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Handbook on Session Initiation Protocol: Networked Multimedia Communications for IP Telephony [Kõva köide]

  • Formaat: Hardback, 860 pages, kõrgus x laius: 280x210 mm, kaal: 2300 g, 52 Tables, black and white; 199 Illustrations, black and white
  • Ilmumisaeg: 09-Mar-2016
  • Kirjastus: CRC Press Inc
  • ISBN-10: 1498747701
  • ISBN-13: 9781498747707
Teised raamatud teemal:
Handbook on Session Initiation Protocol: Networked Multimedia Communications for IP TelephonySuurem pilt
  • Formaat: Hardback, 860 pages, kõrgus x laius: 280x210 mm, kaal: 2300 g, 52 Tables, black and white; 199 Illustrations, black and white
  • Ilmumisaeg: 09-Mar-2016
  • Kirjastus: CRC Press Inc
  • ISBN-10: 1498747701
  • ISBN-13: 9781498747707
Teised raamatud teemal:
Püsilink: https://www.kriso.ee/db/9781498747707.html
Märksõnad:
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.
List of Figures xix
List of Tables xxvii
Preface xxix
Author xxxi
1 Networked Multimedia Services 1(4)
1.1 Introduction
1(1)
1.2 Functional Characteristics
1(1)
1.3 Performance Characteristics
1(2)
1.4 Summary
3(1)
References
3(2)
2 Basic Session Initiation Protocol 5(162)
2.1 Introduction
5(1)
2.2 Terminology
5(1)
2.3 Multimedia Session
5(14)
2.4 Session Initiation Protocol
19(23)
2.4.1 Augmented Backus—Naur Form for the SIP
20(16)
2.4.2 SIP Messages
36(2)
2.4.3 SIP Message Structure
38(1)
2.4.4 SIP Network Functional Elements
39(3)
2.5 SIP Request Messages
42(1)
2.6 SIP Response Messages
42(9)
2.7 SIP Call and Media Trapezoid Operation
51(11)
2.8 SIP Header Fields
62(10)
2.8.1 Overview
62(10)
2.8.2 Header-Field Descriptions
72(1)
2.9 SIP Tags
72(1)
2.10 SIP Option Tags
72(82)
2.11 SIP Media Feature Tags
154(10)
2.11.1 Contact Header Field
154(1)
2.11.2 Feature Tag Name, Description, and Usage
154(1)
2.11.3 Conveying Feature Tags with REFER
155(9)
2.12 Summary
164(1)
References
165(2)
3 SIP Message Elements 167(102)
3.1 Introduction
167(8)
3.1.1 SIP UA General Behavior
167(1)
3.1.2 UAC General Behavior
168(4)
3.1.3 UAS General Behavior
172(3)
3.1.4 Redirect Server General Behavior
175(1)
3.2 Canceling a Request
175(1)
3.3 Registration
176(16)
3.3.1 Registration without Managing Client-Initiated Connection
176(3)
3.3.2 Discovering a SIP Registrar
179(1)
3.3.3 Multiple-AOR Registration
179(1)
3.3.4 Registration Call Flows
179(2)
3.3.5 Registration for Multiple Phone Numbers in SIP
181(11)
3.4 Indicating UA Capabilities
192(4)
3.4.1 Contact Header Field
192(1)
3.4.2 Capability Expression Using Media Feature Tag
193(1)
3.4.3 Usage of the Content Negotiation Framework
193(3)
3.4.4 Indicating Feature Sets in Remote Target URIs
196(1)
3.4.5 OPTIONS Processing
196(1)
3.5 Discovering UA and Proxy Capabilities
196(1)
3.5.1 OPTIONS Request
197(1)
3.5.2 Response to OPTIONS Request
197(1)
3.6 Dialogs
197(9)
