Cover image for SIP : Understanding the Session Initiation Protocol, Third Edition.
SIP : Understanding the Session Initiation Protocol, Third Edition.
Title:
SIP : Understanding the Session Initiation Protocol, Third Edition.
Author:
Johnston, Alan B.
ISBN:
9781607839965
Personal Author:
Edition:
3rd ed.
Physical Description:
1 online resource (426 pages)
Contents:
SIP: Understanding the Session Initiation Protocol Third Edition -- Contents -- Foreword to the First Edition -- Preface to the Third Edition -- Preface to the Second Edition -- Preface to the First Edition -- 1 SIP and the Internet -- 1.1 Signaling Protocols -- 1.2 Internet Multimedia Protocol Stack -- 1.2.1 Physical Layer -- 1.2.2 Data/Link Layer -- 1.2.3 Network Layer -- 1.2.4 Transport Layer -- 1.2.5 Application Layer -- 1.2.6 Utility Applications -- 1.2.7 Multicast -- 1.3 Internet Names -- 1.4 URLs, URIs, and URNs -- 1.5 Domain Name Service -- 1.5.1 DNS Resource Records -- 1.5.2 Address Resource Records (A or AAAA) -- 1.5.3 Service Resource Records (SRV) -- 1.5.4 Naming Authority Pointer Resource Records (NAPTR) -- 1.5.5 DNS Resolvers -- 1.6 Global Open Standards -- 1.7 Internet Standards Process -- 1.8 A Brief History of SIP -- 1.9 Conclusion -- References -- 2 Introduction to SIP -- 2.1 A Simple Session Establishment Example -- 2.2 SIP Call with a Proxy Server -- 2.3 SIP Registration Example -- 2.4 SIP Presence and Instant Message Example -- 2.5 Message Transport -- 2.5.1 UDP Transport -- 2.5.2 TCP Transport -- 2.5.3 TLS Transport -- 2.5.4 SCTP Transport -- 2.6 Transport Protocol Selection -- 2.7 Conclusion -- 2.8 Questions -- References -- 3 SIP Clients and Servers -- 3.1 SIP User Agents -- 3.2 Presence Agents -- 3.3 Back-to-Back User Agents -- 3.4 SIP Gateways -- 3.5 SIP Servers -- 3.5.1 Proxy Servers -- 3.5.2 Redirect Servers -- 3.5.3 Registrar Servers -- 3.6 Uniform Resource Indicators -- 3.7 Acknowledgment of Messages -- 3.8 Reliability -- 3.9 Multicast Support -- 3.10 Conclusion -- 3.11 Questions -- References -- 4 SIP Request Messages -- 4.1 Methods -- 4.1.1 INVITE -- 4.1.2 REGISTER -- 4.1.3 BYE -- 4.1.4 ACK -- 4.1.5 CANCEL -- 4.1.6 OPTIONS -- 4.1.7 SUBSCRIBE -- 4.1.8 NOTIFY -- 4.1.9 PUBLISH -- 4.1.10 REFER -- 4.1.11 MESSAGE.

4.1.12 INFO -- 4.1.13 PRACK -- 4.1.14 UPDATE -- 4.2 URI and URL Schemes Used by SIP -- 4.2.1 SIP and SIPS URIs -- 4.2.2 Telephone URLs -- 4.2.3 Presence and Instant Messaging URLs -- 4.3 Tags -- 4.4 Message Bodies -- 4.5 Conclusion -- 4.6 Questions -- References -- 5 SIP Response Messages -- 5.1 Informational -- 5.1.1 100 Trying -- 5.1.2 180 Ringing -- 5.1.3 181 Call is Being Forwarded -- 5.1.4 182 Call Queued -- 5.1.5 183 Session Progress -- 5.2 Success -- 5.2.1 200 OK -- 5.2.2 202 Accepted -- 5.2.3 204 No Notifi cation -- 5.3 Redirection -- 5.3.1 300 Multiple Choices -- 5.3.2 301 Moved Permanently -- 5.3.3 302 Moved Temporarily -- 5.3.4 305 Use Proxy -- 5.3.5 380 Alternative Service -- 5.4 Client Error -- 5.4.1 400 Bad Request -- 5.4.2 401 Unauthorized -- 5.4.3 402 Payment Required -- 5.4.4 403 Forbidden -- 5.4.5 404 Not Found -- 5.4.6 405 Method Not Allowed -- 5.4.7 406 Not Acceptable -- 5.4.8 407 Proxy Authentication Required -- 5.4.9 408 Request Timeout -- 5.4.10 409 Confl ict -- 5.4.11 410 Gone -- 5.4.12 411 Length Required -- 5.4.13 412 Conditional Request Failed -- 5.4.14 413 Request Entity Too Large -- 5.4.15 414 Request-URI Too Long -- 5.4.16 415 Unsupported Media Type -- 5.4.17 416 Unsupported URI Scheme -- 5.4.18 417 Unknown Resource Priority -- 5.4.19 420 Bad Extension -- 5.4.20 421 Extension Required -- 5.4.21 422 Session Timer Interval Too Small -- 5.4.22 423 Interval Too Brief -- 5.4.23 428 Use Identity Header -- 5.4.24 429 Provide Referror Identity -- 5.4.25 430 Flow Failed -- 5.4.26 433 Anonymity Disallowed -- 5.4.27 436 Bad Identity-Info Header -- 5.4.28 437 Unsupported Certifi cate -- 5.4.29 438 Invalid Identity Header -- 5.4.30 439 First Hop Lacks Outbound Support -- 5.4.31 440 Max-Breadth Exceeded -- 5.4.32 470 Consent Needed -- 5.4.33 480 Temporarily Unavailable -- 5.4.34 481 Dialog/Transaction Does Not Exist.

