SIP

Name: SIP | Understanding the Session Initiation Protocol, Third Edition
Brand: Artech House
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Understanding the Session Initiation Protocol, Third Edition

Alan B. Johnston(Author)

Artech House (Publisher)

1st Edition

Published on 1. January 2009

424 pages

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978-1-60783-996-5 (ISBN)

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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
11.4.2 Example of H.323
11.4.3 Versions
References
12 Media Transport
12.1 Real-Time Transport Protocol (RTP)
12.2 RTP Control Protocol (RTCP)
12.2.1 RTCP Reports
12.2.2 RTCP Extended Reports
12.3 Compression
12.4 RTP Audio Video Profi les
12.4.1 Audio Codecs
12.4.2 Video Codecs
12.5 Conferencing
12.6 ToIP-Conversational Text
12.7 DTMF Transport
12.8 Questions
References
13 Negotiating Media Sessions
13.1 Session Description Protocol (SDP)
13.1.1 Protocol Version
13.1.2 Origin
13.1.3 Session Name and Information
13.1.4 URI
13.1.5 E-Mail Address and Phone Number
13.1.6 Connection Data
13.1.7 Bandwidth
13.1.8 Time, Repeat Times, and Time Zones
13.1.9 Encryption Keys
13.1.10 Media Announcements
13.1.11 Attributes
13.2 SDP Extensions
13.3 The Offer Answer Model
13.3.1 Rules for Generating an Offer
13.3.2 Rules for Generating an Answer
13.3.3 Rules for Modifying a Session
13.3.4 Special Case-Call Hold
13.4 Static and Dynamic Payloads
13.5 SIP Offer Answer Exchanges
13.6 Conclusion
13.7 Questions
References
14 SIP Security
14.1 Basic Security Concepts
14.1.1 Encryption
14.1.2 Public Key Cryptography
14.1.3 Diffi e-Hellman Cryptography
14.1.4 Message Authentication
14.1.5 Digital Certifi cates
14.2 Threats
14.3 Security Protocols
14.3.1 IPSec
14.3.2 TLS
14.3.3 DNSSec
14.3.4 Secure MIME
14.4 SIP Security Model
14.4.1 SIP Digest Authentication
14.4.2 SIP Authentication Using TLS
14.4.3 Secure SIP
14.4.4 Identity
14.4.5 Enhanced SIP Identity
14.5 SIP Certifi cate Service
14.6 Media Security
14.6.1 Non-RTP Media
14.6.2 Secure RTP
14.6.3 Keying SRTP
14.6.4 Best Effort Encryption
14.6.5 ZRTP
14.7 Questions
References
15 Peer-to-Peer SIP
15.1 P2P Properties
15.2 P2P Properties of SIP
15.3 P2P Overlays
15.4 RELOAD
15.5 Host Identity Protocol
15.6 Conclusion
15.7 Questions
References
16 Call Flow Examples
16.1 SIP Call with Authentication, Proxies, and Record-Route
16.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy
16.3 SIP to PSTN Call Through Gateways
16.4 PSTN to SIP Call Through a Gateway
16.5 Parallel Search
16.6 Call Setup with Two Proxies
16.7 SIP Presence and Instant Message Example
References
17 Future Directions
17.1 Bug Fixes and Clarifi cations
17.2 More Extensions
17.3 Better Identity
17.4 Interdomain SIP
17.5 Making Features Work Better
17.6 Emergency Calling
17.7 More SIP Trunking
17.8 P2P and HIP
17.9 Improved NAT Traversal
17.10 Security Deployment
17.11 Better Interoperability
References
Appendix Introduction to ABNF and XML
A.1 ABNF Rules
A.2 Introduction to XML
References
About the Author
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