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An increasing number of companies are taking advantage of Voice over IP (VoIP) technology. Before implementing VoIP, companies must decide if the packetized voice should be delivered over Layer 2 networks such as ATM or Frame Relay, or over Layer 3 IP networks. This comprehensive book presents the latest standards and practices so you can determine which approach is best for your company. Drawing upon years of practical experience, the authors explore the unique engineering problems posed by running voice over a connectionless network. They clearly explain the advantages and disadvantages of IP versus traditional packet networks. This book is a significant expansion of the authors' first edition, which published in 1998 and had the distinction of being the first book in the industry dedicated to this important topic of key interest to enterprise customers.
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Product Details

  • ISBN-13: 9780471254829
  • Publisher: Wiley, John & Sons, Incorporated
  • Publication date: 4/3/1998
  • Edition description: Older Edition
  • Edition number: 1
  • Pages: 288
  • Product dimensions: 7.77 (w) x 9.58 (h) x 0.86 (d)

Meet the Author

Daniel Minoli is the Director of Engineering and Development for Teleport Communications Group (TCG). Prior to joining TCG, he was a senior scientist at Bellcore. A pioneer in the field, Dan's research into voice over data began in 1975. He has worked on several major implementations over the last twenty years and has published widely on the subject.

Emma Minoli is founder and Managing Director of Red Hill Consulting, which specializes in web-based technologies for e-commerce and other corporate productivity-enhancing applications.

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Read an Excerpt

Delivering Voice over IP Networks

By Daniel Minoli Emma Minoli

John Wiley & Sons

ISBN: 0-471-38606-5

Chapter One

Introduction and Motivation

1.1 Introduction

Data networks have progressed to the point that it is now possible to support voice and multimedia applications right over the corporate enterprise network and the intranet, for both on-net and off-net applications. Many companies have already deployed IP-based backbones that provide both broadband capabilities and Quality of Service (QoS)-enabled communications. Some companies have deployed Asynchronous Transfer Mode (ATM) networks. Switching technology, particularly in terms of the switched local area network (LAN), has gone a long way in the past five years, providing higher-capacity, lower-contention services across the enterprise campus network. High-speed wide area technology, such as Packet over Synchronous (POS) Optical Network and metro optical services as metro gigabit Ethernet, provide increased bandwidth across the enterprise regional, nationwide, and international networks.

The Internet Engineering Task Force (IETF) Multiprotocol Label Switching (MPLS) specification also directly or indirectly provides improved support of IP services. In addition, QoS-supporting protocols, such as IPv6, Resource, Integrated Services Architecture (intserv), differentiated services (diffserv) in IPv4, and Real-time Transport Protocol (RTP), are now entering the corporate enterprise network (Referenceprovides a treatment of the trends listed here). A lot of industry effort has gone into supporting IP over ATM using a number of technologies, such as Classical IP over ATM (CIOA). All of this opens the door for the possibility of carrying voice over the enterprise network. Interest exists in the carrier arena as to the possibility of modernizing the existing public-switched telephone network (PSTN) with an IP-based infrastructure that would support multiple services, including Voice over IP (VOIP) (see Figure 1.1).

At the same time, commercialized Internet use has increased significantly in the past few years, as companies ventured into Web-based commerce. During the late 1990s, the Internet was reportedly growing 100 percent per quarter. More recently, people have quoted 80 percent growth (or perhaps slightly less) per year. That large collection of backbones, access subnetworks, server farms, and hypertext information that is known as the Internet is acquiring ever-increasing importance, not only for the business community but also for the population at large. Access to information is proving increasingly valuable for education, collaborative work, scientific research, commerce, and entertainment. The advent of HTML-formatted, URL-addressable, and HTTP-obtainable information over the Internet-what is often called the World Wide Web (WWW or W3)-has generated a lot of attention in the past 10 years. Now there is a movement afoot to make the transition to fully multimedia enabled sites that allow voice, video, data, and graphics to be accessed anywhere in the world. The issue so far, however, has been that voice and video, by and large, have been of the stored kind-namely, a one-way download of sound files that are played out in non-real time at the user's PC.

Given this extensive deployment of data networking resources, the question naturally presents itself, is it possible to use the investment already made to carry real-time voice in addition to the data? The desire to build one integrated network goes back to the 1970s, if not even earlier. The Advanced Research Projects Agency, with project DACH-15-75-C0135 (and many other projects with many other researchers), funded the senior author's work in 1975 to look at the feasibility of integrated voice and data packet networks. And Integrated Services Digital Network (ISDN) research started in Japan in the early 1970s (before the idea started to get some real attention in the late 1970s and early 1980s) with the explicit goal of developing and deploying integrated networks. However, a lot of the mainstream work has been in supporting voice and data over circuit-switched time-division multiplexed (TDM) networks. Only some early packet over data work, and then some Fiber Distributed Data Interface II (FDDI II) and Integrated Voice/Data LAN (IEEE 802.9) work, looked at voice support in a non-circuit-mode network. Even for ATM, the emphasis has been, until the past few years, on data services.

