S I P: Understanding the Session Initiation Protocol

by Johnston, Alan B.

• ISBN13: 9781580536554
• ISBN10: 1580536557
• Edition: 2nd
• Format: Hardcover
• Copyright: 2003-11-01
• Publisher: Artech House

Summary

This newly revised edition of the ground-breaking Artech House bestseller, SIP: Understanding the Session Initiation Protocol gives you a thorough and up-to-date understanding of this revolutionary protocol for call signaling and IP Telephony. The second edition includes brand new discussions on the use of SIP for wireless multimedia communications. It explains how SIP is powerful "rendezvous" protocol that leverages mobility and presence to allow users to communicate using different devices, modes, and services anywhere they are connected to the Internet You learn why SIP has been chosen by the 3GPP (3rd Generation Partnership Program for wireless cell phones) as the core signaling, presence, and instant messaging protocol.

Author Biography

Alan B. Johnston is a distinguished technical member at MCI and an adjunct assistant professor of electrical and systems engineering at Washington University in St. Louis, Missouri

Table of Contents

Foreword to the First Edition	xvii
Preface to the Second Edition	xix
Preface to the First Edition	xxi
1 SIP and the Internet	1	(16)
1.1 Signaling Protocols	1	(1)
1.2 The Internet Engineering Task Force	2	(1)
1.3 A Brief History of SIP	3	(1)
1.4 Internet Multimedia Protocol Stack	4	(5)
1.4.1 Physical Layer	4	(1)
1.4.2 Internet Layer	4	(1)
1.4.3 Transport Layer	5	(3)
1.4.4 Application Layer	8	(1)
1.5 Utility Applications	9	(1)
1.6 DNS and IP Addresses	10	(2)
1.7 URLs and URIs	12	(1)
1.8 Multicast	12	(1)
1.9 ABNF Representation	13	(1)
References	14	(3)
2 Introduction to SIP	17	(26)
2.1 A Simple Session Establishment Example	17	(8)
2.2 SIP Call with Proxy Server	25	(6)
2.3 SIP Registration Example	31	(2)
2.4 SIP Presence and Instant Message Example	33	(5)
2.5 Message Transport	38	(4)
2.5.1 UDP Transport	38	(2)
2.5.2 TCP Transport	40	(1)
2.5.3 TLS Transport	40	(1)
2.5.4 SCTP Transport	41	(1)
References	42	(1)
3 SIP Clients and Servers	43	(28)
3.1 SIP User Agents	43	(1)
3.2 Presence Agents	44	(1)
3.3 Back-to-Back User Agents	45	(1)
3.4 SIP Gateways	45	(2)
3.5 SIP Servers	47	(8)
3.5.1 Proxy Servers	47	(5)
3.5.2 Redirect Servers	52	(3)
3.5.3 Registration Servers	55	(1)
3.6 Acknowledgment of Messages	55	(1)
3.7 Reliability	56	(1)
3.8 Authentication	57	(2)
3.9 S/MIME Encryption	59	(1)
3.10 Multicast Support	60	(1)
3.11 Firewalls and NAT Interaction	61	(1)
3.12 Protocols and Extensions for NAT Traversal	62	(6)
3.12.1 STUN Protocol	63	(2)
3.12.2 TURN Protocol	65	(1)
3.12.3 Other SIP/SDP NAT-Related Extensions	66	(2)
References	68	(3)
4 SIP Request Messages	71	(36)
4.1 Methods	71	(27)
4.1.1 INVITE	72	(2)
4.1.2 REGISTER	74	(2)
4.1.3 BYE	76	(1)
4.1.4 ACK	77	(2)
4.1.5 CANCEL	79	(2)
4.1.6 OPTIONS	81	(1)
4.1.7 REFER	82	(4)
4.1.8 SUBSCRIBE	86	(3)
4.1.9 NOTIFY	89	(1)
4.1.10 MESSAGE	90	(3)
4.1.11 INFO	93	(1)
4.1.12 PRACK	94	(2)
4.1.13 UPDATE	96	(2)
4.2 URI and URL Schemes Used by SIP	98	(4)
4.2.1 SIP and SIPS URIs	98	(2)
4.2.2 Telephone URLs	100	(1)
4.2.3 Presence and Instant Messaging URLs	101	(1)
4.3 Tags	102	(1)
4.4 Message Bodies	102	(2)
References	104	(3)
