SIP: Understanding the Session Initiation Protocol, Third Edition (Artech House Telecommunications)

SIP: Understanding the Session Initiation Protocol is a pick for any communications engineering library at the college level. It appears in its third edition and provides details on call signaling and IP telephony, and has been expanded with numerous new chapters covering peer-to-peer SIP, ABNF and XML, and other details key to telecommunications network development and SIP uses in Internet architecture. SIP: Understanding the Session Initiation Protocol Third Edition......Page 2 Contents......Page 8 Foreword to the First Edition......Page 22 Preface to the Third Edition......Page 24 Preface to the Second Edition......Page 26 Preface to the First Edition......Page 28 1.1 Signaling Protocols......Page 32 1.2.2 Data/Link Layer......Page 33 1.2.3 Network Layer......Page 34 1.2.4 Transport Layer......Page 35 1.2.6 Utility Applications......Page 40 1.2.7 Multicast......Page 41 1.4 URLs, URIs, and URNs......Page 42 1.5 Domain Name Service......Page 44 1.5.1 DNS Resource Records......Page 45 1.5.3 Service Resource Records (SRV)......Page 46 1.5.5 DNS Resolvers......Page 47 1.6 Global Open Standards......Page 48 1.7 Internet Standards Process......Page 49 1.8 A Brief History of SIP......Page 51 References......Page 52 2.1 A Simple Session Establishment Example......Page 54 2.2 SIP Call with a Proxy Server......Page 62 2.3 SIP Registration Example......Page 67 2.4 SIP Presence and Instant Message Example......Page 69 2.5.1 UDP Transport......Page 74 2.5.2 TCP Transport......Page 76 2.5.4 SCTP Transport......Page 77 2.6 Transport Protocol Selection......Page 78 2.8 Questions......Page 79 References......Page 81 3.1 SIP User Agents......Page 82 3.2 Presence Agents......Page 83 3.3 Back-to-Back User Agents......Page 84 3.4 SIP Gateways......Page 85 3.5.1 Proxy Servers......Page 87 3.5.2 Redirect Servers......Page 92 3.5.3 Registrar Servers......Page 94 3.6 Uniform Resource Indicators......Page 95 3.7 Acknowledgment of Messages......Page 96 3.8 Reliability......Page 97 3.9 Multicast Support......Page 99 3.11 Questions......Page 100 References......Page 103 4.1.1 INVITE......Page 104 4.1.2 REGISTER......Page 107 4.1.4 ACK......Page 109 4.1.5 CANCEL......Page 112 4.1.6 OPTIONS......Page 113 4.1.7 SUBSCRIBE......Page 115 4.1.8 NOTIFY......Page 118 4.1.9 PUBLISH......Page 119 4.1.10 REFER......Page 122 4.1.11 MESSAGE......Page 125 4.1.12 INFO......Page 127 4.1.13 PRACK......Page 128 4.1.14 UPDATE......Page 130 4.2 URI and URL Schemes Used by SIP......Page 131 4.2.1 SIP and SIPS URIs......Page 132 4.2.2 Telephone URLs......Page 133 4.3 Tags......Page 135 4.4 Message Bodies......Page 136 4.6 Questions......Page 138 References......Page 139 5 SIP Response Messages......Page 142 5.1.1 100 Trying......Page 143 5.1.5 183 Session Progress......Page 144 5.2.1 200 OK......Page 145 5.3.1 300 Multiple Choices......Page 146 5.4.1 400 Bad Request......Page 147 5.4.4 403 Forbidden......Page 148 5.4.8 407 Proxy Authentication Required......Page 149 5.4.13 412 Conditional Request Failed......Page 150 5.4.18 417 Unknown Resource Priority......Page 151 5.4.23 428 Use Identity Header......Page 152 5.4.28 437 Unsupported Certifi cate......Page 153 5.4.34 481 Dialog/Transaction Does Not Exist......Page 154 5.4.36 483 Too Many Hops......Page 155 5.4.38 485 Ambiguous......Page 156 5.4.43 491 Request Pending......Page 157 5.4.45 494 Security Agreement Required......Page 158 5.5.5 504 Gateway Timeout......Page 159 5.6.2 603 Decline......Page 160 5.7 Questions......Page 161 References......Page 162 6 SIP Header Fields......Page 164 6.1.2 Accept-Encoding......Page 165 6.1.4 Alert-Info......Page 167 6.1.8 