5.4.35 482 Loop Detected -- 5.4.36 483 Too Many Hops -- 5.4.37 484 Address Incomplete -- 5.4.38 485 Ambiguous -- 5.4.39 486 Busy Here -- 5.4.40 487 Request Terminated -- 5.4.41 488 Not Acceptable Here -- 5.4.42 489 Bad Event -- 5.4.43 491 Request Pending -- 5.4.44 493 Request Undecipherable -- 5.4.45 494 Security Agreement Required -- 5.5 Server Error -- 5.5.1 500 Server Internal Error -- 5.5.2 501 Not Implemented -- 5.5.3 502 Bad Gateway -- 5.5.4 503 Service Unavailable -- 5.5.5 504 Gateway Timeout -- 5.5.6 505 Version Not Supported -- 5.5.7 513 Message Too Large -- 5.5.8 580 Preconditions Failure -- 5.6 Global Error -- 5.6.1 600 Busy Everywhere -- 5.6.2 603 Decline -- 5.6.3 604 Does Not Exist Anywhere -- 5.6.4 606 Not Acceptable -- 5.7 Questions -- References -- 6 SIP Header Fields -- 6.1 Request and Response Header Fields -- 6.1.1 Accept -- 6.1.2 Accept-Encoding -- 6.1.3 Accept-Language -- 6.1.4 Alert-Info -- 6.1.5 Allow -- 6.1.6 Allow-Events -- 6.1.7 Answer-Mode -- 6.1.8 Call-ID -- 6.1.9 Contact -- 6.1.10 CSeq -- 6.1.11 Date -- 6.1.12 Encryption -- 6.1.13 Expires -- 6.1.14 From -- Untitled -- 6.1.15 History Info -- 6.1.16 Organization -- 6.1.17 Path -- 6.1.18 Priv-Answer-Mod -- 6.1.19 Record-Route -- 6.1.20 Recv-Info -- 6.1.21 Refer-Sub -- 6.1.22 Retry-After -- 6.1.23 Subject -- 6.1.24 Supported -- 6.1.25 Timestamp -- 6.1.26 To -- 6.1.27 User-Agent -- 6.1.28 Via -- 6.2 Request Header Fields -- 6.2.1 Accept-Contact -- 6.2.2 Authorization -- 6.2.3 Call-Info -- 6.2.4 Event -- 6.2.5 Hide -- 6.2.6 Identity -- 6.2.7 Identity-Info -- 6.2.8 In-Reply-To -- 6.2.9 Info-Package -- 6.2.10 Join -- 6.2.11 Priority -- 6.2.12 Privacy -- 6.2.13 Proxy-Authorization -- 6.2.14 Proxy-Require -- 6.2.15 P-OSP-Auth-Token -- 6.2.16 P-Asserted-Identity -- 6.2.17 P-Preferred-Identity -- 6.2.18 Max-Breadth -- 6.2.19 Max-Forwards -- 6.2.20 Reason -- 6.2.21 Refer-To.

