A SEMINAR REPORT ON “INTERNET PROTOCOL TELEPHONY”

SUBMITTED IN PARTIAL FULFILLMENT OF THE REQUIREMENT FOR THE AWARD OF THE DEGREE OF BACHELOR OF ENGINEERING IN ELECTRONICS AND COMMUNICATION ENGINEERING

GUIDED BY: MR. TOUSIF KAMAAL LECTURER (E.C.E DEPTT.)

SUBMITTED BY: NARENDRA BAGORIA 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: PLACE: S.E.C., SIKAR

(MR TOUSIF KAMAAL) (SEMINAR GUIDE)

(MR PRADEEP SHARMA) (SEMINAR INCHARGE)

(PROF. K. B.SINGH) (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 FINAL YEAR, E.C.E.

PLACE: S.E.C.,SIKAR

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 circuitswitched 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 1.2 Internet Protocol Telephony 1.1.1 Principles of IP telephony VoIP 1.2.1 Resources from around the Web

1 1 2 3 4 7 7 8 8 9 10 10 11 11 12 12 13 13 14 15 16 16 17 19 19 20 21 22 23 24 25 27 29 30 32

Chapter 2

Implementation
2.1 2.2 2.3 2.4 2.5 2.6 Implementation Reliability Quality of service Difficulty with sending faxes Emergency calls IP telephony scenarios 2.6.1 PC to PC 2.6.2 PC to Telephone 2.6.3 Telephone to PC 2.6.4 Telephone to Telephone Integration into global telephone number system VoIP phone accessibility and portability Mobile phones & Hand held Devices Security

2.7 2.8 2.9 2.10

Chapter 3

H.323 Protocols
3.1 3.2 H.323 Protocols H.323 Architecture components

Chapter 4

Adoption
4.1 4.2 4.3 4.4 Mass-market telephony Corporate and Telco use Uses in Amateur Radio Click to call

Chapter 5

Legal issues in different countries
5.1 5.2 5.3 IP telephony in Japan Technical details Applications

Conclusion References Appendix: List of figures

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

through a gateway point on the Internet in major cities.

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.

1.2VoIP
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 circuitcommitted 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. 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.

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. Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations. VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones.

Chapter: 2 Implementation
2.1 Implementation
Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round-trip propagation delay (400–600

milliseconds for links through geostationary satellites). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This function is usually accomplished by means of a jitter buffer in the voice engine. Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse firewalls involve using protocols such as STUN or ICE. VoIP challenges:
• • • • • • • •

Available bandwidth Network Latency Packet loss Jitter Echo Security Reliability In rare cases, decoding of pulse dialing

Many VoIP providers do not decode pulse dialing from older phones. The VoIP user may use a pulse-to-tone converter, if needed. The principal cause of packet loss is congestion, which can sometimes be managed or avoided. Carrier VoIP networks avoid congestion by means of teletraffic engineering. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio, although this increases delay. This avoids a condition known as buffer under run, in which the voice engine is missing audio since the next voice packet has not yet arrived. Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.

2.2

Reliability

Conventional phones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages in the absence of an uninterruptible power supply or generator. Early adopters of VoIP may also be users of other phone equipment, such as PBX and cordless phone bases that rely on power not provided by the telephone company. Even with local power still available, the broadband carrier itself may experience outages as well. While the PSTN has been matured over decades and is typically reliable, most broadband networks are less than 10 years old, and even the best are still subject to intermittent outages. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T-1 connection.

2.3

Quality of Service
Proved call quality calculation and a variety of other applications. Some

broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary dropout of voice. This is more noticeable in highly congested networks and/or where there are long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on. It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable. A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics

block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer. RFC3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. RFC3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.

2.4

Difficulty with Sending Faxes
The support of sending faxes over VoIP is still limited. The existing voice codecs

are not designed for fax transmission. (They are designed to digitize an analog representation of a human voice efficiently, but the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax “sounds” simply doesn’t fit in the VoIP channel.) An effort is underway to remedy this by defining an alternate IP-based solution for delivering faxover-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as a message switching system, which does not need a real-time data transmission—such as sending a fax as an email attachment (see Fax) or remote printout (see Internet Printing Protocol). The end system can completely buffer the incoming fax data before displaying or printing the fax image.

2.5

Emergency Calls
The nature of IP makes it difficult to locate network users geographically.

