This newly revised edition of the ground-breaking Artech House bestseller, SIP: Understanding the Session Initiation Protocol offers a thorough and up-to-date understanding of this revolutionary technology for IP Telephony. Essential reading for anyone involved in the development and operation of voice or data networks, the second edition includes brand new discussions on the use of SIP as a wireless communications protocol and mobility technology. Professionals find details on the latest application areas such as instant messaging.The book explains how SIP is a highly-scalable and cost-effective way to offer new and exciting telecommunication feature sets. From an examination of SIP as a key component in the Internet multimedia conferencing architecture: to a look at the future direction of SIP, practitioners get the knowledge they need to design "next generation" networks and develop new applications and software stacks.
Author(s): Alan B. Johnston
Edition: 2
Publisher: Artech House Publishers
Year: 2003
Language: English
Pages: 309
SIP
Understanding the Session Initiation Protocol......Page 5
Copyright......Page 6
Contents......Page 9
Foreword to the First Edition......Page 19
Preface to the Second Edition......Page 21
Preface to the First Edition......Page 23
1.1 Signaling Protocols......Page 27
1.2 The Internet Engineering Task Force......Page 28
1.3 A Brief History of SIP......Page 29
1.4.2 Internet Layer......Page 30
1.4.3 Transport Layer......Page 31
1.4.4 Application Layer......Page 34
1.5 Utility Applications......Page 35
1.6 DNS and IP Addresses......Page 36
1.8 Multicast......Page 38
1.9 ABNF Representation......Page 39
References......Page 40
2.1 A Simple Session Establishment Example......Page 43
2.2 SIP Call with Proxy Server......Page 51
2.3 SIP Registration Example......Page 57
2.4 SIP Presence and Instant Message Example......Page 59
2.5.1 UDP Transport......Page 64
2.5.3 TLS Transport......Page 66
2.5.4 SCTP Transport......Page 67
References......Page 68
3.1 SIP User Agents......Page 69
3.2 Presence Agents......Page 70
3.4 SIP Gateways......Page 71
3.5.1 Proxy Servers......Page 73
3.5.2 Redirect Servers......Page 78
3.6 Acknowledgment of Messages......Page 81
3.7 Reliability......Page 82
3.8 Authentication......Page 83
3.9 S/ MIME Encryption......Page 85
3.10 Multicast Support......Page 86
3.11 Firewalls and NAT Interaction......Page 87
3.12 Protocols and Extensions for NAT Traversal......Page 88
3.12.1 STUN Protocol......Page 89
3.12.2 TURN Protocol......Page 91
3.12.3 Other SIP/ SDP NAT- Related Extensions......Page 92
References......Page 94
4.1 Methods......Page 97
4.1.1 INVITE......Page 98
4.1.2 REGISTER......Page 100
4.1.3 BYE......Page 102
4.1.4 ACK......Page 103
4.1.5 CANCEL......Page 105
4.1.6 OPTIONS......Page 107
4.1.7 REFER......Page 108
4.1.8 SUBSCRIBE......Page 112
4.1.9 NOTIFY......Page 115
4.1.10 MESSAGE......Page 116
4.1.11 INFO......Page 119
4.1.12 PRACK......Page 120
4.1.13 UPDATE......Page 122
4.2.1 SIP and SIPS URIs......Page 124
4.2.2 Telephone URLs......Page 126
4.2.3 Presence and Instant Messaging URLs......Page 127
4.4 Message Bodies......Page 128
References......Page 130
5 SIP Response Messages......Page 133
5.1 Informational......Page 134
5.1.4 182 Call Queued......Page 135
5.1.5 183 Session Progress......Page 136
5.3 Redirection......Page 138
5.4 Client Error......Page 139
5.4.3 402 Payment Required......Page 140
5.4.8 407 Proxy Authentication Required......Page 141
5.4.12 411 Length Required......Page 142
5.4.18 421 Extension Required......Page 143
5.4.22 429 Provide Referror Identity......Page 144
5.4.26 483 Too Many Hops......Page 145
5.4.28 485 Ambiguous......Page 146
5.4.29 486 Busy Here......Page 147
5.4.34 493 Request Undecipherable......Page 148
5.5.1 500 Server Internal Error......Page 149
5.5.6 505 Version Not Supported......Page 150
5.6.4 606 Not Acceptable......Page 151
References......Page 152
6 SIP Header Fields......Page 153
6.1.1 Alert- Info......Page 154
6.1.3 Call- ID......Page 155
6.1.4 Contact......Page 156
6.1.6 Date......Page 158
6.1.8 From......Page 159
6.1.10 Record- Route......Page 160
6.1.12 Subject......Page 161
6.1.14 Timestamp......Page 162
6.1.16 User- Agent......Page 163
6.1.17 Via......Page 164
6.2.2 Accept- Contact......Page 166
6.2.4 Accept- Language......Page 167
6.2.6 Call- Info......Page 168
6.2.10 Join......Page 169
6.2.11 Priority......Page 170
6.2.15 P- OSP- Auth- Token......Page 171
6.2.19 Reason......Page 173
6.2.21 Referred- By......Page 174
6.2.22 Reply- To......Page 175
6.2.24 Reject- Contact......Page 176
6.2.26 Require......Page 177
6.2.29 RAck......Page 178
6.3.1 Authenticaton- Info......Page 179
6.3.4 Min- SE......Page 180
6.3.7 Unsupported......Page 181
6.3.10 RSeq......Page 182
6.4.4 Content- Language......Page 184
6.4.6 Content- Type......Page 185
References......Page 186
7.1 SDP ¡ª Session Description Protocol......Page 189
7.1.2 Origin......Page 191
7.1.6 Connection Data......Page 192
7.1.9 Encryption Keys......Page 193
7.1.11 Attributes......Page 194
7.1.12 Use of SDP in SIP......Page 195
7.2 RTP ¡ª Real- Time Transport Protocol......Page 197
7.3 RTP Audio Video Profiles......Page 200
7.4.3 ISDN Signaling......Page 202
7.5 SIP for Telephones......Page 203
References......Page 204
8.1 Introduction to H. 323......Page 207
8.2 Example of H. 323......Page 210
8.4 Comparison......Page 213
8.4.1 Fundamental Differences......Page 214
8.4.2 Strengths of Each Protocol......Page 216
References......Page 217
9.1 IP Mobility......Page 219
9.2 SIP Mobility......Page 220
9.3 3GPP Architecture and SIP......Page 227
9.4.3 Other P- Headers......Page 229
References......Page 230
10.1 SIP Call with Authentication, Proxies, and Record- Route......Page 233
10.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy......Page 240
10.3 SIP to PSTN Call Through Gateway......Page 244
10.4 PSTN to SIP Call Through Gateway......Page 248
10.5 Parallel Search......Page 251
10.6 H. 323 to SIP Call......Page 256
10.7 3GPP Wireless Call Flow......Page 261
10.8 Call Setup Example with Two Proxies......Page 280
10.9 SIP Presence and Instant Message Example......Page 282
References......Page 285
11.1 SIP, SIPPING, and SIMPLE Working Group Design Teams......Page 287
11.1.2 Conferencing Design Team......Page 288
11.2.3 Service Examples......Page 289
References......Page 290
Appendix A: Changes in the SIP Specification from RFC 2543 to RFC 3261......Page 293
About the Author......Page 297
Index......Page 299