Call-ID......Page 168 6.1.9 Contact......Page 169 6.1.10 CSeq......Page 171 6.1.14 From......Page 172 6.1.15 History Info......Page 173 6.1.18 Priv-Answer-Mod......Page 174 6.1.21 Refer-Sub......Page 175 6.1.23 Subject......Page 176 6.1.24 Supported......Page 177 6.1.27 User-Agent......Page 178 6.1.28 Via......Page 179 6.2.1 Accept-Contact......Page 180 6.2.4 Event......Page 181 6.2.8 In-Reply-To......Page 182 6.2.10 Join......Page 183 6.2.13 Proxy-Authorization......Page 184 6.2.14 Proxy-Require......Page 185 6.2.18 Max-Breadth......Page 186 6.2.21 Refer-To......Page 187 6.2.23 Reply-To......Page 188 6.2.25 Reject-Contact......Page 189 6.2.27 Require......Page 190 6.2.30 Route......Page 191 6.2.32 Security-Client......Page 192 6.2.36 Subscription-State......Page 193 6.3.1 Accept-Resource-Priority......Page 194 6.3.3 Error-Info......Page 195 6.3.7 Permission-Missing......Page 196 6.3.11 Service-Route......Page 197 6.3.14 Warning......Page 198 6.3.16 RSeq......Page 199 6.4.2 Content-Disposition......Page 200 6.4.5 Content-Type......Page 201 6.5 Questions......Page 202 References......Page 203 7.1 IP Mobility......Page 208 7.2 SIP Mobility......Page 209 7.3 IMS and SIP......Page 215 7.5 Conclusion......Page 217 References......Page 218 8.2 History of IM and Presence......Page 220 8.4.1 SIP Events Framework......Page 222 8.4.2 Presence Bodies......Page 223 8.4.3 Resource Lists......Page 225 8.4.4 Filtering......Page 231 8.4.5 Conditional Event Notifi cations and ETags......Page 232 8.4.6 Partial Publication......Page 233 8.4.7 Presence Documents Summary......Page 235 8.5.2 Common Profi le for Instant Messaging......Page 236 8.5.3 Instant Messaging Delivery Notifi cation......Page 237 8.5.4 Message Composition Indication......Page 239 8.5.5 Multiple Recipient Messages......Page 240 8.5.6 Session Mode Instant Messaging......Page 241 8.6.1 Standardization as Extensible Messaging and Presence Protocol......Page 244 8.7 Conclusion......Page 245 8.8 Questions......Page 246 References......Page 247 9.1 Gateway Services......Page 250 9.3 SIP Service Examples......Page 252 9.4 Voicemail......Page 254 9.5 SIP Video......Page 256 9.6 Facsimile......Page 257 9.7.1 Focus......Page 258 9.7.3 Non-SIP Conference Control......Page 259 9.8 Application Sequencing......Page 260 9.9 Other SIP Service Architectures......Page 261 9.9.3 Service Delivery Platform......Page 262 References......Page 263 10.1 Introduction to NAT......Page 266 10.2 Advantages of NAT......Page 267 10.3 Disadvantages of NAT......Page 268 10.4 How NAT Works......Page 269 10.5 Types of NAT......Page 270 10.5.2 Address Dependent Mapping NAT......Page 271 10.5.4 Hairpinning Support......Page 272 10.5.7 Mapping Refresh......Page 273 10.5.8 Filtering Modes......Page 274 10.6 NAT Mapping Examples......Page 275 10.7 NATs and SIP......Page 276 10.8 Properties of a Friendly NAT or How a NAT Should BEHAVE......Page 278 10.9 STUN Protocol......Page 279 10.11 SIP Problems with NAT......Page 280 10.11.2 Connection Reuse......Page 281 10.12.1 Symmetric RTP......Page 282 10.13 Hole Punching......Page 284 10.14 TURN: Traversal Using Relays Around NAT......Page 288 10.15 ICE: Interactive Connectivity Establishment......Page 289 10.16 Conclusion......Page 290 10.17 Questions......Page 291 References......Page 292 11.1.1 Circuit Associated Signaling......Page 294 11.2 SIP for Telephones......Page 295 11.3 Media Gateway Control Protocols......Page 296 11.4.1 Introduction to H.323......Page 297 11.4.2 Example of H.323......Page 299 References......Page 302 12.1 Real-Time Transport Protocol (RTP)......Page 304 12.2 RTP Control Protocol (RTCP)......Page 309 12.2.2 RTCP Extended Reports......Page 310 12.3 Compression......Page 311 12.4 RTP Audio