A Seminar Report On

VoIP TECHNOLOGY
Submitted in partial fulfillment of award of degree of Bachelor of Technology In Electronics and Communication SUBMITTED BY: ROLL NoB.TECH 3rd YEAR SEMINAR IN-CHARGE Mr. PRADEEP KUMAR

Department Of Electronics and Communication Kamla Nehru Institute of Physical & Social Sciences Faridipur, Sultanpur-228118

Session 2012-13

Department of Electrical & Electronic Engineering Kamla Nehru Institute of Physical & Social Sciences, Sultanpur CERTIFICATE
This is certified that seminar entitled VOIP TECHNOLOGY which is submitted by Ms.GARIMA BHARDWAJ, B.Tech (Third Year) Electronics and communication, is a partial fulfillment towards the award of Degree of Bachelor of Technology in Electronics and Communication Engg. Ms.GARIMA BHARDWAJ, B.Tech (Third Year), Ec, has prepared the seminar under my guidance in the session 2012-13 and delivered it successfully.

Date: March/19/2013 HOD INCHARGE Mr. R .K.YADAV KUMAR Electronics and communication Electronics and communication
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SEMINAR

Mr.PRADEEP

ACKNOWLEDGEMENT

I thank my seminar guide Mr. PRADEEP KUMAR, Lecturer, for his proper guidance, and valuable suggestions. I am indebted to Mr. R.K. YADAV, the HOD, Electronics and communication division & other faculty members for giving me an opportunity to learn and present the seminar on the topic "voip technology. If not for the above mentioned people my seminar would never have been completed successfully. I once again extend my sincere thanks to all of them.

ABSTRACT
The growing excitement surrounding the transport of telephony services over traditional data networks such as the Internet, corporate-enterprise intranets and new service provider extranets has led to the development of cost efficient gateway equipment based on embedded systems that converts analog telephony information such as voice and fax into packet data suitable for transport over IP, Frame Relay and ATM networks. As a result, the long-time promise of being able to replace or enhance the traditional PBX by combining voice and data services onto a single network can now finally is realized. In order to do so, a very low-cost telephony device capable of directly exchanging IP packets with the data network is required. Development of this 'VoIP Phone' will require the development of a 'system on a chip' which combines digital signal processing functions, microcontroller functions, analog interface, telephone user interface, network interface, and associated glue logic. Voice over IP (VoIP) is an alternative to traditional circuit-switched telephony that allows human voice and video to travel over existing packet data networks along with traditional data packets. This report looks at the functional requirements and features of an IP Telephone and examines the implementation issues that must be considered.

TABLE OF CONTENT
Chap-1 Introduction 10-12
1.1 VoIP Overview 10 1.2 VoIP History 11 1.3 Consumer Market 11-12

Chap-2 VoIP Principles 18

2.1 13 2.1.1 14 2.1.2 Accuracy Resolution Analog to Digital Conversion

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14 2.1.3 14 2.1.4 14-15 2.2 15-16 2.3 16 2.3.1 16-17 2.3.2 17 2.4 18 End to End VoIP Network H.323 Signaling Protocol SIP Signaling Protocol Transport and Signaling Protocol in VoIP Processing of Digital Signal and Compression Sampling Rate Quantization Error

Chap-3 VoIP Advantages and Disadvantages 19-21

3.1 19 3.2 19-20 3.3 20 6 Quality of Service Disadvantages Advantages

Chap-4

Mobile VoIP

21-22
4.1 21 4.2 22 4.3 23 Software Clients Recent Developments Technologies

Chap-5 VoLTE 24 Chap-6 International VoIP Implementation 25-26 6.1 25-26 Chap-7 conclusion 27-28 References 29 IP telephony in Japan

LIST OF FIGURES
1. Residential Network Using VoIP 11 2. 4-Channel Stereo Multiplexed Analog-to-Digital converter 13 3. Analog-to-Digital conversion Signal 15 4. End to End VoIP Network 18