The idea of carrying voice over data networks has received considerable commercial attention in the past five years. The ATM Forum, the Frame Relay Forum, and the MPLS Forum have published specifications, and a whole range of voice over data network has appeared and/or is appearing. The work of the ATM Forum and the Frame Relay Forum has focused on connection-oriented networks. However, connectionless IP-based networks are ubiquitous, and so there is a desire to carry business-quality voice over them. The major challenge in this regard is that IP networks do not yet support QoS features. Nonetheless, a plethora of IP phones and IP-to-public-network gateways has entered the market.

This book is one of two related Wiley books published by the authors. This book focuses on the IP telephony technology itself. Figure 1.2 depicts the various voice over data network technologies now evolving, including VOIP. Also note that IP can utilize a number of data link layer services, such as ATM, MPLS, and frame relay. Figure 1.3 depicts a possible scenario of VOIP, as is addressed in this book.

After this introduction in Chapter 1, a basic review of IP technologies is provided in Chapter 2, which covers IP, IPv6, RSVP, RTP, and MPLS. Chapter 3 discusses voice characteristics that can be utilized in packet networks. Chapter 4 discusses adaptive differential pulse code modulation (ADPCM) as applied to packet network environments. Chapter 5 provides an overview of vocoder-based compression methods used in IP. Chapter 6 covers various proposals for delivery of voice in IP environments. Chapter 7 covers the important topic of signaling. Chapter 8 provides a major review of QoS technologies. Chapter 9 covers voice over MPLS. Chapter 10 addresses directory services. Chapter 11 looks at opportunities for traditional carriers. Finally, Chapter 12 briefly looks at wireless opportunities.

1.2 Drivers for Voice over IP

This section discusses a number of drivers for voice over IP.

The Positive Drivers

Besides the potential for savings on long-distance phone charges to communicate with friends or relatives, Internet phones already have a place in the business world. For example, one can leave Internet phones turned on and ready for calls throughout the day; the technology is useful for communicating with coworkers in other parts of the building and at other locations by simply dialing them up on the Internet videophone. If they are at their desks, they can answer immediately. It can be a fine way to ask work-related questions without taking one's hands off the keyboard. The technology is good for telecommuters, who can dial in to the office and see and speak to coworkers while getting a glimpse of the office from home. Similarly, it can be good for distance learning applications. There are both market and business drivers for the introduction of voice telephony over IP at this time.

There have been four main stages of VOIP evolution in the past few years:

1. PC-to-PC (since 1994)

* Connects multimedia PC users, simultaneously online

* Cheap, good for chat, but inconvenient and low quality

2. PC-to-phone (since 1996)

* PC users make domestic and international calls via gateway

* Increasingly services are "free" (e.g.,

3. Phone-to-phone (since 1997)

* Accounting rate bypass

* Low-cost market entry (e.g., using calling cards)

4. Voice/Web integration (since 1998)

* Calls to Web site/call centers and freephone numbers

* Enhanced voice services (e.g., integrated messaging)

Deregulation in the United States and elsewhere could mean that both incumbent carriers and new carriers can enter the market with new services. At various times in the past twenty years, a variety of carriers in the United States were precluded from entering certain telecommunication service sectors. One of the goals of the Telecommunications Act of 1996 was to change that. However, there has recently been a major slowdown in the competitive carrier landscape. This slowdown will be a major drag on the introduction of VOIP, since the new carriers were the principal beneficiaries of a less expensive technology that could be deployed in greenfield environments.

The technology to carry voice over data networks is evolving, as noted in the introduction. There are economic advantages to end users in utilizing an integrated network, not only in terms of direct transmission costs, but also in reducing the network management costs of running separate and technologically different networks. That is the ultimate goal. In the meantime, many companies are, and will be for some time in the future, supporting the infrastructure and cost of multiple networks, including PSTN, private enterprise networks, wireless networks, intranets, business video networks, Internet access networks, and Internet-based Virtual Private Networks. Hence, the need to optimize the usage of all media components on all networks simultaneously, and to take advantage of pricing alternatives between networks, will become even more important as these networks proliferate in the corporate environment, and as the service providers offer increasingly competitive prices.