6.2.22 Referred-By -- 6.2.23 Reply-To -- 6.2.24 Replaces -- 6.2.25 Reject-Contact -- 6.2.26 Request-Disposition -- 6.2.27 Require -- 6.2.28 Resource-Priority -- 6.2.29 Response-Key -- 6.2.30 Route -- 6.2.31 RAck -- 6.2.32 Security-Client -- 6.2.33 Security-Verify -- 6.2.34 Session-Expires -- 6.2.35 SIP-If-Match -- 6.2.36 Subscription-State -- 6.2.37 Suppress-If-Match -- 6.2.38 Target-Dialog -- 6.2.39 Trigger-Consent -- 6.3 Response Header Fields -- 6.3.1 Accept-Resource-Priority -- 6.3.2 Authentication-Info -- 6.3.3 Error-Info -- 6.3.4 Flow-Timer -- 6.3.5 Min-Expires -- 6.3.6 Min-SE -- 6.3.7 Permission-Missing -- 6.3.8 Proxy-Authenticate -- 6.3.9 Security-Server -- 6.3.10 Server -- 6.3.11 Service-Route -- 6.3.12 SIP-ETag -- 6.3.13 Unsupported -- 6.3.14 Warning -- 6.3.15 WWW-Authenticate -- 6.3.16 RSeq -- 6.4 Message Body Header Fields -- 6.4.1 Content-Encoding -- 6.4.2 Content-Disposition -- 6.4.3 Content-Language -- 6.4.4 Content-Length -- 6.4.5 Content-Type -- 6.4.6 MIME-Version -- 6.5 Questions -- References -- 7 Wireless, Mobility, and IMS -- 7.1 IP Mobility -- 7.2 SIP Mobility -- 7.3 IMS and SIP -- 7.4 IMS Header Fields -- 7.5 Conclusion -- 7.6 Questions -- References -- 8 Presence and Instant Messaging -- 8.1 Introduction -- 8.2 History of IM and Presence -- 8.3 SIMPLE -- 8.4 Presence with SIMPLE -- 8.4.1 SIP Events Framework -- 8.4.2 Presence Bodies -- 8.4.3 Resource Lists -- 8.4.4 Filtering -- 8.4.5 Conditional Event Notifi cations and ETags -- 8.4.6 Partial Publication -- 8.4.7 Presence Documents Summary -- 8.5 Instant Messaging with SIMPLE -- 8.5.1 Page Mode Instant Messaging -- 8.5.2 Common Profi le for Instant Messaging -- 8.5.3 Instant Messaging Delivery Notifi cation -- 8.5.4 Message Composition Indication -- 8.5.5 Multiple Recipient Messages -- 8.5.6 Session Mode Instant Messaging -- 8.6 Jabber.

8.6.1 Standardization as Extensible Messaging and Presence Protocol -- 8.6.2 Interworking with SIMPLE -- 8.6.3 Jingle -- 8.6.4 Future Standardization of XMPP -- 8.7 Conclusion -- 8.8 Questions -- References -- 9 Services in SIP -- 9.1 Gateway Services -- 9.2 SIP Trunking -- 9.3 SIP Service Examples -- 9.4 Voicemail -- 9.5 SIP Video -- 9.6 Facsimile -- 9.7 Conferencing -- 9.7.1 Focus -- 9.7.2 Mixer -- 9.7.3 Non-SIP Conference Control -- 9.8 Application Sequencing -- 9.9 Other SIP Service Architectures -- 9.9.1 Service Oriented Architecture -- 9.9.2 Servlets -- 9.9.3 Service Delivery Platform -- 9.10 Conclusion -- 9.11 Questions -- References -- 10 Network Address Translation -- 10.1 Introduction to NAT -- 10.2 Advantages of NAT -- 10.3 Disadvantages of NAT -- 10.4 How NAT Works -- 10.5 Types of NAT -- 10.5.1 Endpoint Independent Mapping NAT -- 10.5.2 Address Dependent Mapping NAT -- 10.5.3 Address and Port Dependent Mapping NAT -- 10.5.4 Hairpinning Support -- 10.5.5 IP Address Pooling Options -- 10.5.6 Port Assignment Options -- 10.5.7 Mapping Refresh -- 10.5.8 Filtering Modes -- 10.6 NAT Mapping Examples -- 10.7 NATs and SIP -- 10.8 Properties of a Friendly NAT or How a NAT Should BEHAVE -- 10.9 STUN Protocol -- 10.10 UNSAF Requirements -- 10.11 SIP Problems with NAT -- 10.11.1 Symmetric SIP -- 10.11.2 Connection Reuse -- 10.11.3 SIP Outbound -- 10.12 Media NAT Traversal Solutions -- 10.12.1 Symmetric RTP -- 10.12.2 RTCP Attribute -- 10.12.3 Self-Fixing Approach -- 10.13 Hole Punching -- 10.14 TURN: Traversal Using Relays Around NAT -- 10.15 ICE: Interactive Connectivity Establishment -- 10.16 Conclusion -- 10.17 Questions -- References -- 11 Related Protocols -- 11.1 PSTN Protocols -- 11.1.1 Circuit Associated Signaling -- 11.1.2 ISDN Signaling -- 11.1.3 ISUP Signaling -- 11.2 SIP for Telephones -- 11.3 Media Gateway Control Protocols -- 11.4 H.323.

11.4.1 Introduction to H.323.
Abstract:
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.
Local Note:
Electronic reproduction. Ann Arbor, Michigan : ProQuest Ebook Central, 2017. Available via World Wide Web. Access may be limited to ProQuest Ebook Central affiliated libraries.
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