3.6.1 Creation of a Dialog
198(1)
3.6.2 Requests within a Dialog
199(2)
3.6.3 Termination of a Dialog
201(1)
3.6.4 Example of Dialog State
201(1)
3.6.5 Multiple Dialogs
202(1)
3.6.6 Early Dialog Termination Indication
202(4)
3.7 Initiating a Session
206(5)
3.7.1 Overview of Operation
206(2)
3.7.2 UAC Processing
208(2)
3.7.3 UAS Processing
210(1)
3.8 Modifying an Existing Session
211(22)
3.8.1 UAC Behavior
212(1)
3.8.2 UAS Behavior
212(1)
3.8.3 UPDATE
213(2)
3.8.4 SDP Offer and Answer
215(9)
3.8.5 Re-INVITE and Target Refresh Request Handling in SIP
224(9)
3.9 Handling Message Body
233(6)
3.9.1 Objective
233(1)
3.9.2 Message-Body Encoding
233(1)
3.9.3 Message Bodies: Multipart
234(1)
3.9.4 Message Bodies: Multipart/Mixed
235(1)
3.9.5 Message Bodies: Multipart/Alternative
235(1)
3.9.6 Message Bodies: Multipart/Related
235(1)
3.9.7 Disposition Types
236(2)
3.9.8 Message-Body Processing
238(1)
3.9.9 Future SIP Extensions
238(1)
3.10 Terminating a Session
239(1)
3.10.1 Terminating a Session with a BYE Request
239(1)
3.11 Proxy Behavior
240(13)
3.11.1 Overview
240(1)
3.11.2 Stateful Proxy
240(1)
3.11.3 Request Validation
241(1)
3.11.4 Route Information Preprocessing
242(1)
3.11.5 Determining Request Targets
242(1)
3.11.6 Request Forwarding
243(4)
3.11.7 Response Processing
247(3)
3.11.8 Processing Timer C
250(1)
3.11.9 Handling Transport Errors
250(1)
3.11.10 CANCEL Processing
250(1)
3.11.11 Stateless Proxy
250(1)
3.11.12 Summary of Proxy Route Processing
251(2)
3.12 Transactions
253(9)
3.12.1 Client Transaction
256(4)
3.12.2 Server Transaction
260(2)
3.13 Transport
262(3)
3.13.1 Clients
263(1)
3.13.2 Servers
264(1)
3.13.3 Framing
265(1)
3.13.4 Error Handling
265(1)
3.14 Summary
265(1)
References
266(3)
4 Addressing in SIP 269(34)
4.1 Introduction
269(1)
4.2 SIP Public Address
269(15)
4.2.1 SIP and SIPS Uniform Resource Indicators
270(4)
4.2.2 Telephone URI
274(2)
4.2.3 Use of SIPS URI Scheme in SIP
276(8)
4.3 Globally Routable UA URI
284(11)
4.3.1 Overview
284(1)
4.3.2 GRUU Grammar
284(1)
4.3.3 Operation
284(3)
4.3.4 Obtaining a GRUU
287(1)
4.3.5 Using a GRUU
287(1)
4.3.6 Dereferencing a GRUU
287(1)
4.3.7 UA Behavior
287(3)
4.3.8 Registrar Behavior
290(2)
4.3.9 Proxy Behavior
292(1)
4.3.10 GRUU Example
293(2)
4.4 Services URI
295(7)
4.4.1 Messaging Services
296(2)
4.4.2 Media Services
298(4)
4.5 Summary
302(1)
5 SIP Event Framework and Packages 303(6)
5.1 Introduction
303(1)
5.2 Event Framework
303(4)
5.2.1 Overview
303(1)
5.2.2 Subscription, Notification, and Publication Event Model
304(3)
5.3 Event Package
307(1)
5.4 Summary
307(2)
6 Presence and Instant Messaging in SIP 309(8)
6.1 Introduction
309(1)
6.2 SIP Presence
309(3)
6.2.1 Overview
309(1)
6.2.2 SIP Extensions for Presence
310(1)
6.2.3 Presence Data Formats and Processing
311(1)
6.2.4 Presence Operations
311(1)
6.3 SIP Instant Messaging
312(4)
6.3.1 Pager-Mode Single Recipient