Separate from technology considerations, business drivers must come into play. Carriers need to make a positive bottom line (e.g., a 15 to 25 percent net bottom line) and be sustainable and self-sufficient. There are new revenue opportunities for Internet service providers (ISPs) in bundling voice service with Internet access. The interexchange carriers (IXCs) can avoid access charges. The local exchange carriers (LECs) can undercut the long-distance prices and offer Inter- LATA services without necessarily having to follow the traditional approach. Cable TV operators can bundle packet voice with cable services and perhaps find a better way to enter the telephony business without having to follow the classical time-slot-interchange method. Wireless companies can make more efficient use of the precious radio spectrum. Figures 1.4 to 1.9 depict typical carrier applications, based on reference. All of these stakeholders can benefit by adding value to the network instead of just growing linearly to simply reach more physical points, and they can benefit by optimizing the economics of both packet-switched and circuits-witched networks. However, the major breakthrough has to come in the form of new services. Simply replacing a circuit-based transport mode with a packet-based transport mode will not justify the replacement of the old network with the new.

In particular, the past few years have seen the emergence of reduced bit-rate voice compression algorithms that can increase the carrying capacity of a network by nearly tenfold (that is, by an order of magnitude) without the investment of additional resources in long-haul transmission facilities. The deployment, for example, of a network supporting near-toll-quality voice at 5.3 kbps rather than the twenty-five-year-old method of 64-kbps-per-call pulse code modulation (PCM) is not likely to be feasible in the context of an existing public switched telephone network because of the extensive embedded base of legacy equipment. Hence, if there is a desire to use the new compression algorithms and achieve a tenfold efficiency gain, then the IP route may be the way to go.

Voice over IP can be deployed in private enterprise networks, but some technology suppliers are concentrating on providing new solutions for carriers, consistent with the approach just outlined. Applications of the evolving VOIP technology include the following:

Internet voice telephony

Intranet and enterprise network voice telephony

Internet fax service

Internet videoconferencing

Multimedia Internet collaboration

Internet call centers

PBX interconnection

It is worth noting that there has been considerable progress recently in developing standards (with supporting equipment to follow) in the area of LAN/intranet-based multimedia (with compressed speech), as shown in Figure 1.9. These efforts will likely become the underpinnings of standards-based approaches to VOIP.

Recent analysis from the firm Frost & Sullivan, published in the 2001 report U.S. Market for Enhanced IP-Based Voice Services, reveals that the VOIP industry generated a revenue of $520 million in 2001 and that the market is projected to reach $31.8 billion by 2007. There has been some penetration in intranet environments. For example, Cisco reportedly expects to deploy VOIP service at over 41 sites, connecting the 35,000 Cisco IP phones already in place, to provide enhanced internal communications.

A few years ago, VOIP was the domain of just a handful of early pioneering companies, including 3Com, Cisco, Clarent, Nuera Communications, and Hypercom. Now, however, this convergence technology has finally been embraced by the more traditional networking and telecommunications vendors who had previously viewed VOIP as a serious threat to their installed bases. The early VOIP pioneering companies have been joined by the classic PBX vendors-Alcatel, Avaya, Ericsson, Mitel, NEC, Nortel Networks, and Siemens. By 2000, all of the vendors had introduced viable VOIP products-often in the form of add-ons, which "IP-enabled" the latest versions of the vendors' TDMs and switching matrix-based PBXs. There were 175 VOIP vendors in 2000; a segment-by-segment breakdown of the key players is shown in Figure 1.10.

Interoperability among VOIP products has been a major stumbling block to widespread acceptance of the technology. The International Telecommunications Union-Telecommunications (ITU-T) H.323 "umbrella" standard, the first posed for VOIP interoperability, proved complex to implement. As a result, in its place were proposed other less complicated standards, from which a number of loosely related standards-for example, the ITU-T H.323 and H.248/MEGACO, the IETF Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and the International Softswitch Consortium (ISC) specifications-emerged. The expectation is that no single standard will be predominant over the next couple of years; interoperability and coexistence will therefore be important.

Naturally, there are going to be challenges in deploying IP-based voice services. Table 1.1 depicts some of these challenges and some potential ways around them.


Excerpted from Delivering Voice over IP Networks by Daniel Minoli Emma Minoli Excerpted by permission.
All rights reserved. No part of this excerpt may be reproduced or reprinted without permission in writing from the publisher.
Excerpts are provided by Dial-A-Book Inc. solely for the personal use of visitors to this web site.

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Table of Contents

About the Authors
Ch. 1 Introduction and Motivation 1
Ch. 2 An Overview of IP, IPOATM, MPLS, and RTP 21
Ch. 3 Issues in Packet Voice Communication 63
Ch. 4 Voice Technologies for Packet-Based Voice Applications 101
Ch. 5 Technology and Standards for Low-Bit-Rate Vocoding Methods 125
Ch. 6 Voice over IP and the Internet 153
Ch. 7 Signaling Approaches 183
Ch. 8 Quality of Service 279
Ch. 9 Voice over MPLS and Voice over IP over MPLS 343
Ch. 10 Telephone Number Mapping ENUM 377
Ch. 11 Carrier Applications 427
Index 477
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