Emergency calls, therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the US, at least one major

police department has strongly objected to this practice as potentially endangering the public. Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way. Following the lead of mobile phone operators, several VoIP carriers are already implementing a technical workaround. For instance, one large VoIP carrier requires the registration of the physical address where the VoIP line will be used. When you dial the emergency number for your country, they will route it to the appropriate local system. They also maintain their own emergency call center that will take non-routable emergency calls (made, for example, from a software-based service that is not tied to any particular physical location) and then will manually route your call after learning your physical location. E911 is another method by which VoIP providers in the US are able to support emergency services. The e911 emergency-calling system automatically associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999 and is being successfully used by many VoIP providers to provide physical address information to emergency service operators.

2.6

IP Telephony Scenarios
The IP telephony usage scenarios, as shown in Figure 1, are commonly classified

by the type of devices terminating an IP call. Because there may be either a PSTN device (e.g. telephone) or a data-oriented terminal (e.g. personal computer) on each side of a call, there are four generic classes as below: 2.6.1 PC-To-PC PC-to-PC communication can be provided for multimedia PCs (i.e. Personal Computers with a microphone, speaker and a sound card) operating over an IP-based network without connecting to the PSTN. PC applications (and IP-enabled telephones) can communicate using point-to-point or multipoint sessions. This set up requires that parties be equipped to talk at the time of the call. This class is attractive especially for private users who already have an Internet access and a multimedia PC. Necessary software is available from several comp anises for free or at a very low cost. There is usually no charge for PC-to-PC calls, except for the cost of Internet access. The user

doesn’t even have to pay for long-distance calls. This pure-IP scenario can also take advantage of integration with other Internet services, such as instant messaging, video conferencing, etc.

Fig.2.1 PC to PC telephone call

2.6.2

PC-To-Telephone The PC-to-Telephone method allows a user to call any ordinary telephone on a

PSTN from his computer. In this case, a gateway converting the IP call into a PSTN call has to be used. The gateway is required to be located as near to the called party as possible, to minimize the price for the gateway-to-called party connection. The call is converted to a PSTN call at the gateway and is then sent over PSTN to its destination. Like PC-to-PC calling; this scenario requires a software client. The software is usually free, but the caller may have to pay a small per-minute charge to a gateway operator. The cost charged by the operator is determined mainly by the cost of the call placed from the gateway to the called party. This solution is commercially available from Net2Phone, Phone serve and many other companies. 2.6.3 Telephone -To-PC The Telephone-to-PC method allows a user to call from any ordinary telephone on a PSTN to a PC connected to the Internet. In this case, a gateway converting the PSTN call into an IP call has to be used, and the gateway is required to be located as near to the caller as possible. The call is converted to an IP call at the gateway. The voice data then

“Hops on” the Internet and finds the PC on the other end by using the unique IP address. A few companies are providing special numbers or calling cards that allow a standard telephone user to initiate a call to a computer user. The caveat is that the computer user must have the vendor's software installed and running on his computer. This solution may require user to pay local call charges, in addition to small per-minute charge to a gateway operator. 2.6.4 Telephone -To- Telephone The Telephone-to-Telephone communication appears like a normal telephone to the caller but may actually consist of various forms of voice over packet network, all Interconnected to the PSTN. In this scenario, a caller dials into a gateway using a regular telephone. The call is converted to an IP call at the gateway and the voice data “hops on” the Internet. At the end point the voice data hits another gateway and “hops off” the Internet. The voice data is converted back to PSTN format and sent over the PSTN to its destination. This class is attractive for those who want to save on long-distance call and do not want to use PC. Since the call has to pass through two gateways – PSTN-toInternet and Internet-to-PSTN, the cost is charged by both the gateway operators. In addition, the user may have to pay local call charges. This solution is commercially available from many companies, offering discounted rates for long distance IP telephony calls.

Fig.2.2 Phone to Phone call via IP

2.7

Integration into Global Telephone Number System

While the wired public switched telephone network (PSTN) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be used for VoIP as well as incoming and external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider-specific short codes.

