S I P: Understanding the Session Initiation Protocol,9781580536554

S I P: Understanding the Session Initiation Protocol

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Edition: 2nd
ISBN13: 9781580536554
ISBN10: 1580536557
Format: Hardcover
Pub. Date: 2003-11-01
Publisher(s): Artech House
List Price: $99.00

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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

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