ABBREVIATIONS
VoIP PSTN Network VoBB IP SMB SIP QoS IETF Force TCP Protocol UDP SCTP User Datagram Protocol Stream Control Transmission Transmission Control Voice over Broad Band Internet Protocol Small to Medium Business Session Initiate Protocol Quality of Service Internet Engineering Task Voice over Internet Protocol Public Switch Telephone

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Protocol HTTP SMTP Protocol MIKEY RAS status PC UMA VoLTE Evolution ITSP Provider HSDPA Access High Speed Downlink Packet Internet Telephony Service Personal Computer Unlicensed Mobile Access Voice over Long Term Multimedia Internet Keying Registration, Admission and Hyper Text Transfer Protocol Simple Mail Transfer

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Chapter-1 INTRODUCTION
1.1 VoIP Overview
Voice over IP (VoIP, or Voice over Internet Protocol) commonly refers to the communication protocols, technologies, methodologies, and transmission techniques involved in the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Other terms commonly associated with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. Internet telephony refers to communications services voice, fax, SMS, and/or voice-messaging applications that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication, while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls

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1.2 VoIP History

Early providers of Voice over IP services offered business models (and technical solutions) that mirrored the architecture of the legacy telephone network. Second generation providers, such as Skype have built closed networks for private user bases, offering the benefit of free calls and convenience, while denying their users the ability to call out to other networks. This has severely limited the ability of users to mix-and-match third-party hardware and software. Third generation providers, such as Google Talk have adopted the concept of Federated VoIP - which is a complete departure from the architecture of the legacy networks. These solutions typically allow arbitrary and dynamic interconnection between any two domains on the Internet whenever a user wishes to place a call.

1.3 Consumer Market

A major development that started in 2004 was the introduction of mass-market VoIP services that utilize existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing.

Fig:1:- Residential network using VoIP

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VoIP solutions aimed at businesses have evolved into unified communications services that treat all communicationsphone calls, faxes, voice mail, e-mail, Web conferences and moreas discrete units that can all be delivered via any means and to any handset, including cell phones. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.

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Chapter-2 VoIP Phone Principles

VoIP works like that. First the A/DC (Analog to Digital Converter) to convert analog voice to digital signals. Now the bits have to be compressed in a good format for transmission. Then we have to insert our voice packets in data packets using a realtime protocol (RTP over UDP over IP). For signaling between customers terminal unit we need a signaling protocol ITU-T H.323 or SIP (Session Initiate Protocol). At Rx site we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card (or phone). All that must be done in a real time fashion cause we cannot waiting for too long a vocal answer. On voice transport via packet oriented network could be respect requirements of QoS (Quality of Service)

2.1 Analog to digital conversion

An analog-to-digital converter (abbreviated ADC, A/D or A to D) is a device that converts a continuous quantity to a discrete time digital representation. An ADC may also provide an isolated measurement. The reverse operation is performed by a digital-to-analog converter (DAC). An ADC has an analog reference voltage or current against which the analog input is compared. The digital output word tells us what fraction of the reference voltage or current is the input voltage or current.

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Fig:2:- 4-channel stereo multiplexed analog-to-digital converter

Factors for analog to digital conversion 2.1.1 Resolution

The resolution of the converter indicates the number of discrete values it can produce over the range of analog values. The values are usually stored electronically in binary form, so the resolution is usually expressed in bits. In consequence, the number of discrete values available, or "levels", is a power of two. For example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels, since 28 = 256. The values can represent the ranges from 0 to 255 (i.e. unsigned integer) or from 128 to 127 (i.e. signed integer), depending on the application.

2.1.2 Accuracy
An ADC has several sources of errors. Quantization error and (assuming the ADC is intended to be linear) non-linearity are intrinsic to any analog-to-digital conversion. There is also a so-called aperture error which is due to a clock jitter and is revealed when digitizing a time-variant signal (not a constant value). These errors are measured in a unit called the least significant bit (LSB). In the above example of an eight-bit ADC, an error of one LSB is 1/256 of the full signal range, or about 0.4%.