2.8

VoIP Phone Accessibility and Portability
If using a software based soft-phone, calls can only be placed from the computer

on which the soft-phone software resides. Thus with a soft-phone the caller is typically limited to a single point of calling. When using a hardware based VoIP phone-device/ phone-adapter it is possible to connect traditional analog phones directly to a VoIP phone-adapter without the need to operate a computer. The converted analog phone signal can then be connected to multiple house phones or extensions, just as any traditional phone company signal can be connected. A second VoIP hardware configuration option involves the use of a specially designed VoIP telephone which incorporates a VoIP phone adapter directly into the phone itself, and which also does not require the use of a computer. A third VoIP hardware configuration option involves the use of a WiFi router and a WiFi SIP phone which can extend a service range throughout a home or office. WiFi SIP phones can also be used at any location where an "unauthenticated" open hotspot Wi-Fi signal is available. However, note that many hotspots require browser-based authentication, which most SIP phones do not support.

2.9

Mobile Phones & Hand Held Devices
Telcos and consumers have invested billions of dollars in mobile phone

equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G networks.

However, "dual mode" telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular. Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network and therefore most VoIP calls from such devices are done over WiFi. Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available. Another addition to hand held devices are ruggedized bar code type devices that are used in warehouses and retail environments. These types of devices rely on "inside the 4 walls" type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.

2.10 Security
Many consumer VoIP solutions do not support encryption yet, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. [9] There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications, however such security through obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a consumer level. The existing secure standard SRTP and the new ZRTP protocol are available on Analog Telephone Adapters (ATAs) as well as various soft phones. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider.

Chapter: 3 H.323 Protocols
3.1 H.323 Protocols
The H.323 is an umbrella recommendation from the Telecommunication Standardization Sector of the International Telecommunications Union (ITU-T) that specifies the components, protocols and procedures that provide multimedia communication services (real-time voice, video, chat, whiteboard, file sharing, etc.) over packet–based networks, including IP. By conforming to H.323 standards, multimedia products and applications from different vendors can interoperate across IP based networks, including the Internet. H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia communication services over a variety of networks. H.323 covers both protected and unprotected connections. Control and data information requires a protected transmission to prevent packets from being lost or not received in the right order. In IP-based networks, TCP protocol guarantees an error-free

transmission in the right order but causes delays and has a lower throughput. Therefore, unprotected connections are used for audio and video transmissions, which are more efficient. H.323 covers both protected and unprotected connections. Control and data information requires a protected transmission to prevent packets from being lost or not received in the right order. In IP-based networks, TCP protocol guarantees an error-free transmission in the right order but causes delays and has a lower throughput. Therefore, unprotected connections are used for audio and video transmissions, which are more efficient. The H.323 standard's mandatory components are transmission of audio, connection control according to Q.931, communication with the gatekeeper over the RAS protocol, and use of the H.245 signaling protocol; the rest of the text, including coverage o f the ability to transmit video and data, is optional. Although H.323 uses TCP to carry the signaling channels, the real-time media streams are transported on RTP/RTCP (discussed earlier). RTP carries the actual media and RTCP carries status and control information. Being the first widely available VoIP protocol, H.323 enjoyed a head start as developers implemented it as toll-bypass systems as well as PC-to-phone and videoconferencing applications. The best-known H.323 application was Microsoft NetMeeting.

3.2

H.323 Architecture Components
The H.323 standard specifies a number of components (entities), which, when

networked together, provide the point-to-point and point-to-multipoint multimedia communication services. Some components are mandatory, while others are optional.

Fig.3.1 H.323 VoIP gateway architecture

The five most important components are listed below: Terminal: An H.323 terminal is an endpoint on a network which provides two-way communications with another terminal, gateway or a Multipoint Control Unit (MCU). An H.323 terminal can either be a personal computer or a stand-alone device such as IP telephone running H.323 and the multimedia applications. It supports audio communications and can optionally support video or data communications. Gatekeeper: A gatekeeper provides basic admission control onto a network by allowing or refusing communications between other H.323 entities within its zone of control. They also provide call-control services for H.323 endpoints, such as address translation (to use name instead of IP address), authentication, accounting and bandwidth management. Gatekeepers in H.323 networks are optional. Multipoint Control Unit (MCU): An MCU provide services that allow three or more endpoints to take part in a conference call. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video coder/decoder (codec) to use, and may handle the media stream. H.245 Control Signaling: H.245 control signaling is used to exchange end -to-end control messages governing the operation of the H.323 endpoint. The messages carried include messages to exchange capabilities of terminals and to open and close logical channels. The H.245 control messages are carried over H.245 control channels.

RTP: Real-Time Transport Protocol (RTP) provides end-to-end network transport functions for applications transmitting real-time data over IP networks. It provides services such as payload type identification, sequence numbering, time-stamping, and delivery monitoring to real-time applications. RTP can also be used with other transport protocols.

