SIP IMS Specifications For Dummies v7 4 PDF

SIPKnowledge
SIP/IMS Specifications For Dummies

This e-BOOK provides a categorized list of SIP/IMS specification and a brief
explanation of each one of them
Document ID: SK_500195

Edition: 7.4
Date: July, 2012
Author: SIPKnowledge
Key sources: IETF RFC 5411, 3GPP TS 21.202, 3GPP meetings.
Dedicated To: http://www.youtube.com/watch?v=Qmg5VUMcTn8&feature=related
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This document and the information contained in it is registered information of SIPKnowledge, and
shall not be used, or publish, or disclosed, or disseminated outside of SIPKnowledge in whole or
in part without SIPKnowledges consent.
Note: A special thank you to Doctor SIP, Ph.D. Jonathan Rosenberg, who has paved the SIP way
in the IP telephony galaxy for many followers; the author of this eBook among them.
Table of Contents
Table of Contents ............................................................................................................ 2
1. Introduction ................................................................................................................. 4
1.1 Purpose of this e-book ...................................................................................... 4
1.2 Acronyms, Abbreviations and Symbols ............................................................. 4
1.3 Conventions ...................................................................................................... 5
1.4 The SIP/IMS standard bodies and how they relate to each other ...................... 5
1.5 SIP/VoIP related IETF Work Groups (WGs) ...................................................... 6
1.6 IMS related 3GPP Specification Groups (SGs) ................................................. 8
1.7 Download Tips: ................................................................................................. 9
2. SIP Specifications (developed by IETF) .................................................................... 10
2.1 Scope of the SIP specifications listed in this e-book. ....................................... 10
2.2 Core SIP Specifications................................................................................... 11
2.2.1 Overview .................................................................................................. 11
2.2.2 List of specifications ................................................................................. 11
2.3 Public Switched Telephone Network (PSTN) Interworking .............................. 14
2.3.1 Overview .................................................................................................. 14
2.4 General Purpose Infrastructure Extensions ..................................................... 15
2.4.1 Overview .................................................................................................. 15
2.5 NAT Traversal ................................................................................................. 17
2.5.1 Overview .................................................................................................. 17
2.6 Minor Extensions............................................................................................. 17
2.6.1 Overview .................................................................................................. 17
2.7 Conferencing................................................................................................... 20
2.7.1 Overview .................................................................................................. 20
2.8 Call Control Primitives ..................................................................................... 20
2.8.1 Overview .................................................................................................. 20
2.9 Event Framework and Packages..................................................................... 21
2.9.1 Overview .................................................................................................. 21
2.10 Quality of Service ........................................................................................ 23
2.10.1 Overview .................................................................................................. 23
2.11 Operations and Management ...................................................................... 23
2.11.1 Overview .................................................................................................. 23
2.12 SIP Compression ........................................................................................ 24
2.12.1 Overview ................................................................................................. 24
2.12.2 List of specifications ................................................................................ 24
2.13 SIP Service URIs ........................................................................................ 24
2.13.1 Overview ................................................................................................. 24
2012, SIPKnowledge SIP/IMS Specifications for Dummies, Ed. 7.4, July/2012 Page: 2
2.14 Security Mechanisms.................................................................................. 25
2.14.1 Overview ................................................................................................. 25
2.15 Instant Messaging, Presence and Multimedia ............................................. 27
2.15.1 Overview ................................................................................................. 27
2.16 Emergency Services ................................................................................... 28
2.16.1 Overview ................................................................................................. 28
3. IMS Specifications (developed by 3GPP) .............................................................. 29
3.1 Scope and release of the IMS specifications listed in this e-book .................... 29
3.2 Why only 3GPP specs? How about 3GPP2, TISPAN, CableLabs, Broadband
Forum and OMA? ...................................................................................................... 29
3.3 IMS Main functionality ..................................................................................... 30
3.4 Wide-scope, generic documents (e.g. service brokering) ................................ 30
3.5 Non-SIP Interfaces .......................................................................................... 31
3.6 QoS Support (Policy and Charging Control) .................................................... 32
3.7 Intelligent Networks Support ........................................................................... 34
3.8 Identities, Profile and User data ...................................................................... 34
3.9 Network Interworking and Combining .............................................................. 35
3.10 Accounting/Charging ................................................................................... 37
3.11 Security ....................................................................................................... 38
3.12 Lawful Intercept ........................................................................................... 39
3.13 Group Management ..................................................................................... 39
3.14 Presence ..................................................................................................... 39
3.15 Emergency Services .................................................................................... 40
3.16 Conferencing ............................................................................................... 40
3.17 Messaging and Push To Talk ...................................................................... 40
3.18 I-WLAN Interworking with IMS ..................................................................... 42
3.19 Seamless Mobility (AKA Voice Call Continuity (VCC)) ................................. 42
3.20 Media handling and characteristics (codecs) ............................................... 43
3.21 End Point requirements ............................................................................... 43
3.22 Location ....................................................................................................... 44
3.23 Access Network Aspects (3GPP (LTE, UTRAN, GERAN) and non-3GPP
(3GPP2, WiFi, Wireline, Femto)) ............................................................................... 45
4. References ............................................................................................................ 46
4.1 IETF References ............................................................................................. 46
4.2 3GPP References ........................................................................................... 57
Appendix A a brief summary of all Common IMS specs (3GPP specs, which are
common across 3GPP, 3GPP2 and TISPAN). .............................................................. 63
Appendix B - Core IMS Specifications transferred from ETSI TISPAN to 3GPP ............ 68
Appendix C - IMS related Specifications and Reports in 3GPP2 ................................... 70
1. Introduction
1.1 Purpose of this e-book
It is quite easy to get lost in the jungle of SIP and IMS specifications. The enabler
protocols of these technologies (e.g. SIP, RTP, DNS, Diameter) are the subject of
numerous specifications that have been produced by different standard organizations
such as IETF, 3GPP, 3GPP2 and TISPAN. It can be difficult to locate the right
document, or even to determine the set of specs about these protocols.
Don't Panic! This paper serves as a guide to the SIP/IMS specs. It lists the
specifications under the SIP/IMS umbrella, briefly summarizes each, and groups them
into categories.
1.2 Acronyms, Abbreviations and Symbols