5 SIP Response Messages	107	(20)
5.1 Informational	108	(4)
5.1.1 100 Trying	109	(1)
5.1.2 180 Ringing	109	(1)
5.1.3 181 Call Is Being Forwarded	109	(1)
5.1.4 182 Call Queued	109	(1)
5.1.5 183 Session Progress	110	(2)
5.2 Success	112	(1)
5.2.1 2000K	112	(1)
5.2.2 202 Accepted	112	(1)
5.3 Redirection	112	(1)
5.3.1 300 Multiple Choices	113	(1)
5.3.2 301 Moved Permanently	113	(1)
5.3.3 302 Moved Temporarily	113	(1)
5.3.4 305 Use Proxy	113	(1)
5.3.5 380 Alternative Service	113	(1)
5.4 Client Error	113	(10)
5.4.1 400 Bad Request	114	(1)
5.4.2 401 Unauthorized	114	(1)
5.4.3 402 Payment Required	114	(1)
5.4.4 403 Forbidden	115	(1)
5.4.5 404 Not Found	115	(1)
5.4.6 405 Method Not Allowed	115	(1)
5.4.7 406 Not Acceptable	115	(1)
5.4.8 407 Proxy Authentication Required	115	(1)
5.4.9 408 Request Timeout	116	(1)
5.4.10 409 Conflict	116	(1)
5.4.11 410 Gone	116	(1)
5.4.12 411 Length Required	116	(1)
5.4.13 413 Request Entity Too Large	117	(1)
5.4.14 414 Request-URI Too Long	117	(1)
5.4.15 415 Unsupported Media Type	117	(1)
5.4.16 416 Unsupported URI Scheme	117	(1)
5.4.17 420 Bad Extension	117	(1)
5.4.18 421 Extension Required	117	(1)
5.4.19 422 Session Timer Interval Too Small	118	(1)
5.4.20 423 Interval Too Brief	118	(1)
5.4.21 428 Use Authentication Token	118	(1)
5.4.22 429 Provide Referror Identity	118	(1)
5.4.23 480 Temporarily Unavailable	119	(1)
5.4.24 481 Dialog/Transaction Does Not Exist	119	(1)
5.4.25 482 Loop Detected	119	(1)
5.4.26 483 Too Many Hops	119	(1)
5.4.27 484 Address Incomplete	120	(1)
5.4.28 485 Ambiguous	120	(1)
5.4.29 486 Busy Here	121	(1)
5.4.30 487 Request Terminated	122	(1)
5.4.31 488 Not Acceptable Here	122	(1)
5.4.32 489 Bad Event	122	(1)
5.4.33 491 Request Pending	122	(1)
5.4.34 493 Request Undecipherable	122	(1)
5.5 Server Error	123	(2)
5.5.1 500 Server Internal Error	123	(1)
5.5.2 501 Not Implemented	124	(1)
5.5.3 502 Bad Gateway	124	(1)
5.5.4 503 Service Unavailable	124	(1)
5.5.5 504 Gateway Timeout	124	(1)
5.5.6 505 Version Not Supported	124	(1)
5.5.7 513 Message Too Large	125	(1)
5.6 Global Error	125	(1)
5.6.1 600 Busy Everywhere	125	(1)
5.6.2 603 Decline	125	(1)
5.6.3 604 Does Not Exist Anywhere	125	(1)
5.6.4 606 Not Acceptable	125	(1)
References	126	(1)
6 SIP Header Fields	127	(36)
6.1 Request and Response Header Fields	128	(12)
6.1.1 Alert-Info	128	(1)
6.1.2 Allow-Events	129	(1)
6.1.3 Call-ID	129	(1)
6.1.4 Contact	130	(2)
6.1.5 CSeq	132	(1)
6.1.6 Date	132	(1)
6.1.7 Encryption	133	(1)
6.1.8 From	133	(1)
6.1.9 Organization	134	(1)
6.1.10 Record-Route	134	(1)
6.1.11 Retry-After	135	(1)
6.1.12 Subject	135	(1)
6.1.13 Supported	136	(1)
6.1.14 Timestamp	136	(1)
6.1.15 To	137	(1)
6.1.16 User-Agent	137	(1)
6.1.17 Via	138	(2)
6.2 Request Header Fields	140	(13)
6.2.1 Accept	140	(1)
6.2.2 Accept-Contact	140	(1)
6.2.3 Accept-Encoding	141	(1)
6.2.4 Accept-Language	141	(1)
6.2.5 Authorization	142	(1)
6.2.6 Call-Info	142	(1)
6.2.7 Event	143	(1)
6.2.8 Hide	143	(1)
6.2.9 In-Reply-To	143	(1)
6.2.10 Join	143	(1)
6.2.11 Priority	144	(1)
6.2.12 Privacy	145	(1)
6.2.13 Proxy-Authorization	145	(1)
6.2.14 Proxy-Require	145	(1)
6.2.15 P-OSP-Auth-Token	145	(2)
6.2.16 P-Asserted-Identity	147	(1)
6.2.17 P-Preferred-Identity	147	(1)
6.2.18 Max-Forwards	147	(1)
6.2.19 Reason	147	(1)
6.2.20 Refer-To	148	(1)