312(1)
6.3.2 Pager-Mode Multiple Recipients
313(1)
6.3.3 Two-Party Session Mode
313(2)
6.3.4 Multiparty Session Mode
315(1)
6.4 Summary
316(1)
7 Media Transport Protocol and Media Negotiation 317(34)
7.1 Introduction
317(1)
7.2 Real-Time Transmission and Control Protocol
318(5)
7.2.1 Overview
318(1)
7.2.2 RTP Specification
318(3)
7.2.3 RTCP Specification
321(2)
7.3 Secure RTP (SRTP)
323(1)
7.4 ZRTP
324(1)
7.5 Real-Time Streaming Protocol (RTSP)
324(2)
7.6 Media Resource Control Protocol (MRCP)
326(1)
7.7 Session Description Protocol (SDP)
327(22)
7.7.1 Overview
327(1)
7.7.2 SDP Specification
327(1)
7.7.3 SDP Field Description
328(5)
7.7.4 SDP Media
333(1)
7.7.5 SDP Content-Agnostic Attributes
334(5)
7.7.6 SDP Transport-Independent Bandwidth Modifier
339(6)
7.7.7 SDP Format for BFCP Streams
345(2)
7.7.8 SDP Content Attribute
347(2)
7.8 Summary
349(1)
References
350(1)
8 DNS and ENUM in SIP 351(32)
8.1 Introduction
351(1)
8.2 Domain Name System
352(9)
8.2.1 Namespace
352(1)
8.2.2 Resource Records
353(1)
8.2.3 Name Servers
353(3)
8.2.4 Locating/Discovering SIP Entities
356(5)
8.3 ENUM
361(19)
8.3.1 DDDS Algorithm
366(1)
8.3.2 DDDS Algorithm Application to ENUM
366(4)
8.3.3 ENUM with Compound NAPTRs
370(1)
8.3.4 ENUM Operations
370(1)
8.3.5 ENUM Service Registration for SIP Addresses of Record (AORs)
371(1)
8.3.6 ENUM Services Registration in XML Chunk
372(1)
8.3.7 Using E.164 Numbers with SIP
373(4)
8.3.8 ENUM for SIP Services
377(2)
8.3.9 ENUM Implementation Issues
379(1)
8.4 DSN and ENUM Security
380(1)
8.4.1 Cache Poisoning
380(1)
8.4.2 Client Flooding
381(1)
8.4.3 Dynamic Updates Vulnerability
381(1)
8.4.4 Information Leakage
381(1)
8.4.5 Compromising Authoritative Data
381(1)
8.5 Summary
381(2)
9 Routing in SIP 383(32)
9.1 Introduction
383(1)
9.2 SIP Registrar
383(2)
9.3 SIP Proxy
385(1)
9.4 Traversing a Strict-Routing Proxy
386(1)
9.5 Rewriting Record-Route Header Field Values
387(1)
9.5.1 Problems and Recommendation
387(1)
9.6 Record-Routing with Globally Routable UA URI
387(1)
9.7 Double Route-Record
388(2)
9.8 Transport Parameter Usage Problems and Remedies
390(4)
9.8.1 UA Implementation
390(2)
9.8.2 Proxy Implementation
392(1)
9.8.3 Symmetric Response Routing
392(2)
9.9 Caller Preferences-Based Routing
394(8)
9.9.1 Overview
394(1)
9.9.2 Operation
395(1)
9.9.3 UAC Behavior
395(1)
9.9.4 UAS Behavior
396(1)
9.9.5 Proxy Behavior
397(3)
9.9.6 Mapping Feature Parameters to a Predicate
400(1)
9.9.7 Header Field Definitions
401(1)
9.9.8 Augmented BNF
402(1)
9.10 Location-Based Routing
402(5)
9.10.1 Overview
402(1)
9.10.2 Basic SIP Location Conveyance Operations
403(2)
9.10.3 Geolocation Examples
405(2)
9.11 Loop Detection
407(5)
9.11.1 Enhancements in Loop-Detection Algorithm
408(1)
9.11.2 Max-Breadth Header Field
409(3)
9.12 Summary
412(3)
10 User and Network-Asserted Identity in SIP 415(10)