Table 1: Acronyms, Abbreviations and Symbols
Acronym, Description
Abbreviation
or Symbol
3GPP 3rd Generation Partnership Project (GSM/GPRS/UMTS access)
3GPP2 3rd Generation Partnership Project 2 (CDMA access)
CSCF Call Session Control Function
I-D Internet Draft
IETF Internet Engineering Task Force
IMS IP Multimedia Subsystem
ISC IP multimedia Service Control
MIME Multipurpose Internet Mail Extensions
NAT Network Address Translator
OMA Open Mobile Alliance
P-CSCF Proxying Call Session Control Function
RFC Request For Comment
SDP Session Description Protocol
SIP Session Initiation Protocol
TISPAN Telecoms & Internet converged Services & Protocols for Advanced
Networks
TR Technical Report (by 3GPP)
TS Technical Specification (by 3GPP)
1.3 Conventions
Reference number
[number] = [reference number in the list of references], e.g. [1] = reference #1 in the list
of references = RFC 3261.
IETF Standards-track-category (per section 4.1)
(S) = Standards Track (Proposed Standard, Draft Standard, or Standard)
(E) = Experimental
(B) = Best Current Practice
(I) = Informational
1.4 The SIP/IMS standard bodies and how they relate to each other
IETF (www.ietf.org) - Short for Internet Engineering Task Force, the main standards
organization for the Internet. The IETF is a large open international community of
network designers, operators, vendors, and researchers concerned with the evolution of
the Internet architecture and the smooth operation of the Internet. It is open to any
interested individual. IETF has the (IP Telephony community) charter of developing
the SIP standard. Sometimes 3GPP or/and any other SIP/IMS related standard group
identifies the need to extend SIP in order to satisfy their unique environment
requirements. Normally when that happens they will try to push to IETF as an extension
to SIP in order to get the consensus of the SIP community. This guarantees that SIP
standardization takes place only within IETF. This arrangement makes everyones
life easier, and help to keep things under control.
3GPP (www.3gpp.org) The 3rd Generation Partnership Project (3GPP) is a