6.2.21 Referred-By	148	(1)
6.2.22 Reply-To	149	(1)
6.2.23 Replaces	150	(1)
6.2.24 Reject-Contact	150	(1)
6.2.25 Request-Disposition	151	(1)
6.2.26 Require	151	(1)
6.2.27 Response-Key	152	(1)
6.2.28 Route	152	(1)
6.2.29 RAck	152	(1)
6.2.30 Session-Expires	153	(1)
6.2.31 Subscription-State	153	(1)
6.3 Response Header Fields	153	(5)
6.3.1 Authenticator-Info	153	(1)
6.3.2 Error-Info	154	(1)
6.3.3 Min-Expires	154	(1)
6.3.4 Min-SE	154	(1)
6.3.5 Proxy-Authenticate	155	(1)
6.3.6 Server	155	(1)
6.3.7 Unsupported	155	(1)
6.3.8 Warning	156	(1)
6.3.9 -Authenticate	156	(1)
6.3.10 RSeq	156	(2)
6.4 Message Body Header Fields	158	(2)
6.4.1 Allow	158	(1)
6.4.2 Content-Encoding	158	(1)
6.4.3 Content-Disposition	158	(1)
6.4.4 Content-Language	158	(1)
6.4.5 Content-Length	159	(1)
6.4.6 Content-Type	159	(1)
6.4.7 Expires	160	(1)
6.4.8 MIME-Version	160	(1)
References	160	(3)
7 Related Protocols	163	(18)
7.1 SDP-Session Description Protocol	163	(8)
7.1.1 Protocol Version	165	(1)
7.1.2 Origin	165	(1)
7.1.3 Session Name and Information	166	(1)
7.1.4 URI	166	(1)
7.1.5 E-Mail Address and Phone Number	166	(1)
7.1.6 Connection Data	166	(1)
7.1.7 Bandwidth	167	(1)
7.1.8 Time, Repeat Times, and Time Zones	167	(1)
7.1.9 Encryption Keys	167	(1)
7.1.10 Media Announcements	168	(1)
7.1.11 Attributes	168	(1)
7.1.12 Use of SDP in SIP	169	(2)
7.2 RTP-Real-Time Transport Protocol	171	(3)
7.3 RTP Audio Video Profiles	174	(2)
7.4 PSTN Protocols	176	(1)
7.4.1 Circuit Associated Signaling	176	(1)
7.4.2 ISUP Signaling	176	(1)
7.4.3 ISDN Signaling	176	(1)
7.5 SIP for Telephones	177	(1)
7.6 Universal Plug and Play Protocol	178	(3)
References
8 Comparison to H.323	181	(12)
8.1 Introduction to H.323	181	(3)
8.2 Example of H.323	184	(3)
8.3 Versions	187	(1)
8.4 Comparison	187	(4)
8.4.1 Fundamental Differences	188	(2)
8.4.2 Strengths of Each Protocol	190	(1)
8.5 Conclusion	191	(1)
References	191	(2)
9 Wireless and 3GPP	193	(14)
9.1 P Mobility	193	(1)
9.2 SIP Mobility	194	(7)
9.3 3GPP Architecture and SIP	201	(2)
9.4 3GPP Header Fields	203	(1)
9.4.1 Service-Route	203	(1)
9.4.2 Path	203	(1)
9.4.3 Other P-Headers	203	(1)
9.5 Future of SIP and Wireless	204	(1)
References	204	(3)
10 Call Flow Examples	207	(54)
10.1 SIP Call with Authentication, Proxies, and Record-Route	207	(7)
10.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy	214	(4)
10.3 SIP to PSTN Call Through Gateway	218	(4)
10.4 PSTN to SIP Call Through Gateway	222	(3)
10.5 Parallel Search	225	(5)
10.6 H.323 to SIP Call	230	(5)
10.7 3GPP Wireless Call Flow	235	(19)
10.8 Call Setup Example with Two Proxies	254	(2)
10.9 SIP Presence and Instant Message Example	256	(3)
References	259	(2)
11 Future Directions	261	(6)
11.1 SIP, SIPPING, and SIMPLE Working Group Design Teams	261	(2)
11.1.1 SIP and Hearing Impairment Design Team	262	(1)
11.1.2 Conferencing Design Team	262	(1)
11.1.3 Application Interaction Design Team	263	(1)
11.1.4 Emergency Calling Design Team	263	(1)
11.2 Other SIP Work Areas	263	(1)
11.2.1 Emergency Preparedness	263	(1)
11.2.2 Globally Routable Contact URIs	263	(1)
11.2.3 Service Examples	263	(1)
11.3 SIP Instant Message and Presence Work	264	(1)
References	264	(3)
Appendix A: Changes in the SIP Specification from RFC 2543 to RFC 3261	267	(4)
About the Author	271	(2)
Index	273