Chapter: 4 Adoption
4.1 Mass-Market Telephony
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. 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 phrases "Internet Phone", "Digital Phone" or "Soft 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. No additional charges are incurred, as call diversion via the PSTN would, and the called party does not have to pay for the call. For example, if a subscriber with a home phone number in the U.S. or Canada calls someone else within his local calling area, it will be treated as a local call regardless of where that person is in the world. Often the user may elect to use someone else's area code as his own to minimize phone costs to a frequently called long-distance number. For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a number of inconveniences compared to traditional services. VoIP requires a broadband Internet connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a power failure, VoIP services will generally not function. Additionally, a call to an emergency services number may not automatically be routed to the nearest local emergency dispatch center. Some VoIP providers only handle emergency call for one country. Some VoIP providers offer users the ability to register their address so that emergency services work as expected.

Another challenge for these services is the proper handling of outgoing calls from fax machines, DVR boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voicegrade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without any problems, but in other cases they will not go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western Europe. The TestYourVoIP Web site offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN.

4.2

Corporate and Telco use
Although few office environments and even fewer homes use a pure VoIP

infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect switching stations, converting voice signals to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes. Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations. VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones. Many telecommunications companies are looking at the IP Multimedia Subsystem (IMS) which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and mobile phones.

Electronic Numbering (ENUM) uses standard phone numbers (E.164), but allows connections entirely over the Internet. If the other party uses ENUM, the only expense is the Internet connection. Virtual PBX (or IP PBX) allows companies to control their internal phone network over an existing LAN and server without needing to wire a separate telephone network. Users within this environment can then use standard telephones coupled with an FXS, IP Phones connected to a data port or a Soft phone on their PC. Internal VoIP phone networks allow outbound and inbound calling on standard PSTN lines through the use of FXO adapters

4.3

Uses in Amateur Radio
Sometimes called Radio Over Internet Protocol or RoIP, Amateur radio has

adopted VoIP by linking repeaters and users with Echolink, IRLP, D-STAR, Dingotel and EQSO. In fact, Echolink allows users to connect to repeaters via their computer (over the Internet) rather than by using a radio. By using VoIP Amateur Radio operators are able to create large repeater networks with repeaters all over the world where operators can access the system with actual ham radios. Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use DTMF signals to command the repeater to connect to various other repeaters, thus allowing them to talk to people all around the world, even with "line of sight" VHF radios.

4.4

Click to Call
Click-to-call is a service which lets users click a button and immediately speak

with a customer service representative. The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to click-to-call providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel.

Chapter: 5 Legal issues in different countries
As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services, especially with the encouragement of the state-mandated telephone monopolies/oligopolies in a given country, who see this as a way to stifle the new competition. In the U.S., the Federal Communications Commission now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VoIP operators in the U.S. are required to support local number portability; make service accessible to people with disabilities; pay regulatory fees, universal service contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act(CALEA). VoIP operators also must provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VoIP operators also receive the benefit of certain U.S. telecommunications

regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VoIP service -- those who are unable to determine the location of their users -- are exempt from state telecommunications regulation. Throughout the developing world, countries where regulation is weak or captured by the dominant operator, restrictions on the use of VoIP are imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service. In Ethiopia, where the government is monopolizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after popularity in VoIP reduced the income generated by the state owned telecommunication company. In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet). VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power". The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g., the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007. In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India. This effectively means that people who have PCs can use them to make a VoIP call to any number, but if the remote side is a normal phone, the gateway that converts the VoIP call to a POTS call should not be inside India.

In the UAE, it is illegal to use any form of VoIP, to the extent that websites of Skype and Gizmo Project don't work.

5.1

IP Telephony in Japan
In Japan, IP telephony (IP 電 話 IP Denwa) is regarded as a service applied by

VoIP technology to whole or a part of the telephone line. As of 2003, IP telephony services have been assigned telephone numbers. IP telephony services also often include videophone/video conferencing services. According to the Telecommunication Business Law, the service category for IP telephony also implies the service provided via Internet, which is not assigned any telephone number. IP telephony is basically regulated by Ministry of Internal Affairs and Communications (MIC) as a telecommunication service. The operators have to disclose necessary information on its quality, etc., prior to making contracts with customers, and have an obligation to respond to their complaints cordially. Many Japanese Internet service providers (ISP) are including IP telephony services. An ISP who also provides IP telephony service is known as a "ITSP (Internet Telephony Service Provider)". Recently, the competition among ITSPs has been activated, by option or set sales, in connection with ADSL or FTTH services. The tariff system normally applied to Japanese IP telephony is described below;

A call between IP telephony subscribers, limited to the same group, is usually free of charge. A call from IP telephony subscribers to a fixed line or PHS is usually a uniformly fixed rate all over the country. Between ITSPs, the interconnection is mostly maintained at VoIP level.