collaboration agreement that was established in December 1998. It is a co-operation
between ETSI (Europe), ARIB/TTC (Japan), CCSA [1] (China), ATIS (North America)
and TTA (South Korea). The scope of 3GPP is to make a globally applicable third
generation (3G) mobile phone system specification within the scope of the ITU's IMT-
2000 project. 3GPP specifications are based on evolved GSM specifications, now
generally known as the UMTS system.
3GPP has the (IP Telephony community) charter of developing the IMS standard.
Other IMS related standard organizations are briefly mentioned below. They represent
special environments (AKA Access networks, e.g. Cable, Wireline), and as such they try
to make sure that:
1. IMS does not break when operates over their environments.
2. IMS can fulfill the special needs of these environments, e.g. what would be the default
codec that is a good fit for the particular access, or where do you store the user security
info when your endpoint is not a GPRS phone with a smart card.
All in all these groups decided to let 3GPP continue develop most of the IMS standard.
As such they mostly link their IMS specs to the 3GPP ones, with the exception of
generating and maintaining environment-related delta (e.g. IMS behavior of Set Top Box
in the cable environment or residential gateway in the DSL environment). Just like
3GPP tries to push SIP extensions to IETF, these groups try to push IMS
extensions to 3GPP.
3GPP2 (http://www.3gpp2.org CDMA2000), TISPAN (http://www.etsi.org/tispan/ -

Fixed networks, e.g. DSL), CableLabs/PacketCable (http://www.packetcable.com -
Cable), OMA (http://www.openmobilealliance.org - Mobile-data-services (e.g.
PoC/Messaging)), Wi-fi Alliance (http://www.wi-fi.org wi-fi) and the WiMAX Forum
(http://www.wimaxforum.org Wi-MAX Backhaul) represent special environments, which
have interest in IMS. As indicated above, they mostly link their IMS specs to the 3GPP
ones. In addition they maintain environment related delta (with desire to push it to 3GPP,
so their delta becomes part of the IMS main stream).
Note: In the current Edition we will discuss only the IMS main stream specifications, i.e.
the specs generated by 3GPP. In following editions we may summarize the key delta
between those and the specific environments mentioned above. In general as the year
2010 IMS standard work is mostly done 3GPP. This is done in a harmonized way, such
that non 3GPP traditional environments (e.g. fixed access) are well covered.
1.5 SIP/VoIP related IETF Work Groups (WGs)
Conferencing
- The XCON WG - Centralized Conferencing - http://www.ietf.org/html.charters/xcon-

charter.html.
Data for Reachability/Routing
- The DRINKS WG - Data for Reachability of Inter/tra-NetworK SIP -

http://www.ietf.org/dyn/wg/charter/drinks-charter.html.
Features interop
- The BLISS WG - Basic Level of Interoperability for SIP Services -

http://www.ietf.org/dyn/wg/charter/bliss-charter.html.
Instant Messaging and Presence and Voice Messaging
- The IMPP WG - Instant Messaging and Presence Protocol

http://tools.ietf.org/wg/impp/ (WG activity has concluded).
- The SIMPLE WG - SIP for Instant Messaging and Presence Leveraging Extensions
- http://www.ietf.org/html.charters/simple-charter.html.
- The XMPP WG - open, XML-based protocol for near real-time extensible messaging
and presence - http://tools.ietf.org/wg/xmpp/ (WG activity has concluded).
- The VPIM WG - Voice Profile for Internet Mail - http://tools.ietf.org/wg/vpim/ (WG
activity has concluded).
IP Telephony (VoIP and beyond)
- The AVT WG - Audio/Video Transport (RTP) -

http://datatracker.ietf.org/wg/avt/charter/ .
- The IPTEL WG - IP Telephony (CPL, GW location, TRIP) -
http://www.ietf.org/proceedings/61/iptel.html (WG activity has concluded).
- The MMUSIC WG Multiparty Multimedia Session Control (SIP, SDP,
conferencing) - http://datatracker.ietf.org/wg/mmusic/charter/ .
- The SIP WG - signaling for call setup - http://tools.ietf.org/wg/sip/ - (WG activity has
concluded).
- The SIPPING WG - Session Initiation Proposal Investigation -
http://tools.ietf.org/wg/sipping/ - (WG activity has concluded).
Interop with Circuit domain
- The ENUM WG - Telephone Number Mapping -

http://www.ietf.org/html.charters/enum-charter.html .
- The MEGACO WG Media Gateway Control (IP telephony gateways) -
http://tools.ietf.org/wg/megaco/ - (OLD - WG activity has concluded!)
- The PINT WG PSTN and Internet Internetworking (mixed services) -
http://tools.ietf.org/wg/pint/ - (OLD - WG activity has concluded!)
- The SIGTRAN WG Signaling Transport (PSTN signaling over IP) -
http://tools.ietf.org/wg/sigtran/ - (OLD - WG activity has concluded!)
- The SPIRITS WG - Service in the PSTN/IN Requesting InTernet Service -
http://tools.ietf.org/wg/spirits/ - (OLD - WG activity has concluded!)
Media Control
- The MEDIACTRL WG - Media Server Control -

