Über die Autorin bzw. den Autor

Dr. Henry Sinnreich (Richardson, TX) is Chief Technology Officer at Pulver.com, a leading media company for VoIP and Internet communication services. Dr. Sinnreich has held engineering and executive positions at MCI where he was an MCI fellow and has been involved in Internet and multimedia services for more than 12 years, including the development of the flagship MCI Advantage service based on SIP. Henry Sinnreich is also a contributor to IETF standards for Internet communications in such areas as SIP telephony devices and using RTP extensions for voice quality monitoring. He was awarded the title Pioneer for VoIP in 2000 at the VON Europe conference. Henry Sinnreich has been a cofounder and board member of the International SIP Forum based in Stockholm. He is a frequent speaker and is known as the leading evangelist, worldwide, for SIP based VoIP, presence, IM, multimedia, and integration of applications with communications. Dr. Sinnreich is also a guest lecturer at the Engineering School of the Southern Methodist University in Dallas, TX.

Alan B. Johnston (St. Louis, MO) is a Consulting Member of Technical Staff at Avaya, Inc. He has coauthored the core Internet SIP standard RFC 3261 and four other SIP related RFCs. He is the co-chair of the IETF Centralized Conferencing Working Group and is on the board of directors of the International SIP Forum. His current areas of interest include peer-to-peer SIP and security. Dr. Johnston is a frequent speaker and lecturer on SIP and contributor to various publications, and is an adjunct professor at Washington University in St. Louis, MO.

Von der hinteren Coverseite

"This book is like a good tour guide.It doesn't just describe the major attractions; you share in the history, spirit, language, and culture of the place."<br /> —Henning Schulzrinne, Professor, Columbia University <p>Since its birth in 1996, Session Initiation Protocol (SIP) has grown up. As a richer, much more robust technology, SIP today is fully capable of supporting the communication systems that power our twenty-first century work and life.</p> <p>This second edition handbook has been revamped to cover the newest standards, services, and products. You'll find the latest on SIP usage beyond VoIP, including Presence, instant messaging (IM), mobility, and emergency services, as well as peer-to-peer SIP applications, quality-of-service, and security issues—everything you need to build and deploy today's SIP services.</p> <p>This book will help you</p> <ul> <li>Work with SIP in Presence and event-based communications</li> <li>Handle SIP-based application-level mobility issues</li> <li>Develop applications to facilitate communications access for users with disabilities</li> <li>Set up Internet-based emergency services</li> <li>Explore how peer-to-peer SIP systems may change VoIP</li> <li>Understand the critical importance of Internet transparency</li> <li>Identify relevant standards and specifications</li> <li>Handle potential quality-of-service and security problems</li> </ul>

Aus dem Klappentext

This book is like a good tour guide.It doesn't just describe the major attractions; you share in the history, spirit, language, and culture of the place.
--Henning Schulzrinne, Professor, Columbia University

Since its birth in 1996, Session Initiation Protocol (SIP) has grown up. As a richer, much more robust technology, SIP today is fully capable of supporting the communication systems that power our twenty-first century work and life.

This second edition handbook has been revamped to cover the newest standards, services, and products. You'll find the latest on SIP usage beyond VoIP, including Presence, instant messaging (IM), mobility, and emergency services, as well as peer-to-peer SIP applications, quality-of-service, and security issues--everything you need to build and deploy today's SIP services.

This book will help you

• Work with SIP in Presence and event-based communications
• Handle SIP-based application-level mobility issues
• Develop applications to facilitate communications access for users with disabilities
• Set up Internet-based emergency services
• Explore how peer-to-peer SIP systems may change VoIP
• Understand the critical importance of Internet transparency
• Identify relevant standards and specifications
• Handle potential quality-of-service and security problems

Auszug. © Genehmigter Nachdruck. Alle Rechte vorbehalten.

Internet Communications Using SIP

John Wiley & Sons

Chapter One

The telecommunications, television, and information technology (IT) network industries are all transformed by the Internet. The transformation is driven by the need for growth based on new services, more complete global coverage, and consolidation. In this chapter, we will explore some of the problems and solutions for end users and every type of business because of the profound disruptions caused by the Internet.