10.1 Introduction
415(1)
10.2 Multiple User Identities
415(1)
10.3 Public User Identity
416(1)
10.4 Private User Identity
416(4)
10.4.1 P-Asserted-Identity
417(1)
10.4.2 P-Preferred-Identity
417(1)
10.4.3 Identity
417(1)
10.4.4 Recommended Use of Asserted Identity with SIP Messages
418(2)
10.5 Network-Asserted Identity
420(3)
10.5.1 Overview
420(1)
10.5.2 Trust Domain Identities, NAI, and Trust Domain Specification
421(1)
10.5.3 Generation of NAI
422(1)
10.5.4 Transport of NAI
422(1)
10.5.5 Parties with NAIs
422(1)
10.5.6 Types of NAI
422(1)
10.6 Summary
423(2)
11 Early Media in SIP 425(12)
11.1 Introduction
425(1)
11.2 Early Media and Session Establishment in SIP
425(1)
11.3 Early-Media Solution Models
426(1)
11.4 Early-Media Solution Model with Disposition-Type: Early-Session
426(6)
11.4.1 Overview
426(1)
11.4.2 Issues Related to Early-Media Session Establishment
427(1)
11.4.3 Early-Session Disposition Type
427(1)
11.4.4 Preconditions
428(1)
11.4.5 Option Tag
428(1)
11.4.6 Example
428(1)
11.4.7 Early-Media Solution with Application Server and Gateway Model
428(4)
11.5 Early-Media Solution Model with P-Early-Media Header
432(4)
11.5.1 Early-Media Policy
432(1)
11.5.2 Early-Media Application Environments
432(1)
11.5.3 Early-Media Authorization
432(1)
11.5.4 Applicability of Content-Disposition and Application/Gateway Model
433(1)
11.5.5 Operation
434(1)
11.5.6 Limitations of the P-Early-Media Header Field
434(1)
11.5.7 P-Early-Media Header Field
435(1)
11.6 Summary
436(1)
12 Service and Served-User Identity in SIP 437(12)
12.1 Introduction
437(1)
12.2 Communications Service ID
438(1)
12.2.1 Derived Service ID
438(1)
12.2.2 SIP's Expressiveness for Negotiation
438(1)
12.2.3 Presence
438(1)
12.2.4 Intradomain
438(1)
12.2.5 Device Dispatch
439(1)
12.3 Asserted- and Preferred-Service ID
439(5)
12.3.1 Overview
439(1)
12.3.2 Applicability Statement
440(1)
12.3.3 Header Fields
440(2)
12.3.4 Usage of Header Fields in Requests
442(1)
12.3.5 Usage of Header Fields in Responses
442(1)
12.3.6 Examples of Usage
442(2)
12.4 Served-User ID for Handling Services
444(4)
12.4.1 P-Served-User Header
444(1)
12.4.2 Application Service Invocation
445(2)
12.4.3 P-Served-User Header Field Usage, Definition, and Syntax
447(1)
12.4.4 Proxy Behavior: Generating the P-Served-User Header
447(1)
12.4.5 Proxy Behavior: Consuming the P-Served-User Header
447(1)
12.4.6 Applicability and Limitations
448(1)
12.5 Summary
448(1)
13 Connections Management and Overload Control in SIP 449(40)
13.1 Introduction
449(1)
13.2 Connections Management in SIP Network
449(23)
13.2.1 Overview
449(1)
13.2.2 Flow-Based Connections Setup
450(1)
13.2.3 Keep-Alive Mechanisms
450(1)
13.2.4 Grammar
451(1)
13.2.5 Connections Management Procedures for SIP Entities
451(9)
13.2.6 Keep-Alive Mechanisms in SIP Network
460(3)
13.2.7 Connection Management Example
463(4)