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2.1.3 Quantization Error

Quantization error (or quantization noise) is the difference between the original signal and the digitized signal. Hence, the magnitude of the quantization error at the sampling instant is between zero and half of one LSB. Quantization error is due to the finite resolution of the digital representation of the signal, and is an unavoidable imperfection in all types of ADCs.

2.1.4 Sampling Rate

The analog signal is continuous in time and it is necessary to convert this to a flow of digital values. It is therefore required to define the rate at which new digital values are sampled from the analog signal. The rate of new values is called the sampling rate or sampling frequency of the converter. A continuously varying band limited signal can be sampled (that is, the signal values at intervals of time T, the sampling time, are measured and stored) and then the original signal can be exactly reproduced from the discrete-time values by an interpolation formula. The accuracy is limited by quantization error. However, this faithful reproduction is only possible if the sampling rate is higher than twice the highest frequency of the signal. This is essentially what is embodied in the Shannon-Nyquist sampling theorem .Since a practical ADC cannot make an instantaneous conversion, the input value must necessarily be held constant during the time that the converter performs a conversion (called the conversion time). An input circuit called a sample and hold performs this taskin most cases by using a capacitor to store the analog voltage at the input, and using an electronic switch or gate to disconnect the capacitor from the input. Many ADC integrated circuits include the sample and hold subsystem internally.

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Fig:3:- Analog to Digital Conversion Signal

2.2 Processing of digital signal and Compression

There are so many factors affecting voice quality: the broadband connection, bandwidth, hardware, software and the technology itself. The bandwidth, hardware and software factors are in our control - we can change and tweak and improve on them; so when we speak of voice quality in VoIP, we often point a finger to the underlying technology itself, something which is beyond our control as users. A prominent element of VoIP technology is data compression. Data compression is a process whereby voice data is compressed to render it less bulky for transfer. Compression software (called a codec) encodes the voice signals into digital data that it compresses into lighter packets that are then transported over the Internet. At the destination, these packets are decompressed and given their original size (though not always), and converted back to analog voice again, so that the user can hear. Codecs are not only used for compression, but also for encoding, which, simply said is the translation of analog voice into digital data that can be transmitted over IP networks. During compression, the data is confined to a structure (packet) proper to the

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compression algorithm. The compressed data is sent over the network and once it reaches its destination, it is decompressed back to its original state before being decoded. In most cases, however, it is not necessary to decompress the data back, since the compressed data is already in a consumable state.

2.3 Transport and signaling protocol in VoIP network

Voice over IP (VoIP) is an alternative to traditional circuit-switched telephony that allows human voice and video to travel over existing packet data networks along with traditional data packets. VoIP technology uses IP-based networks to establish and manage communication sessions between terminal devices. Two types of protocols are used in VoIP: signaling protocols and media transport protocols. Signaling allows call information to be carried across network boundaries providing session setup, control and teardown. VoIP signaling protocols can be generally divided into two main groups, client-server and user-to-user. In the latter group, SIP and H.323 are the two most popular.

2.3.1 SIP Signaling Protocol

SIP is a more recent standard for multimedia conferencing over IP. The standard was defined by the Internet Engineering Task Force (IETF) and is conceptually simpler than H.323. SIP is used for creating, modifying and terminating sessions between endpoints. The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol(UDP), or Stream Control Transmission Protocol (SCTP). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP) .Some SIP features are * call forwarding

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* * * * * * * *

caller and calling ``number'' delivery, where numbers can be any (preferably unique) naming scheme; personal mobility, i.e., the ability to reach a called party under a single, location independent address even when the user changes terminals; terminal-type negotiation and selection: a caller can be given a choice how to reach the party, e.g., via Internet telephony, mobile phone etc terminal capability negotiation; caller and caller authentication; Invitations to multicast conferences.