Video Profi les......Page 312 12.4.1 Audio Codecs......Page 313 12.4.2 Video Codecs......Page 314 12.5 Conferencing......Page 315 12.7 DTMF Transport......Page 316 12.8 Questions......Page 317 References......Page 318 13.1 Session Description Protocol (SDP)......Page 320 13.1.2 Origin......Page 322 13.1.6 Connection Data......Page 323 13.1.10 Media Announcements......Page 324 13.1.11 Attributes......Page 325 13.2 SDP Extensions......Page 327 13.3 The Offer Answer Model......Page 328 13.3.4 Special Case—Call Hold......Page 330 13.5 SIP Offer Answer Exchanges......Page 331 13.7 Questions......Page 332 References......Page 335 14.1 Basic Security Concepts......Page 338 14.1.1 Encryption......Page 339 14.1.4 Message Authentication......Page 340 14.1.5 Digital Certifi cates......Page 341 14.2 Threats......Page 342 14.3.1 IPSec......Page 343 14.3.3 DNSSec......Page 344 14.4.1 SIP Digest Authentication......Page 345 14.4.2 SIP Authentication Using TLS......Page 347 14.4.4 Identity......Page 348 14.4.5 Enhanced SIP Identity......Page 349 14.5 SIP Certifi cate Service......Page 350 14.6.1 Non-RTP Media......Page 353 14.6.3 Keying SRTP......Page 354 14.6.4 Best Effort Encryption......Page 356 14.6.5 ZRTP......Page 357 14.7 Questions......Page 358 References......Page 359 15.1 P2P Properties......Page 362 15.2 P2P Properties of SIP......Page 363 15.3 P2P Overlays......Page 364 15.4 RELOAD......Page 367 15.5 Host Identity Protocol......Page 369 15.6 Conclusion......Page 370 15.7 Questions......Page 371 References......Page 372 16.1 SIP Call with Authentication, Proxies, and Record-Route......Page 374 16.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy......Page 380 16.3 SIP to PSTN Call Through Gateways......Page 383 16.4 PSTN to SIP Call Through a Gateway......Page 387 16.5 Parallel Search......Page 390 16.6 Call Setup with Two Proxies......Page 394 16.7 SIP Presence and Instant Message Example......Page 396 References......Page 399 17 Future Directions......Page 400 17.2 More Extensions......Page 401 17.5 Making Features Work Better......Page 402 17.10 Security Deployment......Page 403 References......Page 404 A.1 ABNF Rules......Page 406 A.2 Introduction to XML......Page 408 References......Page 410 About the Author......Page 412 Index......Page 414

Now in its third edition, the ground-breaking Artech House bestseller SIP: Understanding the Session Initiation Protocol offers you the most comprehensive and current understanding of this revolutionary protocol for call signaling and IP Telephony. The third edition has been significantly expanded with brand new chapters on NAT traversal, SIP security, services in SIP, presence and instant messaging, Peer-to-Peer SIP, and an introduction to ABNF and XML. This cutting-edge book shows you 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. Other key 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. Moreover, this book covers SIP RFC 3261 and the complete set of SIP extension RFCs.

Now in its third edition, the ground-breaking Artech House bestseller " Understanding the Session Initiation Protocol" offers the most comprehensive and current understanding of this revolutionary protocol for call signaling and IP Telephony. Essential reading for anyone involved in the development and operation of voice or data networks, the third edition has been significantly expanded with brand new chapters on SIP security, services in SIP, presence and instant messaging, and Peer-to-Peer SIP. 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. This title is suitable for telecommunications/wireless engineers and professionals, professors and students whose work involves voice or data networks and the Internet multimedia architecture.