13.2.8 Connection Reuse in SIP
467(5)
13.3 Loss-Based Overload Control in SIP Network
472(11)
13.3.1 Overview
472(1)
13.3.2 Operations
472(1)
13.3.3 Via Header Parameters for Overload Control
473(1)
13.3.4 General Behavior
474(4)
13.3.5 Loss-Based Overload Control Scheme
478(2)
13.3.6 Relationship with Other SIP Load Control Schemes
480(1)
13.3.7 Syntax
480(1)
13.3.8 Design Considerations for Overload Control
480(2)
13.3.9 Salient Features of Overload Control
482(1)
13.4 Rate-Based Overload Control in SIP Network
483(4)
13.4.1 Overview
483(1)
13.4.2 Rate-Based Algorithm Scheme
483(4)
13.4.3 Example
487(1)
13.4.4 Syntax
487(1)
13.5 Summary
487(1)
References
488(1)
14 Interworking Services in SIP 489(16)
14.1 Introduction
489(1)
14.2 SIP Session Border Controller
489(10)
14.2.1 Objective
489(1)
14.2.2 Background on SBCs
490(2)
14.2.3 Functions of SBCs
492(7)
14.2.4 Derived Requirements for Future SIP Standardization Work
499(1)
14.3 NAT Crossing by SIP
499(2)
14.3.1 Overview
499(1)
14.3.2 NAT-Crossing Protocols
499(2)
14.4 SIP—PSTN/ISDN Protocols Interworking
501(2)
14.4.1 Overview
501(1)
14.4.2 SIP-PSTN/ISDN Protocols Interworking Framework
501(2)
14.5 Summary
503(1)
Reference
503(2)
15 Resource Priority and Quality of Service in SIP 505(38)
15.1 Introduction
505(1)
15.2 Communications Resource Priority in SIP
506(10)
15.2.1 Overview
506(1)
15.2.2 Resource-Priority SIP Header Field
507(1)
15.2.3 Behavior of SIP Elements That Receive Prioritized Requests
508(2)
15.2.4 UAC Behavior
510(1)
15.2.5 UAS Behavior
511(1)
15.2.6 Proxy Behavior
511(1)
15.2.7 Third-Party Authentication
511(1)
15.2.8 Backwards Compatibility
511(1)
15.2.9 Examples
512(1)
15.2.10 Handling Multiple Concurrent Namespaces
513(1)
15.2.11 Registering Namespaces
514(1)
15.2.12 Namespace Definitions
515(1)
15.3 Preemption Events in SIP
516(8)
15.3.1 Overview
516(1)
15.3.2 Access Preemption Events
517(1)
15.3.3 Network Preemption Events
518(2)
15.3.4 Hybrid Infrastructure Preemption Events
520(1)
15.3.5 Preemption Reason Header Cause Codes and Semantics
520(4)
15.4 QOS in SIP
524(14)
15.4.1 Overview
524(1)
15.4.2 SDP Parameters
525(1)
15.4.3 Usage of Preconditions with Offer—Answer
525(2)
15.4.4 Suspending and Resuming Session Establishment
527(1)
15.4.5 Status Confirmation
528(1)
15.4.6 Refusing an Offer
528(1)
15.4.7 Unknown Precondition Type
529(1)
15.4.8 Multiple Preconditions per Media Stream
529(1)
15.4.9 Option Tag for Preconditions
529(1)
15.4.10 Indicating Capabilities
529(1)
15.4.11 Examples
529(3)
15.4.12 Enhancements in Precondition Procedures and Use in Session Mobility
532(3)
15.4.13 SIP Performance Metrics
535(3)
15.5 SDP Media Streams Mapping to QOS Flows
538(1)
15.5.1 Overview
538(1)
15.5.2 SRF Semantics
538(1)
15.5.3 Applicability Statement
538(1)
15.5.4 Examples
538(1)
15.5.5 IANA Registration: SDP Attribute for Group
538(1)