Where the IP telephony is assigned normal telephone number (0AB-J), the condition for its interconnection is considered same as normal telephony. Where the IP telephony is assigned specific telephone number (050), the condition for its interconnection is described below. Interconnection is sometimes charged. (Sometimes, it's free of charge.) In case of free-of-charge, mostly, communication traffic is exchanged via a P2P connection with the same VoIP standard. Otherwise, certain conversions are needed at the point of the VoIP gateway which incurs operating costs.

5.2

Technical Details
The two major competing standards for VoIP are the ITU standard H.323 and the

IETF standard SIP. Initially H.323 was the most popular protocol, though in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage. However, in backbone voice networks where everything is under the control of the network operator or Telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP. User will log on to an IP telephone like they log on to a computer. Phone charging could then be made on per user instead on per phone basis, which would greatly enhance user mobility. Many user could log on to the same IP telephone to accept incoming calls without needing to worry about forwarding calls when leaving their office. Logging on to an IP telephone with a smart card will make the logging on procedure very simple and, will furthermore offer a range of new possibilities, from online shopping and payment to accessing a bank account. The preferred messaging system for users would be one universal (in and out) box for voice, fax, SMS and e-mail message. The convergence of these services will enable users to manage fax, e-mail and voicemail from a single application on a dedicated VoIP terminal. The survey is the best way to establish which services, functions and applications are important to users. The question posed to a group of CT (Computer Telephony) users was:” How useful is each of the following CT features? (Answer rate on a scale of 1 to 10 where 10 is extremely useful a 1 is not at all useful)”. As we can see call switching and automated routing were identified as the most important services, followed by CLIP and screen-pops. Name dialing, VoIP and FoIP fared marginally. This example tells us that VoIP without supplementary services, enhanced functionally and new applications will not be sufficient to persuade users to migrate to the technology.

Fig.5.1 Useful of CT features

5.3

Applications
IP telephony enables a whole new generation of applications which are

impossible with other telephony architectures. Some examples of the applications that are likely to be useful are as follows: 1. Advanced Intelligent Network Features: Use advanced intelligent network (AIN) features such as Caller ID, voice mail, call waiting, pre-and-post paid calling cards, call blocking, and auto call-back in IP telephony. 2. Voice Calls from Mobile Laptop PCs: Call office or home, fro m hotel, airport, etc. using multimedia laptop PCs with wireless connection to Internet. This could be ideal for submitting or retrieving voice messages. 3. Airlines Reservations: Use a Java applet to visually display interactive voice response options rather than forcing users to wait through very long recorded instructions and go through multi-level menus requiring the use of a telephone keypad. 4. Internet-aware Telephones: Use enhanced ordinary telephone (wired or wireless) as an Internet access device as well as for normal telephony. Directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice (or text) reply. 5. Voice Annotated Documents: Send voice messages and voice annotated documents to integrated voice/data mailboxes. Voice annotated documents and multimedia files can easily become standard within office suites in the near future.

6. Internet Call Center Access: Access customer service agents online over an Internet call center. 7. Virtual Call Centers: Support the integrated voice and data requirements of call center agents working from their homes. 8. Live Auction Websites: Create live audio auction websites for excess inventory. Use Java applets on the phone to manage the bidding process and to track who “raised a hand” to bid first, etc. 9. Presence and Instant Messaging: Use instant messenger service to determine when geographically distributed colleagues are available for a quick conference call with a customer. 10. Electronic Business Cards: Send an enriched electronic virtual business card (vCard) including photo and audio file automatically with every call as caller ID information (or selectively during the middle of call). 11. Integrated Voice and Data Information: Integrate voice and data information collected during a call with sales force automation applications. 12. Personalized Music On-hold: Play personalized announcements or music from a favorite MP3 recording or Internet radio station while callers are on hold.

