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SIP : understanding the Session Initiation Protocol /

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 w...

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Detalles Bibliográficos
Clasificación:Libro Electrónico
Autor principal: Johnston, Alan B.
Formato: Electrónico eBook
Idioma:Inglés
Publicado: Boston : Artech House, ©2009.
Edición:3rd ed.
Colección:Artech House telecommunications library.
Temas:
Acceso en línea:Texto completo

MARC

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100 1 |a Johnston, Alan B. 
245 1 0 |a SIP :  |b understanding the Session Initiation Protocol /  |c Alan B. Johnston. 
246 3 0 |a Understanding the Session Initiation Protocol 
250 |a 3rd ed. 
260 |a Boston :  |b Artech House,  |c ©2009. 
300 |a 1 online resource (xxix, 395 pages) :  |b illustrations 
336 |a text  |b txt  |2 rdacontent 
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490 1 |a Artech House telecommunications series 
504 |a Includes bibliographical references and index. 
588 0 |a Print version record. 
520 3 |a 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.  |c Publisher abstract 
505 0 |a 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) 
505 8 |a 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 
505 8 |a 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 
505 8 |a 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 Notification; 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 
505 8 |a 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 Conflict; 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 
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650 0 |a Instant messaging  |x Computer programs. 
650 0 |a Internet telephony  |x Computer programs. 
650 0 |a Multimedia systems  |x Computer programs. 
650 0 |a Session Initiation Protocol (Computer network protocol) 
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