15.6 QOS Mechanism Selection in SDP
539(1)
15.6.1 Overview
539(1)
15.6.2 SDP Attribute Definitions
539(1)
15.6.3 Offer—Answer Behavior
539(1)
15.6.4 Example
540(1)
15.6.5 IANA Registration: SDP Attribute and Token for QOS
540(1)
15.7 SIP Signaling Compression
540(1)
15.8 Summary
541(1)
References
542(1)
16 Call Services in SIP 543(86)
16.1 Introduction
543(1)
16.2 Call Transfer and Related Call Services
544(31)
16.2.1 Overview
544(1)
16.2.2 Actors and Roles
544(1)
16.2.3 Requirements
544(1)
16.2.4 Using REFER to Achieve Call Transfer
544(1)
16.2.5 Basic Transfer
545(4)
16.2.6 Transfer with Consultation Hold
549(13)
16.2.7 Transfer with Referred-By
562(2)
16.2.8 Transfer as an Ad Hoc Conference
564(1)
16.2.9 Transfer with Multiple Parties
564(1)
16.2.10 Gateway Transfer Issues
565(2)
16.2.11 Call Services with Shared Appearances of a SIP AOR
567(3)
16.2.12 Completion of Call Services in SIP
570(5)
16.3 Call Diversion Indication
575(7)
16.3.1 Overview
575(1)
16.3.2 Diversion and History-Info Header Interworking in SIP
576(6)
16.4 Call Services Using Session Border Controller
582(6)
16.4.1 Overview
582(1)
16.4.2 Distributed SBC Architecture
583(4)
16.4.3 Conclusion
587(1)
16.5 Referring Call to Multiple Resources
588(3)
16.5.1 Overview
588(1)
16.5.2 Operation
588(1)
16.5.3 Multiple-Refer SIP Option Tag
588(1)
16.5.4 Suppressing REFER's Implicit Subscription
588(1)
16.5.5 URI-List Format
589(1)
16.5.6 Behavior of SIP REFER-Issuers
590(1)
16.5.7 Behavior of REFER-Recipients
590(1)
16.5.8 Example
590(1)
16.6 Call Services with Content Indirection
591(5)
16.6.1 Overview
591(1)
16.6.2 Use-Case Examples
592(1)
16.6.3 Requirements
593(1)
16.6.4 Application of MIME-URI Standard to Content Indirection
593(3)
16.6.5 Examples
596(1)
16.7 Transcoding Call Services
596(11)
16.7.1 Transcoding Services Framework
596(1)
16.7.2 Third-Party Transcoding Services
597(7)
16.7.3 Conference Bridging Transcoding Call Control Flows
604(3)
16.8 INFO Method—Mid-Call Information Transfer
607(9)
16.8.1 Overview
607(1)
16.8.2 Motivation
607(1)
16.8.3 UAs Are Allowed to Enable Both Legacy INFO Usages and Info
608(1)
16.8.4 INFO Method
608(1)
16.8.5 INFO Packages
609(2)
16.8.6 Formal INFO Method Definition and Header Fields
611(1)
16.8.7 INFO Package Considerations
611(1)
16.8.8 Alternative Mechanisms
611(1)
16.8.9 INFO Package Requirements
612(2)
16.8.10 Examples
614(2)
16.9 SIP Call Control UUI Transfer Services
616(7)
16.9.1 Overview
616(1)
16.9.2 Requirements for UUI Transport
616(1)
16.9.3 Possible Approaches for UUI Transport in SIP
617(2)
16.9.4 SIP Extensions for UUI Transport
619(1)
16.9.5 Normative Definition
619(2)
16.9.6 Guidelines for UUI Packages
621(1)
16.9.7 Use Cases
622(1)
16.10 Call Services Using DTMF
623(1)
16.11 Emergency Call Services in SIP
624(2)
16.11.1 Overview
624(1)
16.11.2 Emergency Services Uniform Resource Name
625(1)