2.3.2 H.323 Signaling Protocol

H.323 defines a family of protocols specified by the ITU research group [ITU06]. The standard provides a foundation for signaling in order to exchange voice, video and data communications in an IP-based network. H.323 supports Secure RealTime protocol (SRTP) for media confidentiality, and Multimedia Internet Keying (MIKEY) for key exchange. It is important to emphasize that the signaling is only protected up to the gateway. H.323-based networks have the ability to manage available resources for call routing via H.323 gatekeepers. Gatekeepers are used for address resolution, terminal devices admission control (based on bandwidth availability, concurrent call limitations, or registration privileges), bandwidth management, and zone management (the routing of calls originating or terminating in the gatekeeper zone, including multiple path reroute). Gateways coordinate calls by communicating with gatekeepers using the Registration, Admission, and Status (RAS) protocol. Gatekeepers are the central part of an H.323 network.

2.4 End to End VoIP Network

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Fig:4:- End to End voip Network

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Chapter-3 VoIP ADVANTAGES AND DISADVANTAGES

3.1 Advantages
* * * * * * * * The main advantage is the amount of money is saved on the phone bills as compared to a traditional phone line. Inexpensive and easy to use. Since it is simple, upgrading is relatively simpler too It can be integrated with an existing phone connection. With VOIP PC-to-PC, calls are free no matter the distance and PC-to-Phone charges are nominal. For a monthly fee you may make unlimited free calls within a geographic area. A virtual number enables you to make calls from anywhere as long as a broadband connection is available. You may purchase a number in a geography area of your choice, which works out very cheap. VOIP account can be access just like email Id from anywhere in the world as long as you have an internet phone. This makes it easy for those who travel frequently to make calls frequently to those back at home at local call rates, no matter where they are. * * You may call or message or do both at the same time with VOIP services. VOIP cost about half the cost of traditional phone services and it seems that the taxes and surcharges are much lower. Also your bill is easier to understand and it can be viewed via the Internet.

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3.2 Disadvantages
* * * * * Loss of service during outages Without power VoIP phones are useless so in case of emergencies during power cut it can be a major disadvantage. With VOIP emergency calls, it is hard to locate you and send help in time. Some times during calls, there may be periods of silence when data is lost while it is being unscrambled. Latency, traffic and no standard protocol applicable.

3.3 Quality of Service

Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network, as it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of Service (QoS) guarantees. Therefore, VoIP implementations may face problems mitigating latency and jitter. By default, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimized by marking voice packets as being delay-sensitive. A VoIP packet usually has to wait for the current packet to finish transmission, although it is possible to preempt (abort) a less important packet in mid-transmission, although this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit. But every packet must

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contain protocol headers, so this increases relative header overhead on every link traversed, not just the bottleneck (usually Internet access) link.

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Chapter-4 MOBILE VoIP

Mobile VoIP or simply mVoIP is an extension of mobility to a Voice over IP network. There are several methodologies by which a mobile handset can be integrated into a VoIP network .One implementation turns the mobile device into a standard SIP client, which then uses a data network to send and receive SIP messaging, and to send and receive RTP for the voice path. This methodology of turning a mobile handset into a standard SIP client requires that the mobile handset support ,at minimum, high speed IP communications. In this application,standard VoIP protocols (typically SIP) are used over any broadband IP-capable wireless network connection . Another implementation of mobile integration uses a softswitch like gateway to bridge SIP into the mobile network's SS7 infrastructure. In this implementation, the mobile handset continues to operate as it always has (as a GSM or CDMA based device), but now it can be controlled by a SIP application server which can now provide advanced SIP based services to it. Several vendors offer this kind of capability today. Mobile VoIP will require a compromise between economy and mobility. For example, Voice over Wi-Fi offers potentially free service but is only available within the coverage area of a Wi-Fi Access Point. High speed services from mobile operators using HSDPA may have better audio quality and capabilities for metropolitan-wide coverage including fast handoffs among mobile base stations, yet it will cost more than the typical Wi-Fi-based VoIP service. Mobile VoIP will become an important service in the coming years as device

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manufacturers exploit more powerful processors and less costly memory to meet user needs for ever-more 'power in their pocket'. Smartphones in mid-2006 are capable of sending and receiving email, browsing the web (albeit at low rates) and in some cases allowing a user to watch TV. Juniper research predicts that mobile VoIP users will exceed 100 million by 2012 and InStat projects 288 million subscribers by 2013. The challenge for the mobile operator industry is to deliver the benefits and innovations of IP without losing control of the network service. Users like the Internet to be free and high speed without extra charges for visiting specific sites. Such a service challenges the most valuable service in the telecommunications industry voice and threatens to change the nature of the global communications industry.