Problem: Too Many Public Networks

Before the emergence of the Internet, users and service providers were generally accustomed to thinking in terms of four distinct network types: Networks for IT (data), networks for voice, mobile networks, and networks for television. Each of these dedicated network types could, in turn, be divided into many incompatible regional and even country-specific flavors with different protocol variants.

Thus, we find many types of telephony numbering plans, signaling, and audio encodings; several TV standards; and various types and flavors of what the telecom industry calls data networks-all of them incompatible and impossible to integrate into one single global network.

The mobile telephone networks have converged on a smaller number of standards in the second generation (2G) networks and in the emerging third generation (3G) mobile networks. It may turn out, however, that with the proliferation of new radio technologies for the so-called 4th generation (4G), such as Wi-Fi and WiMAX, all modern mobile networks will become just a wireless access mechanism to the Internet, where all public communications, entertainment, and applications will reside anyhow.

Data networks that originated in the telecom industry came in many forms, such as digital private lines, X.25, Integrated Services Digital Network (ISDN), Switched Multimegabit Data Service (SMDS), Frame Relay, and Asynchronous Transfer Mode (ATM) networks. These so-called data networks were mostly inspired by circuit-switched telephony concepts. Their names are meant to suggest that they were not designed primarily to carry voice.

Voice networks are still used for data and fax because of their general availability, though less and less so. However, these networks have come to the end of their evolution, since they are fundamentally optimized for voice only. TV networks were designed and optimized for the distribution of entertainment video streams.

Needless to say, all network types (data, voice, TV, and mobile) have specific end-user devices that cannot be ported to other service providers or network types, and most often cannot be globally deployed.

The impact of the Internet has made the wired and wireless phone companies and the TV cable companies look for new business models that can take advantage of Internet technologies and protocols, among them the Session Initiation Protocol (SIP) for real-time communications, such as Voice over IP (VoIP), instant messaging (IM), video, conferencing/collaboration, and others. Examples of the various categories and their business models are illustrated in Table 1.1. We assume that most readers are familiar with the acronyms used in the table, and we also explain these acronyms and terms in the book. They can also be found in the index.

The proliferation of isolated communication islands as shown in Table 1.1 makes them less useful as their number keeps increasing (think of many more communication islands all over the world). Building communication islands (also called "walled gardens") is in conflict with Metcalfe's law that the value of the network increases by the square of the number of connected endpoints. Last, but not least, in case of an emergency, having many networks that cannot communicate directly is not very helpful.

Closed networks are an impediment for innovation, since innovators must work (technology and legal agreements) with every closed network separately to bring a new service or product to market. By contrast, the Internet extends the reach for new applications and services instantly to the whole world.

Another observation from Table 1.1 is that the strongest financing available is at present for closed networks (walled gardens), the ones that are most limited in reach and usefulness. This raises business issues and regulatory questions (what are the public interest obligations, if any?) that are beyond the scope of this book.

Incompatible Enterprise Communications

Enterprise communication systems are often an even greater mix of incompatible and disjoint systems and devices:

* Proprietary PBX and their phones. Phones from one PBX cannot be used by another.

* Instant messaging is a separate system from the PBX.

* Various IM systems don't talk to each other.

* Voice conferencing and web-based collaboration use yet other systems.

Maintaining various incompatible and nonintegrated proprietary enterprise systems is quite costly and reduces the overall productivity of the workforce.

Network Consolidation: The Internet

The Internet has benefited from a number of different fundamentals compared to legacy networks, such as the tremendous progress of computing technology and the open standard Internet protocols that define it. This progress can be attributed to the expertise of the research, academic, and engineering communities whose dedication to excellence and open collaboration on a global basis have surpassed the usual commercial pressure for time-to-market and competitive secrecy.

The result is an Internet that uses consistent protocols on a global basis, and is equally well suited to carry data, transactions, and real-time communications, such as instant messaging (IM), voice, video, and conferencing/collaboration. Actually, the Internet is the "dumb network," designed for any application, even those not yet invented. This is in stark contrast to the isolated "walled gardens" with central control of all services illustrated in Table 1.1.