16.11.3 Multilevel Precedence and Preemption
625(1)
16.12 Summary
626(1)
References
627(2)
17 Media Server Interfaces in SIP 629(16)
17.1 Introduction
629(1)
17.2 SIP Interface to VoiceXML Media Server
630(12)
17.2.1 Overview
630(1)
17.2.2 Use Cases
630(2)
17.2.3 VoiceXML Session Establishment and Termination
632(5)
17.2.4 Media Support
637(2)
17.2.5 Returning Data to the Application Server
639(1)
17.2.6 Outbound Calling
640(1)
17.2.7 Call Transfer
640(2)
17.3 Summary
642(1)
References
643(2)
18 Multiparty Conferencing in SIP 645(12)
18.1 Introduction
645(1)
18.2 Multiparty Multimedia Conferencing
645(1)
18.3 Third-Party Multiparty Conferencing
646(8)
18.3.1 3PCC Call Establishment
646(3)
18.3.2 Recommendations for 3PCC Call Setups
649(1)
18.3.3 Multiparty Call Establishment Error Handling
649(1)
18.3.4 Continued Call Processing in 3PCC
650(1)
18.3.5 3PCC and Early Media
650(1)
18.3.6 3PCC and SDP Preconditions
651(1)
18.3.7 3PCC Service Examples
652(2)
18.3.8 3PCC Implementation Recommendations
654(1)
18.3.9 Concluding Remarks
654(1)
18.4 Summary
654(3)
19 Security Mechanisms in SIP 657(122)
19.1 Introduction
657(1)
19.2 Multilevel Security Characteristics in SIP
658(21)
19.2.1 Overview
658(1)
19.2.2 Session-Level Security
658(14)
19.2.3 Media-Level Security
672(7)
19.3 Security Mechanisms Negotiation
679(7)
19.3.1 Security Mechanisms Negotiation
680(3)
19.3.2 Backwards Compatibility
683(1)
19.3.3 Security Algorithms Negotiation Example
683(1)
19.3.4 Security Considerations
684(1)
19.3.5 Syntax of IPsec-3GPP Security Headers
685(1)
19.4 Authentication in SIP
686(39)
19.4.1 Background
686(1)
19.4.2 Framework
687(1)
19.4.3 User-to-User Authentication
688(1)
19.4.4 Proxy-to-User Authentication
689(1)
19.4.5 Digest Authentication Scheme
690(1)
19.4.6 Domain Certificates over TLS for Authentication in SIP
690(7)
19.4.7 Authenticated Identity Body Format in SIP
697(4)
19.4.8 Cryptographic Authentication Scheme
701(15)
19.4.9 HTTP Digest Authentication Using AKA in SIP
716(5)
19.4.10 Key-Derivation Authentication Scheme in SIP
721(2)
19.4.11 DNS-Based Authentication for TLS Sessions in SIP
723(2)
19.5 Authorization in SIP
725(18)
19.5.1 Trait-Based Authorization in SIP
725(7)
19.5.2 Authorization through Dialog Identification in SIP
732(5)
19.5.3 Media Authorization in SIP
737(5)
19.5.4 Early-Media Authorization in SIP
742(1)
19.5.5 Framework for Session Setup with Media Authorization
742(1)
19.6 Integrity and Confidentiality in SIP
743(5)
19.6.1 S/MIME Certificates
743(1)
19.6.2 S/MIME Key Exchange
743(1)
19.6.3 Securing MIME Bodies
744(1)
19.6.4 SIP Header Confidentiality and Integrity Using S/MIME: Tunneling SIP
745(3)
19.7 Security for SIP URI-List Services
748(2)
19.7.1 Objective
748(1)
19.7.2 Requirements
748(1)
19.7.3 Framework
748(1)
19.7.4 Security Considerations
749(1)
19.8 Consent-Based Communications for Enhancing Security in SIP
