Handbook on Session Initiation Protocol
Networked Multimedia Communications for IP Telephony
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|Format: ||Hardback, 908 pages|
|Other Information: ||199 black & white illustrations, 52 black & white tables|
|Published In: ||United States, 09 March 2016|
Session Initiation Protocol (SIP), standardized by the Internet Engineering Task Force (IETF), has emulated the simplicity of the protocol architecture of hypertext transfer protocol (HTTP) and is being popularized for VoIP over the Internet because of the ease with which it can be meshed with web services. However, it is difficult to know exactly how many requests for comments (RFCs) have been published over the last two decades in regards to SIP or how those RFCs are interrelated. Handbook on Session Initiation Protocol: Networked Multimedia Communications for IP Telephony solves that problem. It is the first book to put together all SIP-related RFCs, with their mandatory and optional texts, in a chronological and systematic way so that it can be used as a single super-SIP RFC with an almost one-to-one integrity from beginning to end, allowing you to see the big picture of SIP for the basic SIP functionalities. It is a book that network designers, software developers, product manufacturers, implementers, interoperability testers, professionals, professors, and researchers will find to be very useful. The text of each RFC from the IETF has been reviewed by all members of a given working group made up of world-renowned experts, and a rough consensus made on which parts of the drafts need to be mandatory and optional, including whether an RFC needs to be Standards Track, Informational, or Experimental. Texts, ABNF syntaxes, figures, tables, and references are included in their original form. All RFCs, along with their authors, are provided as references. The book is organized into twenty chapters based on the major functionalities, features, and capabilities of SIP.
Table of Contents
Networked Multimedia Services Introduction Functional Characteristics Performance Characteristics Summary Problems Basic Session Initiation Protocol Introduction Terminology Multimedia Session Session Initiation Protocol SIP Request Messages SIP Response Messages SIP Call and Media Trapezoid Operation SIP Header Fields SIP Tags SIP Option Tags SIP Media Feature Tags Summary Problems SIP Message Elements Introduction Canceling a Request Registration Indicating UA Capabilities Discovering UA and Proxy Capabilities Dialogs Initiating a Session Modifying an Existing Session Handling Message Body Terminating a Session Proxy Behavior Transactions Transport Summary Problems Addressing in SIP Introduction SIP Public Address Globally Routable UA URI Services URI Summary SIP Event Framework and Packages Introduction Event Framework Event Package Summary Problems Presence and Instant Messaging in SIP Introduction SIP Presence SIP Instant Messaging Summary Problems Media Transport Protocol and Media Negotiation Introduction Real-Time Transmission and Control Protocol Secure RTP (SRTP) ZRTP Real-Time Streaming Protocol (RTSP) Media Resource Control Protocol (MRCP) Session Description Protocol (SDP) Summary Problems DNS and ENUM in SIP Introduction Domain Name System ENUM DSN and ENUM Security Summary Problems Routing in SIP Introduction SIP Registrar SIP Proxy Traversing a Strict-Routing Proxy Rewriting Record-Route Header Field Values Record-Routing with Globally Routable UA URI Double Route-Record Transport Parameter Usage Problems and Remedies Caller Preferences-Based Routing Location-Based Routing Loop Detection Summary Problems User and Network-Asserted Identity in SIP Introduction Multiple User Identities Public User Identity Private User Identity Network-Asserted Identity Summary Problems Early Media in SIP Introduction Early Media and Session Establishment in SIP Early-Media Solution Models Early-Media Solution Model with Disposition-Type: Early-Session Early-Media Solution Model with P-Early-Media Header Summary Service and Served-User Identity in SIP Introduction Communications Service ID Asserted- and Preferred-Service ID Served-User ID for Handling Services Summary Problems Connections Management and Overload Control in SIP Introduction Connections Management in SIP Network Loss-Based Overload Control in SIP Network Rate-Based Overload Control in SIP Network Summary Problems Interworking Services in SIP Introduction SIP Session Border Controller NAT Crossing by SIP SIP-PSTN/ISDN Protocols Interworking Summary Problems Resource Priority and Quality of Service in SIP Introduction Communications Resource Priority in SIP Preemption Events in SIP QOS in SIP SDP Media Streams Mapping to QOS Flows QOS Mechanism Selection in SDP SIP Signaling Compression Summary Problems Call Services in SIP Introduction Call Transfer and Related Call Services Call Diversion Indication Call Services Using Session Border Controller Referring Call to Multiple Resources Call Services with Content Indirection Transcoding Call Services INFO Method-Mid-Call Information Transfer SIP Call Control UUI Transfer Services Call Services Using DTMF Emergency Call Services in SIP Media Server Interfaces in SIP Introduction SIP Interface to VoiceXML Media Server Summary Problems Multiparty Conferencing in SIP Introduction Multiparty Multimedia Conferencing Third-Party Multiparty Conferencing Summary Problems Security Mechanisms in SIP Introduction Multilevel Security Characteristics in SIP Security Mechanisms Negotiation Authentication in SIP Authorization in SIP Integrity and Confidentiality in SIP Security for SIP URI-List Services Consent-Based Communications for Enhancing Security in SIP SIP Forking Proxy Security Nonrepudiation Services in SIP Call Flows Explaining SIP Security Features Threat Model and Security Usage Recommendations in SIP Summary Problems Privacy and Anonymity in SIP Introduction Privacy Mechanism in SIP Asserted and Preferred Identity for Privacy in SIP Connected Identity for Privacy in SIP Guidelines for Using Privacy Mechanism in SIP Anonymity in SIP Summary Problems Appendix A: ABNF Appendix B: Reference RFCs
About the Author
Radhika Ranjan Roy is an electronics engineer, US Army Research, Development, and Engineering Command (RDECOM), Communications-Electronics Research, Development, and Engineering Center (CERDEC), Space and Terrestrial Communications Directorate (S&TCD) Laboratories, Aberdeen Proving Ground (APG), Maryland, since 2009. Before joining to US Army Research, he worked in various capacities in CACI, SAIC, AT&T/Bell Laboratories, CSC, and PDB since his graduation. He earned his PhD in electrical engineering with major in computer communications from the City University of New York, New York, in 1984, and MS in electrical engineering from the Northeastern University, Boston, Massachusetts, in 1978. He received his BS in electrical engineering from the Bangladesh University of Engineering and Technology, Dhaka, Bangladesh, in 1967. He has published more than 50 technical papers. He is holding and/or submitted over 30 patents. He authored a book titled Handbook of Mobile Ad Hoc Networks on Mobility Models in 2010.
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