- •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
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8.6.2Interworking with SIMPLE
Currently work is underway in the IETF to standardize the interworking between SIMPLE and Jabber for both presence and IM systems [42]. A gateway must be used to bridge the two protocols. Since XMPP utilizes permanent subscriptions, the gateway must map them to the soft-state subscriptions used in SIMPLE. While an XMPP URI scheme has been defined [43] that SIMPLE can use, XMPP is unable to use URIs so the gateway must map between XMPP or SIP URIs and XMPP JIDs. The basic IM mapping is shown in Table 8.11.
Table 8.12 shows basic presence mapping between XMPP and SIMPLE. The mapping of group chat between SIMPLE and XMPP is defined in [44].
8.6.3Jingle
Jingle [45] is an XMPP protocol extension for initiating and managing media sessions between two XMPP entities. Jingle provides a way to perform an offer/ answer exchange of media capabilities to negotiate an RTP media session. Jingle was designed to easily interwork with SIP [46] and have NAT traversal using ICE, as described in Chapter 10.
8.6.4Future Standardization of XMPP
A new working group has formed in the IETF to standardize additional aspects of XMPP [47] and interworking with SIMPLE. Their work will include revisions to the core protocol, optimizations for mobile devices, and end-to-end encryption and security.
8.7 Conclusion
This chapter showed how presence and instant messaging services can be implemented with SIP using a number of extensions. Additionally, Jabber or XMPP can provide these services and can interwork with SIP through gateways.
Table 8.11
Instant Message Mapping Between XMPP and SIMPLE
XMPP |
SIMPLE |
|
|
<body/> |
Body of MESSAGE |
<subject/> |
Subject |
<thread/> |
Call-ID |
from |
From |
to |
To |
xml:lang |
Content-Language |
|
|
|
Presence and Instant Messaging |
215 |
|
|
Table 8.12 |
|
|
|
Presence Mapping Between XMPP and SIMPLE |
|
|
|
XMPP |
SIMPLE |
|
|
|
|
|
|
<presence/> stanza |
Event: presence |
|
|
XMPP resource identifier |
Tuple ‘id’ attribute |
|
|
from |
From |
|
|
id |
Call-ID |
|
|
to |
To |
|
|
type |
Basic status |
|
|
xml:lang |
Content-Language |
|
|
<priority/> |
PIDF priority for tuple |
|
8.8Questions
Q8.1 Create an example resource list creating subscription, and show the first notification. Include at least three resources in the list and show at least two different presence states in the notification.
Q8.2 Explain the differences between page mode and session mode instant messaging. Which protocols are typically used for each?
Q8.3 Explain the purpose of CPIM.
Q8.4 Explain why you would not expect to see the SIP header field as a header field in a
Q8.5 Map the following Jabber message into a SIP NOTIFY. Make up any URIs and values you might need:
<presence
from=’juliet@example.com/balcony’
to=’romeo@example.net/orchard’
xml:lang=’en’>
<show>away</show>
<status>be right back</status> <priority>0</priority>
</presence>
Q8.6 Create a presence document for the user Bob showing his status as available, his contact URI as sip:bob@pc33.example.com, and his device as capable of both audio and video.
Q8.7 How are entity tags used in SIP?
Q8.8 Explain how partial publication works with an example.
Q8.9 What is a resource list server?
Q8.10 Explain conditional event notification.
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References
[1]http://www.icq.com.
[2]http://www.aim.com.
[3]Rosenberg, J., “SIMPLE made Simple: An Overview of the IETF Specifications for Instant Messaging and Presence Using the Session Initiation Protocol (SIP),” draft-ietf-simple-sim- ple-05 (work in progress), March 2009.
[4]Roach, A., “Session Initiation Protocol (SIP)-Specific Event Notification,” RFC 3265, June 2002.
[5]Niemi, A., “Session Initiation Protocol (SIP) Extension for Event State Publication,” RFC 3903, October 2004.
[6]Sugano, H., et al., “Presence Information Data Format (PIDF),” RFC 3863, August 2004.
[7]Rosenberg, J., “A Presence Event Package for the Session Initiation Protocol (SIP),” RFC 3856, August 2004.
[8]Rosenberg, J., “A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP),” RFC 3857, August 2004.
[9]Rosenberg, J., “An Extensible Markup Language (XML) Based Format for Watcher Information,” RFC 3858, August 2004.
[10]Schulzrinne, H., et al., “RPID: Rich Presence Extensions to the Presence Information Data Format (PIDF),” RFC 4480, July 2006.
[11]Schulzrinne, H., “CIPID: Contact Information for the Presence Information Data Format,” RFC 4482, July 2006.
[12]Rosenberg, J., “A Data Model for Presence,” RFC 4479, July 2006.
[13]Roach, A., B. Campbell, and J. Rosenberg, “A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists,” RFC 4662, August 2006.
[14]Khartabil, H., et al., “An Extensible Markup Language (XML)-Based Format for Event Notification Filtering,” RFC 4661, September 2006.
[15]Rosenberg, J., “The Extensible Markup Language (XML) Configuration Access Protocol (XCAP),” RFC 4825, May 2007.