http://datatracker.ietf.org/wg/mediactrl/charter/ .
- The SPEECHSC WG - Speech Services Control -
http://www.ietf.org/html.charters/speechsc-charter.html .
Misc. (Location Based Services, Compression, Interaction with Firewalls/NATs)
- The GEOPRIV WG - Geographic Location/Privacy

http://www.ietf.org/html.charters/geopriv-charter.html .
- The ROHC WG - Robust Header Compression (SigComp) -
http://www.ietf.org/html.charters/rohc-charter.html .
- The MIDCOM WG - Middlebox Communication (NAT, IPV4-IPV6) -
http://tools.ietf.org/wg/midcom/ - (OLD - WG activity has concluded!)
- The BEHAVE WG - Behavior Engineering for Hindrance Avoidance -
http://www.ietf.org/dyn/wg/charter/behave-charter.html .
QoS
- The DIFFSERV WG Differentiated Services (QoS in backbone) -

http://tools.ietf.org/wg/diffserv/ - (OLD - WG activity has concluded!)
- The INTSERV WG - Integrated Services (end-to-end QoS) -
http://tools.ietf.org/wg/intserv/ - (OLD - WG activity has concluded!)
- The MPLS WG - Multiprotocol Label Switching -
http://www.ietf.org/html.charters/mpls-charter.html .
- The RSVP WG - Resource Reservation Setup Protocol -
http://tools.ietf.org/wg/rsvp/ - (OLD - WG activity has concluded!)
Peer-to-Peer paradigm for SIP
- The P2PSIP WG - Peer-to-Peer Session Initiation Protocol -

http://www.ietf.org/html.charters/p2psip-charter.html .
- The SPEERMINT WG - Session PEERing for Multimedia INTerconnect -
http://www.ietf.org/dyn/wg/charter/speermint-charter.html .
Security and Emergency Calls (911)
- The MSEC WG - Multicast Security - http://www.ietf.org/html.charters/msec-

charter.html .
- The ECRIT WG - Emergency Context Resolution with Internet Technologies (E911)
- http://www.ietf.org/html.charters/ecrit-charter.html .
1.6 IMS related 3GPP Specification Groups (SGs)
Note: As opposed to IETF SIP Work Groups, whose main focus is SIP and sister
technologies, 3GPP Specification Groups deal with areas above and beyond IMS. In
fact, the 3GPP SGs, indicated below, had started their lives before IMS even came to
the 3GPP world.
Services
- The SA1 SG - http://www.3gpp.org/SA1-Services.
Architecture
- The SA2 SG - http://www.3gpp.org/SA2-Architecture.
Security
- The SA3 SG - http://www.3gpp.org/SA3-Security.
Codec
- The SA4 SG - http://www.3gpp.org/SA4-Codec.
Telecom-Management
- The SA5 SG - http://www.3gpp.org/SA5-Telecom-Management.
Call Control, Session Management, Mobility Management, SMS

- The CT1 SG - http://www.3gpp.org/CT1-MM-CC-SM-lu.
Interworking-with-External Networks
- The CT3 SG - Services - http://www.3gpp.org/CT3-Interworking-with-External.
MAP/CAMEL/GTP/BCH/SS/TrFO/IMS/GUP/WLAN
- The CT4 SG - Services - http://www.3gpp.org/CT4-MAP-CAMEL-GTP-BCH-SS-
TrFO-IMS.
OSA - Open Service Access (transferred to OMA -

http://www.openmobilealliance.org/)
- The CT5 SG - Services - http://www.3gpp.org/CT5.
Smart Card Application Aspects