Conclusion
Throughout the developing world, countries where regulation is weak or captured by the dominant operator, restrictions on the use of VoIP are imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service. In Ethiopia, where the government is monopolizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after popularity in VoIP reduced the income generated by the state owned telecommunication company. User will log on to an IP telephone like they log on to a computer. Phone charging could then be made on per user instead on per phone basis, which would greatly enhance user mobility. Many user could log on to the same IP telephone to accept incoming calls without needing to worry about forwarding calls when leaving their office. The preferred messaging system for users would be one universal (in and out) box for voice, fax, SMS and e-mail message. The convergence of these services will enable users to manage fax, e-mail and voicemail from a single application on a dedicated VoIP terminal.

References

• • 2000. • • • • • • • • • • • • • • • •

IEEE Network Magazine, Special Issue on Internet Telephony, May 1999. IEEE Communications Magazine, Special Issue on Internet Telephony, April H. Liu and P. Mouchtaris, “VoIP Signaling: H.323 and Beyond”, IEEE Communications Magazine, October 2000. ITU-T Recommendation G.107, “The E-Model, a computation model for use in transmission planning”, December 1998. ITU-T Recommendation P.861, “Objective quality measurement of telephoneband speech codecs”, August 1996. IETF RFC 2960, “Stream Control Transmission Protocol”, October 2000 “An Introduction to VoIP and VOCAL”, by Luan Dang, Cullen Jennings and David G. Kelly, 10/11/2002 “VoIP In A Nutshell“, Website, http://www.no vastars.com “Protocols.com”, Website, http://www.protocls.com “How IP Telephony Works”, Jeff Tyson, http://comp uter.howstuffworks.co m/ip-telephony1.htm “The Growth of Internet Telephony: Legal and Policy Issues”, Emir A. Mohammed, “VoIP”, http://members.tripod.com/nguyen225/page1.htm “Factors in the Success of Voice Quality in Converging Telephony and ip Networks”, Stefan Pracht “Internet Telephony and Voice Compression”, Kelly Ann Smith and Daniel Brushteyn, “Voice over Internet Protocol (VoIP)”, Balz Wyss, Microsoft Corporation, March 2003 “The Rise of Internet Telephony”, http://www.nortelnetworks.com “Feature Interaction in Internet Telephony”, Jonathan Lennox, Henning Schulzrinne, Columbia University “Next-Gen VoIP Services and Applications Using SIP and Java”, Pingtel, http://www.techguide.com

• • • • 2000. • • • • • • • • • •

“Voice

over

IP

(VoIP)”,Jerry

Ryan,

Telogy

Networks,

http://www.techguide.com “Next Generation Telephony: A Look at Session Initiation Protocol” Thomas Doumas, May 1999, Hewlett Packard. IEEE Network Magazine, Special Issue on Internet Telephony, May 1999. IEEE Communications Magazine, Special Issue on Internet Telephony, April R. Cole and J.H. Rosenbluth, Voice over IP performance monitoring”, ACM Computer Communications Review, Vol. 31, April 2001 H. Liu and P. Mouchtaris, “VoIP Signaling: H.323 and Beyond”, IEEE Communications Magazine, October 2000. ITU-T Recommendation G.107, “The E-Model, a computation model for use in transmission planning”, December 1998. ITU-T Recommendation P.861, “Objective quality measurement of telephone band speech codecs”, August 1996. IETF RFC 2960, “Stream Control Transmission Protocol”, October 2000. ITU-T Recommendation H.225.0, “Call Signaling protocols and media stream packetization for packet based multimedia communication systems”, September 1999. IETF RFC 2705, “Media Gateway Control Protocol (MGCP), Version 1.0”, October 1999 IETF RFC 2508, “Compressing IP/UDP/RTP Headers for Low-Speed Serial Links”, February 1999 Daniel Collins, Carrier Grade Voice Over IP, McGraw-Hill Professional Telecom P. Galiotos, T. Dagiuklas, “QoS Management for an Enhanced VoIP Platform using R-factor and Network Load Estimation Functionality”, has been submitted for presentation at HSNMC’02, January 2002.

APPENDIX

List of Figures
Title 1. An overview of IP Telephony Fig. No. Fig.1.1 Page No. 2

2. 3. 4. 5. 6. 7.

Analog telephone adaptor Complete VoIP solution PC to PC telephone call Phone to phone call via IP H.323 VoIP gateway architecture Useful of CT features

Fig.1.2 Fig.1.3 Fig.2.1 Fig.2.2 Fig.3.1 Fig.5.1

4 5 11 13 17 26