4.1 Technologies
Mobile VoIP relies on two main technologies:
the

Unlicensed Mobile Access Generic Access Network, designed to allow VoIP

to run over the GSM cellular backbone

the

standard used by most VoIP services, and now being implemented on mobile

handsets.

4.2 Recent developments

In the summer of 2006, a SIP (Session Initiation Protocol) stack was introduced and a VoIP client in Nokia E-series dual-mode Wi-Fi handsets (Nokia E60, Nokia E61, Nokia E70). The SIP stack and client have since been introduced in many more E and N-series dual-mode Wi-Fi handsets, most notably the Nokia N95 which has been very popular in Europe. Various services use these handsets. In spring 2008 Nokia introduced a built in VoIP client to the mass market device (Nokia 6300i) running Series 40 operating system. Since then other dualmode WiFi capable Series40 handsets have been equipped with integrated VoIP (Nokia 6260 Slide, Nokia X3-02, Nokia C301). Nokia maintains a list of all phones that have an integrated VoIP client in Forum Nokia. 28

Aircel's battle with some companies allowing VoIP calls on flights is another example of the growing conflict of interest between incumbent operators and new VoIP operators. The company xG Technology, Inc. claims to have produced a mobile VoIP and data system operating in the license-free ISM 900 MHz band (902 MHz 928 MHz). xMax is an end-to-end Internet Protocol (IP) system infrastructure that is currently deployed in Fort Lauderdale, Florida.[4]

4.3 Software clients

Mobile Dialer, as they are termed, enable cell phones are turned into voip enabled devices to exploit and expose the Vo3G/Vowlan Functionalities of the phone. Mobile Dialers are available for various Smartphone/PDA platforms:

Bulk Mobile Dialer for VOIP Companies. Companies like Ascent Telecom (Endura Mobile Dialer), REVE Systems (iTel Mobile Dialer Express), adoresoft, etc. have released Mobile dialers which can be used by other VoIP Providers. REVE Systems, which is a premium VoIP solution provider company, claims that iTel Mobile Dialer Express, which is a Mobile Dialer application for Internet telephony service provider, supports largest range of Symbian Based Nokia handsets. ITel Mobile Dialer Express is also known as "lightest Mobile Dialer" in the industry is available for Symbian, Windows and Blackberry operating System based Mobile Phones.

Mobile Dialers available for retail end user. Many VoIP companies have started providing their customers with a mVoIP client/Mobile Dialer to use the product directly from the user's capable mobile phones

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Chapter-5 VoLTE(Future of VoIP)

The Future: VoLTE As our cellular networks get faster (speed) and quicker (latency), and our handsets get higher-end processors, the possibility of VoIP on our handsets gets closer to reality. Many of us thought HSPA+ and LTE would be the realization of those physical requirements which they still may be.As they are currently deployed HSPA+ and LTE are fairly well matched though LTE has more room to grow than HSPA+. LTE, however, has an ace up its sleeve that may be fatal to HSPA+ and GSM networks and CDMA, too. Why? VoLTE. Voice over LTE is the future. VoLTE shares many of the same benefits that VoIP does, but instead of relying on the hardware at the ends of the call (your smartphones), VoLTE offloads the heavy lifting to the network. Doing so lets VoLTE calls be described as VoIP HD. Calls are crisp and clear, and reportedly sound amazing. Not only that, VoLTE includes the ability to cancel echos and background noise on the back end, not the handset itself. Some smartphones today have two microphones, one listens to your voice, the other listens to the ambient noise and removes that noise from the call using algorithms and hardware. This requires additional hardware, processing power, and 31

time on the handset. The VoLTE solution eliminates that by putting all the work on your carriers network instead.