- The CT6 SG - Services - http://www.3gpp.org/CT6-Smart-Card-Application-Access.
1.7 Download Tips:
Download Tips: You can download an IETF RFC by going to

http://www.ietf.org/rfc.html and entering the RFC number in the dialog box. You can
download an IETF I-D (Internet Draft) by going to http://www.rfc-editor.org/ and
selecting the I-D search link at the top (almost all the way to the right). The
http://www.rfc-editor.org/ link also enables you to lookup RFCs by name or keywords.
Finally you can download a 3GPP spec by going to
http://www.3gpp.org/specs/numbering.htm and selecting the appropriate series
number (e.g. say you want to lookup and download the latest version of 23.228 you
would click on that link and then dive into series 23 (in the first table you come
across).
2. SIP Specifications (developed by IETF)
2.1 Scope of the SIP specifications listed in this e-book.
It is very difficult to enumerate the set of SIP specifications. This is because there are
many protocols that are intimately related to SIP and used by nearly all SIP
implementations, but are not formally SIP extensions. As such, in this e-book we will
refer to a "SIP specification" as:
Any specification that defines an extension to SIP itself, where an extension is a

mechanism that changes or updates in some way a behavior specified in RFC
3261.
Any specification that defines an extension to SDP whose primary purpose is to

support SIP.
Any specification that defines a MIME object whose primary purpose is to

support SIP
We did not include requirements, architectures, registry, definitions, non-normative

frameworks, and processes. Best Current Practices are included when they are
effectively standard mechanisms for accomplishing a task.
The SIP change process [8] defines two types of extensions to SIP. These are normal
extensions and the so-called P-headers, which are meant to be used in areas of limited
applicability. P-headers cannot be defined in the standards track. For the most part, P-
headers are not included in the listing here, with the exception of those which have
seen general usage despite their P-header status (Most of the P-header related
specifications were pushed to IETF by the IMS standard organization(s)).
Each specification below also includes its category in the standards track [2].
The possible (mutually exclusive) values for standards-track-category are:
1. Standards Track (Proposed Standard, Draft Standard, or Standard)
2. Experimental
3. Best Current Practice
4. Informational
These values will be denoted below by a single letter within brackets: (S) = Standards
Track , (E) = Experimental, (B) = Best Current Practice, (I) = Informational.
2.2 Core SIP Specifications
2.2.1 Overview
The core SIP specifications represent the set of specifications whose functionality is
broadly applicable. An extension is broadly applicable if it fits into one of the following
categories:
For specifications that impact SIP session management, the extension would be
used for almost every session initiated by a user agent.
For specifications that impact SIP registrations, the extension would be used for
almost every registration initiated by a user agent
For specifications that impact SIP subscriptions, the extension would be used for
almost every subscription initiated by a user agent
In other words, these are not specifications that are used just for some requests and not
others; they are specifications that would apply to each and every request that the
extension is relevant for.
2.2.2 List of specifications
RFC 3261, The Session Initiation Protocol (S): RFC 3261 [1] is the core SIP protocol
itself. RFC 3261 is an update to RFC 2543 [9]. It is the head of staff in the army of SIP
specifications.
RFC 3263, Locating SIP Servers (S): RFC 3263 [10] provides DNS procedures for
taking a SIP URI, and determining a SIP server that is associated with that SIP URI.
RFC 3263 is essential for any implementation using SIP with DNS. RFC 3263 makes
use of both DNS SRV records [11] and NAPTR records [12].
RFC 3264, An Offer/Answer Model with the Session Description Protocol (S): RFC 3264
[4] defines how the Session Description Protocol (SDP) [78] is used with SIP to
negotiate the parameters of a media session. It is in widespread usage and an integral
part of the behavior of RFC 3261.
RFC 3265, SIP-Specific Event Notification (S): RFC 3265 [13] defines the SUBSCRIBE
and NOTIFY methods. These two methods provide a general event notification
framework for SIP. To actually use the framework, extensions need to be defined for
specific event packages. An event package defines a schema for the event data, and
describes other aspects of event processing specific to that schema. An RFC 3265
implementation is required when any event package is used.
RFC 3325, Private Extensions to SIP for Asserted Identity within Trusted Networks (I):
Though its P-header status implies that it has limited applicability, RFC 3325 [15], which
defines the P-Asserted-ID header field has been widely deployed. It is used as the basic
mechanism for providing secure caller ID services. Its update, [I-D.ietf-sipping-update-
pai][117], clarifies its usage for connected party identification as well.
RFC 3327, SIP Extension Header Field for Registering Non-Adjacent Contacts (S): RFC
3327 [16] defines the Path header field. This field is inserted by proxies between a client
and their registrar. It allows inbound requests towards that client to traverse these
proxies prior to being delivered to the user agent. It is essential in any SIP deployment
that has edge proxies, which are proxies between the client and the home proxy or SIP
registrar (e.g. IMS P-CSCF is an edge proxy between the user agent and the home
proxy/SIP registrar, which is the IMS S-CSCF).
RFC 3581, An Extension to SIP for Symmetric Response Routing (S): RFC 3581 [17]
defines the rport parameter of the Via header. It is an essential piece of getting SIP
through NAT. NAT traversal for SIP is considered a core part of the specifications.
RFC 3840, Indicating User Agent Capabilities in SIP (S): RFC 3840 [33] defines a
mechanism for carrying capability information about a user agent in REGISTER requests
and in dialog-forming requests like INVITE. It has found use with conferencing (the
isfocus parameter declares that a user agent is a conference server) and with
applications like push-to-talk.
RFC 4320, Actions Addressing Issues Identified with the Non-INVITE Transaction in SIP
(S): RFC 4320 [18] formally updates RFC 3261, and modifies some of the behaviors
associated with non-INVITE transactions. These address some problems found in
timeout and failure cases.
RFC 4474, Enhancements for Authenticated Identity Management in SIP (S): RFC 4474
[19] defines a mechanism for providing a cryptographically verifiable identity of the
calling party in a SIP request. Also known as "SIP Identity", this mechanism provides an
alternative to RFC 3325. It has seen little deployment so far, but its importance as a key
construct for anti-spam techniques makes it a core part of the SIP specifications.
RFC 5627, Obtaining and Using Globally Routable User Agent Identifiers (GRUU) in
SIP (S): RFC 5627 [20] defines a mechanism for directing requests towards a specific
UA instance. GRUU is essential for features like transfer and provides another piece of
the SIP NAT traversal story.
RFC 5626, Managing Client Initiated Connections through SIP (S): RFC 5626 [21],
defines important changes to the SIP registration mechanism, which enable delivery of
SIP messages towards a UA when it is behind a NAT. This specification is the
cornerstone of the SIP NAT traversal strategy.
RFC 4566, Session Description Protocol (S): RFC 4566 [78] defines a format for
representing multimedia sessions. SDP objects are carried in the body of SIP
messages, and based on the offer/answer model, are used to negotiate the media
characteristics of a session between users. This RFC updates/replaces the good old
RFC 2327.
RFC 5939, SDP Capability Negotiation (S): I-D.ietf-mmusic-sdp-capability-negotiation

[105] defines a set of extensions to SDP that allow for capability negotiation within SDP.
Capability negotiation can be used to select between different profiles of RTP (secure
vs. unsecure) or to negotiate codecs such that an agent has to select one amongst a set
of supported codecs.
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session Description
Protocol (SDP) (S): RFC 3605 [80] defines a way to explicitly signal, within an SDP
message, the IP address and port for RTCP, rather than using the port+1 rule in the
Real Time Transport Protocol (RTP) [3]. It is needed for devices behind NAT and used
by ICE.
RFC 4916, Connected Identity in the Session Initiation Protocol (SIP) (S): RFC 4916
[81] defines an extension to SIP that allows a UAC to determine the identity of the UAS.
Due to forwarding and retargeting services, this may not be the same as the user that
the UAC was originally trying to reach. The mechanism works in tandem with the SIP
identity specification [19] to provide signatures over the connected party identity.
RFC 3311, The SIP UPDATE Method (S): RFC 3311 [29] defines the UPDATE method
for SIP. This method is meant as a means for updating session information prior to the
completion of the initial INVITE transaction. It can also be used to update other
information, such as the identity of the participant [RFC4916], without involving an
updated offer/answer exchange. It was developed initially to support RFC3312 but has
found other uses. In particular, its usage with RFC 4916 means it will typically be
used as part of every session, to convey a secure connected identity.
RFC 5630, The use of the SIPS URI Scheme in the Session Initiation Protocol (SIP) (S):
RFC 5630 [112] revises the processing of the SIPS URI, originally defined in RFC 3261,
to fix many errors and problems that have been encountered with that mechanism.
RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples (B): RFC 3665
[113] contains best practice call flow examples for basic SIP interactions - call
establishment, termination, and registration.
RFC 5638, Simple SIP Usage Scenario for Applications in the Endpoints (I): [151] For
Internet-centric usage, the number of SIP-required standards for presence and IM and
audio/video communications can be drastically smaller than what has been published by
using only the rendezvous and session-initiation capabilities of SIP. The simplification is
achieved by avoiding the emulation of telephony and its model of the intelligent network.
'Simple SIP' relies on powerful computing endpoints. Simple SIP desktop applications
can be combined with rich Internet applications (RIAs). Significant telephony features
may also be implemented in the endpoints.
This approach for SIP reduces the number of SIP standards with which to comply -- from
roughly 100 currently, and still growing, to about 11.
2.3 Public Switched Telephone Network (PSTN) Interworking
2.3.1 Overview
Numerous extensions and usages of SIP related to interoperability and