Voice over IP

Although the Internet has quickly established itself as the preeminent network for data, commercial transactions, and audio-video distribution, the use of voice over the Internet has been slower to develop. This has less to do with the capability of the Internet to carry voice with equal or higher quality than the telephone network but rather with the complex nature of signaling in voice services, as you will see in Chapter 6, "SIP Overview."

There are various approaches for voice services over the Internet, based on different signaling and control design. Some examples include the following:

* Use signaling concepts from the telephone industry-H.323, MGCP, MEGACO/H.248.

* Use control concepts from the telephone industry-central control and softswitches.

* Use the Internet-centric protocol-Session Initiation Protocol (SIP), the topic of this book.

The movement from such concepts as telephony call models to discovery/ rendezvous and session setup between any processes on any platform anywhere on the Internet is opening up completely new types of communication services.

The use of SIP for establishing voice, video, and data sessions places telephony as just another application on the Internet, using similar addressing, data types, software, protocols, and security as found, for example, on the World Wide Web or e-mail.

Separate networks for voice are no longer necessary, and this is of great consequence for all wired and wireless telephone companies.

Complete integration of voice with all other Internet services and applications probably provides the greatest opportunity for innovation. The open and distributed nature of this service and the "dumb" network model will empower many innovators, similar to what has happened with other industries on the Internet and the resulting online economy.

Most IM systems on the Internet already have voice and telephony capability as well, though if it is proprietary, they cannot intercommunicate without IM gateways, although IM gateways inevitably cannot translate all the features from one system to another. IM gateways are also transitory in nature, since any changes to a proprietary IM protocol may render the gateway close to useless. By contrast, SIP-based communications offer a global standards-based approach for interoperability for presence, IM, voice, and video, as we will show in the following chapters.

Presence-The Dial Tone for the Twenty-First Century?

Unsuccessful telephone calls are a serious drag on productivity and a source of frustration, since both parties waste time and talk to voicemail instead to each other. Also, the timing of the phone call may not be appropriate or not reach the called party in a suitable location. The advent of presence, so well-known from IM systems, can provide much more rich information before trying to make a call in the first place, compared to just hearing the dial tone. Another convenience of SIP and presence is that many contact addresses may reside beneath a buddy icon, so the caller need not to know or worry about picking the right phone number or URI. Presence may, therefore, replace the dial tone used in telephony for well over 100 years.

The Value Proposition of SIP

SIP is not just another protocol. SIP redefines communications and is impacting the telecom industry to a similar or greater degree than other industries. This has been recognized by all telecom service providers and their vendors for wired and wireless services, as well as by all IT vendors. Chapter 2 will provide an overview of how the Internet and SIP are redefining communications.

SIP Is Not a Miracle Protocol

As discussed in Chapter 2, "Internet Communications Enabled by SIP," SIP is not a miracle protocol and is not designed to do more than discover remote users and establish interactive communication sessions. SIP is not meant to ensure quality of service (QoS) all by itself or to transfer large amounts of data. It is not applicable for conference floor control. Neither is it meant to replace all known telephony features, many of which are caused by the limitations of circuit-switched voice or to the regulation of voice services. And such a list can go on.

Various other Internet protocols are better suited for other functions. As for legacy telephony, not all telephone network features lend themselves to replication on the Internet.

The Short History of SIP

By 1996, the Internet Engineering Task Force (IETF) had already developed the basics for multimedia on the Internet (see Chapter 14, "SIP Conferencing") in the Multi-Party, Multimedia Working Group. Two proposals, the Simple Conference Invitation Protocol (SCIP) by Henning Schulzrinne and the Session Initiation Protocol (SIP) by Mark Handley, were announced and later merged to form Session Initiation Protocol. The new protocol also preserved the HTTP orientation from the initial SCIP proposal that later proved to be crucial to the merging of IP communications on the Internet.

Schulzrinne focused on the continuing development of SIP with the objective of "re-engineering the telephone system from ground up," an "opportunity that appears only once in 100 years," as we heard him argue at a time when few believed this was practical.

SIP was initially approved as RFC number 2543 in the IETF in March 1999. Because of the tremendous interest and the increasing number of contributions to SIP, a separate SIP Working Group (WG) was formed in September 1999. The SIP for Instant Messaging and Presence Leveraging (SIMPLE) was formed in March 2001, followed by SIPPING for applications and their extensions in 2002. The specific needs of SIP developers and service providers have led to an increasing number of new working groups. This very large body of work attests both to the creativity of the Internet communications engineering community, and also to the vigor of the newly created industry.

We will shorten the narrative on the history of SIP by listing the related working groups (WG) in chronological order in Table 1.2. We have listed for simplicity the year of the first RFC published by the WG, though the WG was sometimes formed one to two years earlier. Years denote a new WG that has not yet produced any RFC.

The growth of SIP-related standards in the IETF is illustrated and discussed in Chapter 21, "Conclusions and Future Directions."

References in This Book

Because of the multiple developments on the Internet, SIP is being used in ever-more services, user software, and various user devices (such as in SIP phones, PCs, laptops, PDAs, and mobile phones). This is, in effect, a new industry and its participants keep making new contributions to the core SIP standards, mainly in the area of new services and new applications. This book reflects SIP developments up to and including the 64th IETF in November 2005.

We have included, by necessity, many Internet drafts that are designated work in progress, since they are the only reference source for this particular information. Some of these drafts may become standards by the time you are ready to use them; some may be a work in progress and have a higher version number than quoted as of this writing; and still others may be found only in an archive for expired drafts.

The SIP WG drafts that are work in progress can be found online at the IETF web site:

http://ietf.org/html.charters/sip-charter.html

Additional individual submissions and Internet drafts from other working groups can be found at the following site:

http://ietf.org/ID.html

SIP-related drafts that have expired (older than six months) can be found on several archives. As of this writing, following are some of the sites:

www.cs.columbia.edu/sip/drafts

www.softarmor.com/sipwg

Readers may also perform a web search, such as Google, for any IETF SIP-related topic or for any Internet draft or RFC.

Several books have been published on Internet multimedia, Voice over IP, and SIP, some of which are listed here. They focus mainly on how SIP works. This book is less about explaining how SIP works, but rather what it does and the new communications and services it enables.

We have reproduced some of the exciting services and features discussed in the IETF SIP WG and its main offsprings, the SIPPING and SIMPLE Working Groups. Also included in our discussion are some drafts from Bird of Feather (BOF) sessions that have not even made it to an accepted WG charter, such as the peer-to-peer (P2P) SIP group.

Many of these expired proposals may not develop into IETF standards for various reasons, but represent good work, often backed up by running code. The references to such expired Internet drafts are intended to make you aware of these ideas that may otherwise remain buried in an archive. Such references are clearly marked as expired, so as to distinguish them from accepted work in progress items of IETF WGs that are on the path toward acceptance as standards.

SIP Open Source Code and SIP Products

There is an ever-increasing amount of open source code for SIP, and it is increasing in quality. Most or many commercial SIP products are actually based on open source SIP code. An authoritative list of SIP open source code is available from the SIP Forum:

www.sipforum.org

The SIP Forum is also an excellent source for finding commercial SIP software and products for the enterprise, for consumer products, for service providers, tools, and others.

Excellent lists of SIP products are also maintained on the web sites of Pulver.com and Ubiquity.com:

www.pulver.com/products/sip

www.sipcenter.com

References for Telephony

We assume throughout this book some understanding of telephone services and of telecommunication protocols. There is a vast literature pool available on telephony and telecommunications. We refer you to Newton's Telecommunications Dictionary to brush up on various terms that will be used in the following chapters.

Summary

This chapter has discussed some of the problems and solutions to the communications industry by the Internet, and also a brief history of the SIP protocol.

(Continues...)

Excerpted from Internet Communications Using SIPby Henry Sinnreich Alan B. Johnston Copyright © 2006 by Henry Sinnreich. Excerpted by permission.
All rights reserved. No part of this excerpt may be reproduced or reprinted without permission in writing from the publisher.
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