- •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
- •1.2.2 Data/Link Layer
- •1.2.3 Network Layer
- •1.2.4 Transport Layer
- •1.2 Internet Multimedia Protocol Stack
- •1.2.1 Physical 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
- •4.1.5 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
- •5.2.2 202 Accepted
- •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
- •5.4.3 402 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
- •5.4.11 410 Gone
- •5.4.12 411 Length Required
- •5.4.13 412 Conditional Request Failed
- •5.4 Client Error
- •5.4.1 400 Bad Request
- •5.4.2 401 Unauthorized
- •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
- •5.4.29 438 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
- •6.1.15 History Info
- •6.1.16 Organization
- •6.1.17 Path
- •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
- •6.1.27 User-Agent
- •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
- •6.3.7 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
- •7.4 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
- •8.4.6 Partial Publication
- •8.4.7 Presence Documents Summary
- •8.5 Instant Messaging with SIMPLE
- •8.5.1 Page Mode Instant Messaging
- •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
- •9.8 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
- •11.4.1 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
- •12.4.1 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
- •13.1.5 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
- •14.1.4 Message Authentication
- •14.2 Threats
- •14.3 Security Protocols
- •14.3.1 IPSec
- •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
- •14.6 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
- •17.2 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
- •17.9 Improved NAT Traversal
- •17.10 Security Deployment
- •17.11 Better Interoperability
- •References
- •A.1 ABNF Rules
- •A.2 Introduction to XML
- •References
- •About the Author
Network Address Translation |
261 |
NAT B is endpoint independent mapping NAT with endpoint dependent filtering.
NAT A is address and port dependent mapping with endpoint independent filtering.
Packet sent from A 192.168.0.1:1234 to a STUN server at
15.1.2.3 creates a mapping of 23.3.2.8:4219
Packet sent from B 10.0.1.13:5678 to a STUN server at
15.1.2.3 creates a mapping of 2.73.3.2:31212
Assume NAT A has only a single IP address. Assume NAT
A and B do not implement port preservation.
Show the filter rules created as A and B begin hole punching. Will hole punching succeed?
Q10.8 Repeat Question Q10.7 with everything the same except:
NAT B is address dependent mapping NAT with endpoint dependent filtering.
NAT A is address and port dependent mapping with endpoint dependent filtering.
Show the filter rules created as A and B begin hole punching.
Will hole punching succeed?
Q10.9 For a SIP message sent by host A (hostname of hosta. mappings.org) to host Y1 in Question Q10.4, show the Via header in the response assuming the rport extension is used.
Q10.10 Explain the relationship between hole punching and ICE.
References
[1]Egevang, K., and P. Francis, “The IP Network Address Translator (NAT),” RFC 1631, May 1994.
[2]Rekhter, Y., et al., “Address Allocation for Private Internets,” BCP 5, RFC 1918, February 1996.
[3]Srisuresh, P., and M. Holdrege, “IP Network Address Translator (NAT) Terminology and Considerations,” RFC 2663, August 1999.
[4]http://www.ietf.org/html.charters/behave-charter.html.
[5]Hain, T., “Architectural Implications of NAT,” RFC 2993, November 2000.
[6]Audet, F., and C. Jennings, “Network Address Translation (NAT) Behavioral Requirements for Unicast UDP,” BCP 127, RFC 4787, January 2007.
[7]Senie, D., “Network Address Translator (NAT)-Friendly Application Design Guidelines,” RFC 3235, January 2002.
262 |
SIP: Understanding the Session Initiation Protocol |
[8]Rosenberg, J., et al., “STUN—Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs),” RFC 3489, March 2003.
[9]Rosenberg, J., “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols,” draft-ietf-mmusic-ice-19 (work in progress), October 2007.
[10]MacDonald, D., and B. Lowekamp, “NAT Behavior Discovery Using STUN,” draft-ietf- behave-nat-behavior-discovery-05 (work in progress), November 2008.
[11]Daigle, L., and IAB, “IAB Considerations for Unilateral Self-Address Fixing (UNSAF) Across Network Address Translation,” RFC 3424, November 2002.
[12]Rosenberg, J., and H. Schulzrinne, “An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing,” RFC 3581, August 2003.
[13]Gurbani, V., R. Mahy, and B. Tate, “Connection Reuse in the Session Initiation Protocol (SIP),” draft-ietf-sip-connect-reuse-14 (work in progress), August 2009.
[14]Jennings, C., and R. Mahy, “Managing Client Initiated Connections in the Session Initiation Protocol (SIP),” draft-ietf-sip-outbound-16 (work in progress), October 2008.
[15]Huitema, C., “Real Time Control Protocol (RTCP) Attribute in Session Description Protocol (SDP),” RFC 3605, October 2003.
[16]Rosenberg, J., R. Mahy, and P. Matthews, “Traversal Using Relays Around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN),” RFC 5389, October 2008.
[17]Boulton, C., et al., “Best Current Practices for NAT Traversal for Client-Server SIP,” Internet Draft draft-ietf-sipping-nat-scenarios-09 (work in progress), September 2008.