[16]Rosenberg, J., “Extensible Markup Language (XML) Formats for Representing Resource Lists,” RFC 4826, May 2007.
[17]Isomaki, M., and E. Leppanen, “An Extensible Markup Language (XML) Configuration Access Protocol (XCAP) Usage for Manipulating Presence Document Contents,” RFC 4827, May 2007.
[18]Rosenberg, J., “Presence Authorization Rules,” RFC 5025, December 2007.
[19]Lonnfors, M., and K. Kiss, “Session Initiation Protocol (SIP) User Agent Capability Extension to Presence Information Data Format (PIDF),” RFC 5196, September 2008.
Presence and Instant Messaging |
217 |
[20]Urpalainen, J., “An Extensible Markup Language (XML) Patch Operations Framework Utilizing XML Path Language (XPath) Selectors,” RFC 5261, September 2008.
[21]Lonnfors, M., et al., “Presence Information Data Format (PIDF) Extension for Partial Presence,” RFC 5262, September 2008.
[22]Lonnfors, M., et al., “Session Initiation Protocol (SIP) Extension for Partial Notification of Presence Information,” RFC 5263, September 2008.
[23]Niemi, A., M. Lonnfors, and E. Leppanen, “Publication of Partial Presence Information,” RFC 5264, September 2008.
[24]Peterson, J., “Address Resolution for Instant Messaging and Presence,” RFC 3861, August 2004.
[25]Niemi, A., “An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification,” draft-ietf-sipcore-subnot-etags-20 (work in progress), April 2009.
[26]Garcia-Martin, M., and G. Camarillo, “Extensible Markup Language (XML) Format Extension for Representing Copy Control Attributes in Resource Lists,” RFC 5364, October 2008.
[27]Campbell, B., et al., “Session Initiation Protocol (SIP) Extension for Instant Messaging,” RFC 3428, December 2002.
[28]Schulzrinne, H., “Indication of Message Composition for Instant Messaging,” RFC 3994, January 2005.
[29]Garcia-Martin, M., and G. Camarillo, “Multiple-Recipient MESSAGE Requests in the Session Initiation Protocol (SIP),” RFC 5365, October 2008.
[30]Campbell, B., R. Mahy, and C. Jennings, “The Message Session Relay Protocol (MSRP),” RFC 4975, September 2007.
[31]Jennings, C., R. Mahy, and A. Roach, “Relay Extensions for the Message Sessions Relay Protocol (MSRP),” RFC 4976, September 2007.
[32]Klyne, G., and D. Atkins, “Common Presence and Instant Messaging (CPIM): Message Format,” RFC 3862, August 2004.
[33]Burger, E., and H. Khartabil, “Instant Message Disposition Notification (IMDN),” RFC 5438, February 2009.
[34]Niemi, A., M. Garcia, and G. Sandbakken, “Multi-Party Chat Using the Message Session Relay Protocol (MSRP),” draft-ietf-simple-chat-04 (work in progress), March 2009.
[35]Holmberg, C., and S. Blau, “An Alternative Connection Model for the Message Session Relay Protocol (MSRP),” draft-ietf-simple-msrp-acm-01 (work in progress), August 2009.
[36]Camarillo, G., A. Roach, and O. Levin, “Subscriptions to Request-Contained Resource Lists in the Session Initiation Protocol (SIP),” RFC 5367, October 2008.
[37]Levin, O., “Ad-Hoc Resource Lists Using SUBSCRIBE in SIMPLE,” draft-levin-simple- adhoc-list-01.txt (work in progress), November 2003.
[38]Khartabil, H., et al., “Functional Description of Event Notification Filtering,” RFC 4660, September 2006.
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[39]http://www.xsf.org.
[40]Saint-Andre, P., ( ed.), “Extensible Messaging and Presence Protocol (XMPP): Core,” RFC 3920, October 2004.
[41]Saint-Andre, P., (ed.), “Extensible Messaging and Presence Protocol (XMPP): Instant Messaging and Presence,” RFC 3921, October 2004.
[42]Saint-Andre, P., “Basic Messaging and Presence Interworking Between the Extensible Messaging and Presence Protocol (XMPP) and Session Initiation Protocol (SIP) for Instant Messaging and Presence Leveraging Extensions (SIMPLE),” draft-saintandre-xmpp- simple-10 (work in progress), August 2007.
[43]Saint-Andre, P., “Internationalized Resource Identifiers (IRIs) and Uniform Resource Identifiers (URIs) for the Extensible Messaging and Presence Protocol (XMPP),” RFC 4622, July 2006.
[44]Saint-Andre, P., S. Loreto, and F. Forno, “Interworking Between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Multi-Party Text Chat,” draft-saintandre-sip-xmpp-groupchat-01 (work in progress), March 2009.
[45]Ludwig, S., et al., “Jingle,” XSF XEP 0166, June 2007.
[46]Saint-Andre, P., “Interworking Between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions,” draft-saintandre- sip-xmpp-media-01.txt (work in progress), March 2009.
[47]http://www.ietf.org/html.charters/xmpp-charter.html.