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Chapter-6 INTERNATIONAL VoIP IMPLEMENTATION

6.1 IP telephony in Japan
In Japan, IP telephony is regarded as a service applied by VoIP technology to the 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)". 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 is usually a uniformly fixed rate all over the country. Between ITSPs, the interconnection is mostly maintained at VoIP level. 33

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 is 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. Since September 2002, the MIC has assigned IP telephony telephone numbers

on the condition that the service falls into certain required categories of quality. High-quality IP telephony is assigned a telephone number, normally starting with the digits 050. When VoIP quality is so high that a customer has difficulty telling the difference between it and a normal telephone, and when the provider relates its number with a location and provides the connection with emergency call capabilities, the provider is allowed to assign a normal telephone number, which is a so-called "0AB-J" number. Voice over IP can be used together with static IP addresses so that one can talk to any computer just the way one uses internet, but instead he can access IP-address as definitive unique 'internet voip'-phone number.

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Chapter-7 CONCLUSION
In this document, I have explained how VoIP in virtual environments can be achieved and what factors play an important role. Additionally, I described my own work on this topic. Now, what are the conclusions that can be drawn from all this? When we consider VoIP applications in general, they will probably become more widely used as time evolves. Currently, the main problem for such applications is the lack of QoS guarantees. When QoS supporting protocols like RSVP are used on a larger scale, this will certainly make VoIP more popular since people can then communicate with the quality that they desire. On LANs, where there is normally plenty of bandwidth, VoIP applications can already be used with little or no problems. However, on a larger scale, like the Internet, such QoS providing protocols will be necessary to make VoIP applications perform adequately. In the computer industry, everything evolves very rapidly. Therefore, I assume that the available bandwidth on networks will keep getting larger. This will also be helpful for the spreading of VoIP applications. When the available capacity is sufficiently large, even high quality sound will be possible, which will certainly be a stimulus for the use of VoIP programs. Furthermore, since compression techniques are still improving, such high-quality communications will be available even sooner. Standards like H.323 and SIP make interoperability between applications of different developers possible. This way, people can choose from a variety of VoIP applications and use the ones they like the most. In turn, this will stimulate the use of VoIP. Also, since more applications will be developed, the possibilities of these applications will keep growing and improving. The use of VoIP as a telephony alternative can save quite some costs. Since voice and

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data traffic can be integrated, the necessary infrastructure to provide both services is reduced. This integration will also make better use of the available bandwidth: first of all, bandwidth on a network is rarely entirely filled with data traffic. Second, classic telephone calls waste a lot of bandwidth since this bandwidth is reserved for the two parties even when someone is not speaking. Making long distance telephone calls over the Internet or another IP network will also be cheaper than using the telephone network for this purpose. For VoIP in networked virtual environments there are certainly a lot of possible applications. Currently, there are not many programs which provide this functionality, but I definitely believe that this will change. When better quality can be guaranteed, such applications can be an attractive alternative to chat environments like IRC. As CPU power keeps growing and more dedicated hardware becomes available, better sound localization will improve the realism of the virtual environment. This in turn will make VoIP in virtual environments even more attractive. Among these programs, the Internet telephony application and the 3D environment are quite useful. When enough bandwidth is available, they allow good quality conversations. These programs also made me realize that VoIP has a lot of potential for future development.

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REFERENCES
1. H.323 and Associated http://www.cse.wustl.edu/~jain/cis788-99/h323/index.html Protocols,

2. Voip Products, services and issues, http://www.cse.wustl.edu/~jain/cis788-99/voip_products/index.html 3. Voice Over IP: Protocols and Standards, http://www.cse.wustl.edu/~jain/cis788-99/voip_protocols/index.html 4. Voice over IP [Audio/Video http://www.cse.wustl.edu/~jain/cis788-99/h_8voip.htm 5. Voice over IP: Issues http://www.cse.wustl.edu/~jain/talks/voip.htm and recording], Challenges,

6. Session Initiation Protocol (sip), http://www.ietf.org/html.charters/sipcharter.html

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