communications with or through the PSTN.
RFC 2848, The PINT Service Protocol (S): RFC 2848 [22] is one of the earliest
extensions to SIP. It defines procedures for using SIP to invoke services that actually
execute on the PSTN. Its main application is for third party call control, allowing an IP
host to set up a call between two PSTN endpoints. PINT has a relatively narrow focus
and has not seen widespread deployment.
RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming PSTN related
extensions with alcohol references -), SPIRITS [23] defines the inverse of PINT. It
allows a switch in the PSTN to ask an IP element about how to proceed with call waiting.
It was developed primarily to support Internet Call Waiting (ICW).
RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I): SIP-T [24] defines
a mechanism for using SIP between pairs of PSTN gateways. Its essential idea is to
tunnel ISUP signaling between the gateways in the body of SIP messages. SIP-T
motivated the development of INFO [30]. SIP-T has seen widespread implementation.
RFC 3398, ISUP to SIP Mapping (S): RFC 3398 [25] defines how to do protocol
mapping from the SS7 ISDN User Part (ISUP) signaling to SIP. It is widely used in SS7
to SIP gateways and is part of the SIP-T framework.
RFC 4497, Interworking between the Session Initiation Protocol (SIP) and QSIG (B):
RFC 4497 [114] defines how to do protocol mapping from Q.SIG, used for PBX
signaling, to SIP.
RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): RFC 3578 [26] defines a
mechanism to map overlap dialing into SIP. This specification is widely regarded as the
ugliest SIP specification, as the introduction to the specification itself advises that it has
many problems. Overlap signaling (the practice of sending digits into the network as
dialed instead of waiting for complete collection of the called party number) is largely
incompatible with SIP at some fairly fundamental levels. That said, RFC 3578 is mostly
harmless and has seen some usage.
RFC 3960, Early Media and Ringtone Generation in SIP (I): RFC 3960 [27] defines
some guidelines for handling early media the practice of sending media from the called
party towards the caller prior to acceptance of the call. Early media is generated only
from the PSTN.
RFC 3959, Early Session Disposition Type for the Session Initiation Protocol (SIP) (S):
RFC 3959 [83] defines a new session disposition type for use with early media. It
indicates that the SDP in the body is for a special early media session.
RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): RFC 3204 [84] defines
MIME objects for representing SS7 and QSIG signaling messages. SS7 signaling
messages are carried in the body of SIP messages when SIP-T is used. QSIG signaling
messages can be carried in a similar way.
RFC 3666, Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN)
Call Flows (B): RFC 3666 [115] provides best practice call flows around interworking
with the PSTN.
2.4 General Purpose Infrastructure Extensions

