them to other parts of its intranet (local area or wide area network) or, using a T-carrier system or E-carrier interface, sends them over the public switched telephone network.

1.1.1 Resources from around the Web

Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network – see attached image - to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to PSTN may have a cost that's borne by the VoIP user.

There are two types of PSTN to VoIP services: DID (Direct Inward Dialing) and access numbers. DID will connect the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user. Access numbers are usually charged as a local call to the caller and free to the VoIP user while DID usually has a monthly fee. There are also DIDs that are free to the VoIP user but chargeable to the caller.

Fig.1.3 A complete VoIP solution

These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrase "Internet Phone" or "Digital Phone" which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the provider touts the advantage of being able to keep one's existing phone number.

At the other extreme are services like Gizmo Project and Skype which rely on a software client on the computer in order to place a call over the network, where one user ID can be used on many different computers or in different locations on a laptop. In the middle lie services which also provide a telephone adapter for connecting to the broadband connection similar to the services offered by broadband providers (and in some cases also allow direct connections of SIP phones) but which are aimed at a more tech-savvy user and allow portability from location to location. One advantage of these two types of services is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost.