SIP: Understanding the Session Initiation Protocol

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Essential reading for anyone involved in the development and operation of voice or data networks, SIP: Understanding the Session Initiation Protocol is a ground-breaking book that quickly provides a thorough understanding of this revolutionary protocol for IP telephony. It shows how SIP offers a highly scalable and cost-effective way to offer new and exciting telecommunication feature sets, helping professionals design their "next generation" networks and develop new applications and software stacks. Discussions ...
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Overview

Essential reading for anyone involved in the development and operation of voice or data networks, SIP: Understanding the Session Initiation Protocol is a ground-breaking book that quickly provides a thorough understanding of this revolutionary protocol for IP telephony. It shows how SIP offers a highly scalable and cost-effective way to offer new and exciting telecommunication feature sets, helping professionals design their "next generation" networks and develop new applications and software stacks. Discussions include SIP as a key component in the Internet multimedia conferencing architecture, request and response messages, devices in a typical network, types of servers, SIP headers, comparisons with existing signaling protocols including H.323, related protocols SDP (Session Description Protocol) and RTP (Real-time Transport Protocol), and the future direction of SIP. Detailed call flow diagrams illustrate how this technology works with other protocols such as H.323 and ISUP.

It shows how SIP provides a highly-scalable and cost-effective way to offer new and exciting telecommunication feature sets, helping you design your "next generation" network and develop new applications and software stacks.

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Editorial Reviews

Booknews
Introduces the session initiation protocol (SIP), which is a new signaling protocol developed to set up, modify, and tear down multimedia sessions over the Internet. The author, who is an advisory engineer at WorldCom, discusses SIP as a component in the Internet multimedia conferencing architecture, request and response messages, devices in a typical network, types of clients and servers, SIP headers, comparisons with existing signaling protocols, and the related session description protocol and real-time transport protocol. Detailed call flow diagrams illustrate how this technology works with other protocols such as H.323 and ISUP. Annotation c. Book News, Inc., Portland, OR (booknews.com)
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Product Details

  • ISBN-13: 9781580531689
  • Publisher: Artech House, Incorporated
  • Publication date: 1/1/2001
  • Series: Telecommunications Library Series
  • Pages: 220
  • Product dimensions: 6.23 (w) x 9.28 (h) x 0.70 (d)

Meet the Author

Alan B. Johnston is an advisory engineer at WorldCom and an adjunct at Washington University. He holds a B.E.(Hons) in electrical and electronic engineering from the University of Melbourne, Australia and a Ph.D. in electrical engineering from Lehigh University.
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1.SIP and the Internet

The Session Initiation Protocol (SIP) is a new signaling protocol developed to set up, modify, and tear down multimedia sessions over the Internet [1]. This chapter covers some background for the understanding of the protocol. SIP was developed by the Internet Engineering Task Force (IETF) as part of the Internet Multimedia Conferencing Architecture, and was designed to dovetail with other Internet protocols such as TCP, UDP, IP, DNS, and others. This organization and these related protocols will be briefly introduced. Related background topics such as Internet URLs, IP multicast routing, and ABNF representations of protocol messages will also be covered.

1.1 Signaling Protocols

This book is about the Session Initiation Protocol, which is a signaling protocol. As the name implies, the protocol allows two end-points to establish media sessions with each other. The main functions of signaling protocols are as follows:

  • Location of an end-point;
  • Contacting end-point to determine willingness to establish a session;
  • Exchange of media information to allow session to be established;
  • Modification of existing media sessions;
  • Tear-down of existing media sessions.

The treatment of SIP in this book will be from a telephony perspective. This is likely to be one of the first applications of SIP, but not the only one. SIP will likely be used to establish a whole set of session types that bear almost no resemblance to a telephone call. The basic protocol operation, however, will be the same. As a result, this book will use familiar telephone examples to illustrate concepts.

1.2 The Internet Engineering Task Force

SIP was developed by the Internet Engineering Task Force (IETF). To quote The Tao of the IETF [2]: "The Internet Engineering Task Force is a loosely self-organized group of people who make technical and other contributions to the engineering and evolution of the Internet and its technologies." The two document types used within the IETF are Internet-Drafts (I-Ds) and Request for Comments (RFCs). I-Ds are the working documents of the group; anyone can author one on any topic and submit it to the IETF. There is no formal membership in the IETF; anyone can participate. Every I-D contains the following paragraph on the first page: "Internet-Drafts are documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as work in progress."

Internet standards are archived by the IETF as the Request for Comments, or RFC, series of numbered documents. As changes are made in a protocol, or new versions come out, a new RFC document with a new number is issued, which "obsoletes" the old RFC. Some I-Ds are cited in this book; I have tried, however, to restrict this to mature documents that are likely to become RFCs by the time this book is published. A standard begins life as an I-D, then progresses to an RFC once there is consensus and there are working implementations of the protocol. Anyone with Internet access can download any I-D or RFC at no charge using the World Wide Web, ftp, or e-mail. Information on how to do so is on the IETF web site: http://www.ietf.org.

The IETF is organized into working groups, which are chartered to work in a particular area and develop a protocol to solve that particular area. Each working group has its own archive and mailing list, which is where most of the work gets done. The IETF also meets three times per year.

1.3 A Brief History of SIP

SIP was originally developed by the IETF Multi-Party Multimedia Session Control Working Group, known as MMUSIC. Version 1.0 was submitted as an Internet-Draft in 1997. Significant changes were made to the protocol and resulted in a second version, version 2.0, which was submitted as an Internet-Draft in 1998. The protocol achieved Proposed Standard status in March 1999 and was published as RFC 2543 [3] in April 1999. In September 1999, the SIP working group was established by the IETF to meet the growing interest in the protocol. An Internet-Draft containing bug fixes and clarifications to SIP was submitted in July 2000, referred to as RFC 2543 "bis". This document will be first published as an Internet-Draft then as an RFC with a new RFC number, which will obsolete RFC 2543. To advance from Proposed Standard to Draft Standard, a protocol must have multiple independent interworking implementations and limited operational experience. To this end, forums of interoperability tests, called "bakeoffs," have been organized by the SIP working group. Three interoperabiliry "bakeoffs" took place for SIP in 1999, with more planned for 2000. The final level, Standard, is achieved after operational success has been demonstrated [4]. With the documented interoperabiliry of the bakeoffs, SIP should move to Draft Standard status sometime in early 2001.

SIP incorporates elements of two widely used Internet protocols: HTTP (Hyper Text Transport Protocol) used for web browsing and SMTP (Simple Mail Transport Protocol) used for e-mail. From HTTP, SIP borrowed a client-server design and the use of uniform resource locators (URLs). From SMTP, SIP borrowed a text-encoding scheme and header style. For example, SIP reuses SMTP headers such as T o, F r o m, u a t e, and s u b j e c t . In keeping with its philosophy of "one problem, one protocol", the IETF designed SIP to be a pure signaling protocol. SIP uses other IETF protocols for transport, media transport, and media description. The interaction of SIP with other Internet protocols such as IP, TCP, UDP, and DNS will be described in the next section.

1.4 Internet Multimedia Protocol Stack

Figure 1.1 shows the four-layer Internet Multimedia Protocol stack. The layers shown and protocols identified will be discussed.

1.4.1 Physical Layer

The lowest layer is the physical and link layer, which could be an Ethernet local area network (LAN), a telephone line (V.90 or 56k modem) running Point-to-Point Protocol (PPP), or a digital subscriber line (DSL) running asynchronous transport mode (ATM), or even a multi-protocol label switching (MPLS) network. This layer performs such functions as symbol exchange, frame synchronization, and physical interface specification.

1.4.2 Internet Layer

The next layer in Figure 1.1 is the Internet layer. Internet Protocol (IP) [5] is used at this layer to route a packet across the network using the destination IP address. IP is a connectionless, best-effort packet delivery protocol. IP packets can be lost, delayed, or received out of sequence. Each packet is routed on its own, using the IP header appended to the physical packet. IP address examples in this book use the current version of IP, Version 4. IPv4 addresses are four octets long, usually written in so-called "dotted decimal" notation (for example, 207.134.3.5). Between each of the dots is a decimal number between 0 and 255. At the IP layer, packets are not acknowledged. A checksum is calculated to detect corruption in the IP header, which could cause a packet to become misrouted. Corruption or errors in the IP payload, however, are not detected; a higher layer must perform this function if necessary. IP uses a single octet protocol number in the packet header to identify the transport layer protocol that should receive the packet. IP addresses used over the public Internet are assigned in blocks by the Internet Assigned

Number Association (IANA). As a result of this centralized assignment, IP addresses are globally unique. This enables a packet to be routed across the public Internet using only the destination IP address. Various protocols are used to route packets over an IP network, but they are outside of the scope of this book. Subnetting and other aspects of the structure of IP addresses are also not covered here. There are other excellent sources [6] that cover the entire suite of TCP/IP protocols in more detail...

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Table of Contents

Foreword xiii
Preface vii
1 SIP and the Internet 1
1.1 Signaling Protocols 1
1.2 The Internet Engineering Task Force 2
1.3 A Brief History of SIP 3
1.4 Internet Multimedia Protocol Stack 3
1.4.1 Physical Layer 4
1.4.2 Internet Layer 4
1.4.3 Transport Layer 5
1.4.4 Application Layer 7
1.5 Utility Applications 7
1.6 DNS and IP Addresses 8
1.7 URLs 10
1.8 Multicast 10
1.9 ABNF Representation 11
References 12
2 Introduction to SIP 15
2.1 A Simple SIP Example 15
2.2 SIP Call with Proxy Server 23
2.3 SIP Registration Example 28
2.4 Message Transport 30
2.4.1 UDP Transport 30
2.4.2 TCP Transport 31
References 33
3 SIP Clients and Servers 35
3.1 SIP User Agents 35
3.2 SIP Gateways 36
3.3 SIP Servers 39
3.3.1 Proxy Servers 39
3.3.2 Redirect Servers 42
3.3.3 Registration Servers 45
3.4 Acknowledgment of Messages 45
3.5 Reliability 46
3.6 Authentication 47
3.7 Encryption 48
3.8 Multicast Support 50
3.9 Firewalls and NAT Interaction 50
References 52
4 SIP Request Messages 53
4.1 Methods 53
4.1.1 Invite 54
4.1.2 Register 57
4.1.3 Bye 58
4.1.4 ACK 60
4.1.5 Cancel 63
4.1.6 Options 65
4.1.7 Info 66
4.1.8 Prack 68
4.2 SIP URLs and URIs 70
4.3 Tags 72
4.4 Message Bodies 73
References 74
5 SIP Response Messages 75
5.1 Informational 76
5.1.1 100 Trying 77
5.1.2 180 Ringing 77
5.1.3 181 Call Is Being Forwarded 77
5.1.4 182 Call Queued 78
5.1.5 183 Session Progress 78
5.2 Success 200 OK 79
5.3 Redirection 80
5.3.1 300 Multiple Choices 81
5.3.2 301 Moved Permanently 81
5.3.3 302 Moved Temporarily 81
5.3.4 305 Use Proxy 81
5.3.5 380 Alternative Service 82
5.4 Client Error 82
5.4.1 400 Bad Request 82
5.4.2 401 Unauthorized 82
5.4.3 402 Payment Required 83
5.4.4 403 Forbidden 83
5.4.5 404 Not Found 83
5.4.6 405 Method Not Allowed 83
5.4.7 406 Not Acceptable 84
5.4.8 407 Proxy Authentication Required 84
5.4.9 408 Request Timeout 84
5.4.10 409 Conflict 84
5.4.11 410 Gone 85
5.4.12 411 Length Required 85
5.4.13 413 Request Entity Too Large 85
5.4.14 414 Request-URI Too Long 85
5.4.15 415 Unsupported Media Type 85
5.4.16 420 Bad Extension 85
5.4.17 421 Extension Required 86
5.4.18 480 Temporarily Unavailable 86
5.4.19 481 Call Leg/Transaction Does Not Exist 86
5.4.20 482 Loop Detected 86
5.4.21 483 Too Many Hops 87
5.4.22 484 Address Incomplete 87
5.4.23 485 Ambiguous 88
5.4.24 486 Busy Here 89
5.4.25 487 Request Canceled 89
5.4.26 488 Not Acceptable Here 89
5.5 Server Error 89
5.5.1 500 Server Internal Error 90
5.5.2 501 Not Implemented 90
5.5.3 502 Bad Gateway 90
5.5.4 503 Service Unavailable 90
5.5.5 504 Gateway Timeout 90
5.5.6 505 Version Not Supported 90
5.6 Global Error 91
5.6.1 600 Busy Everywhere 91
5.6.2 603 Decline 91
5.6.3 604 Does Not Exist Anywhere 91
5.6.4 606 Not Acceptable 91
References 92
6 SIP Headers 93
6.1 General Headers 93
6.1.1 Call-ID 94
6.1.2 Contact 95
6.1.3 CSeq 96
6.1.4 Date 98
6.1.5 Encryption 98
6.1.6 From 99
6.1.7 Organization 99
6.1.8 Retry-After 100
6.1.9 Subject 100
6.1.10 Supported 101
6.1.11 Timestamp 101
6.1.12 To 101
6.1.13 User Agent 102
6.1.14 Via 102
6.2 Request Headers 104
6.2.1 Accept 104
6.2.2 Accept-Contact 105
6.2.3 Accept-Encoding 106
6.2.4 Accept-Language 106
6.2.5 Authorization 107
6.2.6 Hide 107
6.2.7 In-Reply-To 108
6.2.8 Max-Forwards 108
6.2.9 Priority 108
6.2.10 Proxy-Authorization 109
6.2.11 Proxy-Require 110
6.2.12 Record-Route 110
6.2.13 Reject-Contact 111
6.2.14 Request-Disposition 111
6.2.15 Require 111
6.2.16 Response-Key 112
6.2.17 Route 112
6.2.18 RAck 112
6.2.19 Session-Expires 112
6.3 Response Headers 113
6.3.1 Proxy-Authenticate 113
6.3.2 Server 113
6.3.3 Unsupported 113
6.3.4 Warning 114
6.3.5 WWW-Authenticate 114
6.3.6 RSeq 116
6.4 Entity Headers 116
6.4.1 Allow 117
6.4.2 Content-Encoding 117
6.4.3 Content-Disposition 117
6.4.4 Content-Length 117
6.4.5 Content-Type 118
6.4.6 Expires 118
6.4.7 Mime-Version 119
References 119
7 Related Protocols 121
7.1 SDP--Session Description Protocol 121
7.1.1 Protocol Version 124
7.1.2 Origin 124
7.1.3 Session Name and Information 124
7.1.4 URI 124
7.1.5 E-mail Address and Phone Number 124
7.1.6 Connection Data 125
7.1.7 Bandwidth 125
7.1.8 Time, Repeat Times, and Time Zones 125
7.1.9 Encryption Keys 126
7.1.10 Media Announcements 126
7.1.11 Attributes 127
7.1.12 Use of SDP in SIP 127
7.2 RTP--Real-time Transport Protocol 130
7.3 RTP Audio Video Profiles 133
7.4 PSTN Protocols 135
7.4.1 Circuit Associated Signaling 135
7.4.2 ISUP Signaling 135
7.4.3 ISDN Signaling 135
References 136
8 Comparison to H.323 137
8.1 Introduction to H.323 137
8.2 Example of H.323 139
8.3 Versions 144
8.4 Comparison 144
8.4.1 Encoding 145
8.4.2 Transport 147
8.4.3 Addressing 148
8.4.4 Complexity 148
8.4.5 Feature Implementations 149
8.4.6 Vendor Support 149
8.4.7 Conferencing 149
8.4.8 Extensibility 150
8.5 Comparison Summary 151
References 151
9 Call Flow Examples 153
9.1 SIP Call with Authentication, Proxies, and Record-Route 153
9.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy 161
9.3 SIP to PSTN Call Through Gateway 164
9.4 PSTN to SIP Call Through Gateway 169
9.5 Parallel Search 172
9.6 H.323 to SIP Call 177
References 183
10 Future Directions 185
10.1 Changes to RFC 2543 185
10.2 SIP Working Group Design Teams 186
10.2.1 Call Control 187
10.2.2 Convergence with PacketCable Distributed Call Signaling (DCS) Extensions 188
10.2.3 Call Flows 188
10.2.4 SIP/H.323 Interworking 188
10.2.5 Home Extension 188
10.2.6 SIP Security 188
10.2.7 SIP for Telephony 189
10.3 Other Related Drafts 189
References 189
About the Author 191
Index 193
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