- •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
|
|
SIP Header Fields |
171 |
|
|
Table 6.21 |
|
|
Common Content-Types Present in SIP Requests and Responses |
|
|
|
|
|
|
|
Content-Type |
Use |
|
|
|
|
|
|
application/sdp |
SDP in INVITE, ACK, or UPDATE requests [47] |
|
|
message/sipfrag |
SIP fragment in NOTIFY in refer subscription [48] |
|
|
application/xml+dialog |
XML dialog [24] |
|
|
application/xml+conf |
XML conference info [25] |
|
|
text/plain |
Plain text |
|
|
text/html |
HTML text |
|
|
application/isup |
Encapsulated ISUP in INVITE, BYE, or INFO [49] |
|
Content-Type: message/external-body; access-type=”URL”; URL=”http://www.example.com/”
The compact form is c. Examples are:
Content-Type: application/sdp
c:text/html
6.4.6MIME-Version
The MIME-Version header field is used to indicate the version of MIME protocol used to construct the message body. SIP, like HTTP, is not considered MIMEcompliant because parsing and semantics are defined by the SIP standard, not the MIME specification [51]. Version 1.0 is the default value. An example is:
MIME-Version: 1.0
6.5Questions
Q6.1 Show a 200 OK response to an OPTIONS that provides the maximum amount of information about the capabilities of the UA.
Q6.2 Generate a call flow that shows the use of the Reason header field.
Q6.3 Give three examples of header fields that a proxy might remove from a request before forwarding. Explain why each header field would be removed.
Q6.4 Give two examples of header fields that a proxy must modify when forwarding a request.
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Q6.5 Which three header fields can be present in a REFER request but not in other methods?
Q6.6 Which header fields would a topology hiding element likely need to modify or remove from a request?
Q6.7 A single header field is modified by a UAC after receiving a 410 response, after which the request receives a 200 response. Which header field was modified?
Q6.8 Which two header fields contain SIP entity tags? How are they typically used?
Q6.9 If a B2BUA between two UAs modified the Call-ID header field during a call setup, which header fields might fail to work properly?
Q6.10 Explain which features the Answer-Mode header field can be used to implement.
References
[1]Rosenberg, J., H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler, “SIP: Session Initiation Protocol,” RFC 3261, June 2002.
[2]Fielding, R., et al., “Hypertext Transfer Protocol—HTTP/1.1,” RFC 2616, June 1999.
[3]Postel, J., “Media Type Registration Procedure,” RFC 1590, 1994.
[4]Roach, A., “Session Initiation Protocol (SIP)-Specific Event Notification,” RFC 3265, 2002.
[5]Willis, D., and A. Allen, “Requesting Answering Modes for the Session Initiation Protocol (SIP),” RFC 5373, November 2008.
[6]Rosenberg, J., H. Schulzrinne, and P. Kyzivat, “Indicating User Agent Capabilities in the Session Initiation Protocol (SIP),” RFC 3840, August 2004.
[7]Johnston, A., and O. Levin, “Session Initiation Protocol (SIP) Call Control—Conferencing for User Agents,” RFC 4579, August 2006.
[8]Rosenberg, J., H. Schulzrinne, and O. Levin, “A Session Initiation Protocol (SIP) Event Package for Conference State,” RFC 4575, August 2006.
[9]Braden, R., “Requirements for Internet Hosts: Application and Support,” RFC 1123, 1989.
[10]Barnes, M., “An Extension to the Session Initiation Protocol (SIP) for Request History Information,” RFC 4244, November 2005.
[11]Willis, D., and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” RFC 3327, December 2002.
[12]Burger, E., H. Kaplan, and C. Holmberg, “Session Initiation Protocol (SIP) INFO Method and Package Framework,” draft-ietf-sip-info-events-03 (work in progress), January 2009.
SIP Header Fields |
173 |
[13]Levin, O., “Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription,” RFC 4488, May 2006.
[14]Elwell, J., “Connected Identity in the Session Initiation Protocol (SIP),” RFC 4916, June 2007.
[15]Rosenberg, J., “Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP),” draft-ietf-sip-gruu-15 (work in progress), October 2007.
[16]Rosenberg, J., “Indicating Support for Interactive Connectivity Establishment (ICE) in the Session Initiation Protocol (SIP),” draft-ietf-sip-ice-option-tag-02 (work in progress), June 2007.
[17]Mahy, R., and D. Petrie, “The Session Initiation Protocol (SIP) Join Header,” RFC 3911, October 2004.
[18]Camarillo, G., et al., “Referring to Multiple Resources in the Session Initiation Protocol (SIP),” RFC 5368, October 2008.