750(12)
19.8.1 Objective
750(1)
19.8.2 Definitions and Terminology
751(1)
19.8.3 Relays and Translations
751(1)
19.8.4 Architecture
752(2)
19.8.5 Framework Operations
754(7)
19.8.6 Security Considerations
761(1)
19.9 SIP Forking Proxy Security
762(2)
19.9.1 Overview
762(1)
19.9.2 Vulnerability: Leveraging Forking to Flood a Network
762(2)
19.9.3 Security Considerations
764(1)
19.10 Nonrepudiation Services in SIP
764(1)
19.11 Call Flows Explaining SIP Security Features
765(2)
19.11.1 Registration
765(1)
19.11.2 Session Setup
765(2)
19.12 Threat Model and Security Usage Recommendations in SIP
767(8)
19.12.1 Attacks and Threat Models
768(1)
19.12.2 Security Mechanisms
769(2)
19.12.3 Implementing Security Mechanisms
771(3)
19.12.4 Limitations
774(1)
19.13 Summary
775(2)
References
777(2)
20 Privacy and Anonymity in SIP 779(36)
20.1 Introduction
779(1)
20.2 Privacy Mechanism in SIP
780(8)
20.2.1 Background
780(1)
20.2.2 Varieties of Privacy
781(1)
20.2.3 UA Behavior
782(1)
20.2.4 UA Behavior Constructing Private Messages
782(1)
20.2.5 UA Behavior Expressing Privacy Preferences
783(1)
20.2.6 UA Behavior Routing Requests to Privacy Services
784(1)
20.2.7 UA Behavior Routing Responses to Privacy Services
784(1)
20.2.8 Privacy Service Behavior
785(2)
20.2.9 Location Information Privacy
787(1)
20.2.10 Security Considerations
787(1)
20.3 Asserted and Preferred Identity for Privacy in SIP
788(5)
20.3.1 Background
788(1)
20.3.2 P-Asserted-Identity and P-Preferred-Identity for Privacy
788(1)
20.3.3 Proxy Behavior
789(1)
20.3.4 Hints for Multiple Identities
789(1)
20.3.5 Requesting Privacy
789(1)
20.3.6 UAS Behavior
790(1)
20.3.7 Examples
790(2)
20.3.8 Example of Spec(T)
792(1)
20.3.9 Security Considerations
792(1)
20.4 Connected Identity for Privacy in SIP
793(8)
20.4.1 Overview
793(1)
20.4.2 Terminology
793(1)
20.4.3 Overview of Solution
793(1)
20.4.4 UA Behavior outside the Context of an Existing Dialog
794(1)
20.4.5 Behavior of a UA Whose Identity Changes
794(1)
20.4.6 General UA Behavior
795(1)
20.4.7 Authentication Service Behavior
795(1)
20.4.8 Verifier Behavior
795(1)
20.4.9 Proxy Behavior
795(1)
20.4.10 Examples
796(4)
20.4.11 Security Considerations
800(1)
20.5 Guidelines for Using Privacy Mechanism in SIP
801(9)
20.5.1 Definition
801(1)
20.5.2 Semantics of Existing Priv-Values
801(1)
20.5.3 Target for Each Priv-Value
801(2)
20.5.4 Recommended Treatment of User Privacy-Sensitive Information
803(7)
20.6 Anonymity in SIP
810(4)
20.6.1 Overview
810(1)
20.6.2 UA-Driven Anonymity
810(3)
20.6.3 Rejecting Anonymous Requests
813(1)
20.7 Summary
814(1)
Appendix A: ABNF 815(6)
Appendix B: Reference RFCs 821(18)
Index 839
Radhika Ranjan Roy is an electronics engineer, US Army Research, Development, and Engineering Command (RDECOM), CommunicationsElectronics 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.