[19]Levin, O., “Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription,” RFC 4488, May 2006.
[20]Jennings, C., and R. Mahy, “Managing Client Initiated Connections in the Session Initiation Protocol (SIP),” draft-ietf-sip-outbound-20 (work in progress), June 2009.
[21]Willis, D., and B. Hoeneisen, “Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts,” RFC 3327, 2002.
[22]Camarillo, G., W. Marshall, and J. Rosenberg, “Integration of Resource Management and Session Initiation Protocol (SIP),” RFC 3312, October 2002.
[23]Rosenberg, J., H. Schulzrinne, and P. Zyzivat, “Caller Preferences for the Session Initiation Protocol (SIP),” RFC 3841, August 2004.
[24]Peterson, J., “A Privacy Mechanism for the Session Initiation Protocol,” RFC 3323, November 2002.
[25]Camarillo, G., and A. Johnston, “Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP),” RFC 5366, October 2008.
[26]Garcia-Martin, M., and G. Camarillo, “Multiple-Recipient MESSAGE Requests in the Session Initiation Protocol (SIP),” RFC 5365, October 2008.
[27]Camarillo, G., A. Roach, and O. Levin, “Subscriptions to Request-Contained Resource Lists in the Session Initiation Protocol (SIP),” RFC 5367, October 2008.
[28]Rosenberg, J., and H. Schulzrinne, “Reliability of Provisional Responses in Session Initiation Protocol (SIP),” RFC 3262, 2002.
[29]Mahy, R., B. Biggs, and R. Dean, “The Session Initiation Protocol (SIP) Replaces Header,” RFC 3891, September 2004.
[30]Schulzrinne, H., and J. Polk, “Communications Resource Priority for the Session Initiation Protocol (SIP),” RFC 4412, February 2006.
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[31]Camarillo, G., and J. Rosenberg, “Usage of the Session Description Protocol (SDP) Alternative Network Address Types (ANAT) Semantics in the Session Initiation Protocol (SIP),” RFC 4092, June 2005.
[32]Arkko, J., et al., “Security Mechanism Agreement for the Session Initiation Protocol (SIP),” RFC 3329, January 2003.
[33]Rosenberg, J., “Request Authorization Through Dialog Identification in the Session Initiation Protocol (SIP),” RFC 4538, June 2006.
[34]Donovan, S., and J. Rosenberg, “Session Timers in the Session Initiation Protocol (SIP),” RFC 4028, April 2005.
[35]Peterson, J., and C. Jennings, “Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP),” RFC 4474, August 2006.
[36]Johnston, A., et al., “Session Initiation Protocol Private Extension for an OSP Authorization Token,” IETF Internet-Draft, Work in Progress, February 2003.
[37]European Telecommunications Standards Institute, “Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); Open Settlement Protocol (OSP) for Inter-Domain Pricing, Authorization, and Usage Exchange,” Technical Specification 101 321, Version 2.1.0.
[38]Jennings, C., J. Peterson, and M. Watson, “Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity Within Trusted Networks,” RFC 3325, 2002.
[39]Sparks, R., et al., “Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies,” RFC 5393, December 2008.
[40]Schulzrinne, H., D. Oran, and G. Camarillo, “The Reason Header Field for the Session Initiation Protocol (SIP),” RFC 3326, 2002.
[41]Sparks, R., “The Session Initiation Protocol (SIP) Refer Method,” RFC 3515, 2003.
[42]Sparks, R., “The SIP Referred-By Mechanism,” RFC 3892, September 2004.
[43]Rosenberg, J., G. Camarillo, and D. Willis, “A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP),” RFC 5360, October 2008.
[44]Willis, D., and B. Hoeneisen, “Session Initiation Protocol Extension Header Field for Service Route Discovery During Registration,” RFC 3608, October 2003.
[45]Niemi, A., “Session Initiation Protocol (SIP) Extension for Event State Publication,” RFC 3903, October 2004.
[46]Niemi, A., “An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification,” draft-ietf-sipcore-subnot-etags-02 (work in progress), April 2009.
[47]Handley, M., V. Jacobson, and C. Perkins, “SDP: Session Description Protocol,” RFC 4566, July 2006.
[48]Sparks, R., “Internet Media Type message/sipfrag,” RFC 3420, November 2002.
[49]Zimmerer, E., et al., “MIME Media Types for ISUP and QSIG Objects,” RFC 3204, December 2001.
SIP Header Fields |
175 |
[50]Burger, E., “A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages,” RFC 4483, May 2006.
[51]Freed, M., and N. Borenstein, “Multipurpose Internet Mail Extensions (MIME). Part One: Format of Internet Message Bodies,” RFC 2045, 1996.