2.4.1 Overview
These extensions are general purpose enhancements to SIP, SDP and MIME that can
serve a wide variety of uses. However, they are not as widely used or as essential as
the core specifications.
RFC 3262, Reliability of Provisional Responses in SIP (S): SIP defines two types of
responses to a request - final and provisional. Provisional responses are numbered
from 100 to 199. In SIP, these responses are not sent reliably. This choice was made in
RFC 2543, since the messages were meant to just be truly informational, and rendered
to the user. However, subsequent work on PSTN interworking demonstrated a need to
map provisional responses to PSTN messages that needed to be sent reliably. RFC
3262 [28] was developed to allow reliability of provisional responses. The specification
defines the PRACK method, used for indicating that a provisional response was
received. Though it provides a generic capability for SIP, RFC 3262 implementations
have been most common in PSTN interworking devices. However, PRACK brings a
great deal of complication for relatively small benefit. As such, it has seen only mild
levels of deployment.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol (SIP) (S): RFC 3323
[14] defines the Privacy header field, used by clients to request anonymity for their
requests. Though it defines numerous privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-ID header field [15].
RFC 2976, The INFO Method (S): RFC 2976 [30] was defined as an extension to RFC
2543. It defines a method, INFO, used to transport mid-dialog information that has no
impact on SIP itself. Its driving application was the transport of PSTN related
information when using SIP between a pair of gateways. Though originally conceived
for broader use, it only found standardized usage with SIP-T [24]. It has been used to
support numerous proprietary and non-interoperable extensions due to its poorly
defined scope.
RFC 3326, The Reason header field for SIP (S): RFC 3326 [31] defines the Reason
header field. It is used in requests, such as BYE, to indicate the reason that the request
is being sent.
RFC 3388, Grouping of Media Lines in the Session Description Protocol (S): RFC 3388
[79] defines a framework for grouping together media streams in an SDP message.
Such a grouping allows relationships between these streams, such as which stream is
the audio for a particular video feed, to be expressed.
RFC 3420, Internet Media Type message/sipfrag (S): RFC 3420 [85] defines a MIME
object that contains a SIP message fragment. Only certain header fields and parts of
the SIP message are present. For example, it is used to report back on the responses
received to a request sent as a consequence of a REFER.
RFC 3608, SIP Extension Header Field for Service Route Discovery During Registration
(S): RFC 3608 [32] allows a client to determine, from a REGISTER response, a path of
proxies to use in requests it sends outside of a dialog. In many respects, it is the inverse
of the Path header field, but has seen less usage since default outbound proxies have
been sufficient in many deployments.
RFC 3841, Caller Preferences for SIP (S): RFC 3841 [34] defines a set of headers that a
client can include in a request to control the way in which the request is routed
downstream. It allows a client to direct a request towards a UA with specific capabilities.
RFC 4028, Session Timers in SIP (S): RFC 4028 [35] defines a keep alive mechanism
for SIP signaling. It is primarily meant to provide a way to cleanup old state in proxies
that are holding call state for calls from failed endpoints which were never terminated
normally. Despite its name, the session timer is not a mechanism for detecting a
network failure mid-call. Session timers introduces a fair bit of complexity for relatively
little gain, and has thus seen little deployment.
RFC 4168, SCTP as a Transport for SIP (S): RFC 4168 [36] defines how to carry SIP
messages over the Stream Control Transmission Protocol (SCTP). SCTP has seen very
limited usage for SIP transport.
RFC 4244, An Extension to SIP for Request History Information (S): RFC 4244 [37]
defines the History-Info header field, which indicates information on how a call came to
be routed to a particular destination. Its primary application was in support of
voicemail services.
RFC 4145, TCP-Based Media Transport in the Session Description Protocol (SDP) (S):
RFC 4145 [86] defines an extension to SDP for setting up TCP-based sessions between
user agents. It defines who sets up the connection and how its lifecycle is managed. It
has seen relatively little usage due to the small number of media types to date which use
TCP.
RFC 4091, The Alternative Network Address Types (ANAT) Semantics for the Session
Description Protocol (SDP) Grouping Framework (S): RFC 4091 [87] defines a
mechanism for including both IPv4 and IPv6 addresses for a media session as
alternates.
RFC 3407, SDP Simple Capability Declaration (S): RFC 3407 [106] defines a set of
Session Description Protocol (SDP) attributes that enables SDP to provide a minimal
and backwards compatible capability declaration mechanism.
RFC 5621, Message Body Handling in the Session Initiation Protocol (SIP): RFC 5621
[119] clarifies handling of bodies in SIP, focusing primarily on multi-part behavior, which
was underspecified in SIP.
2.5 NAT Traversal

2.5.1 Overview
These SIP extensions are primarily aimed at addressing NAT traversal for SIP.
Draft-ietf-mmusic-ice-tcp, TCP Candidates with Interactive Connectivity Establishment

(ICE) (S): I-D.ietf-mmusic-ice-tcp [88] specifies the usage of ICE for TCP streams. This
allows for selection of RTP-based voice on top of TCP only when NAT or firewalls would
prevent UDP-based voice from working.
RFC 5425, Interactive Connectivity Establishment (ICE): A Protocol for Network Address
Translator (NAT) Traversal for Offer/Answer Protocols (S): RFC 5425 [150] describes a
protocol for Network Address Translator (NAT) traversal for UDP-based multimedia
sessions established with the offer/answer model. This protocol is called Interactive
Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for
NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). ICE can
be used by any protocol utilizing the offer/answer model, such as the Session Initiation
Protocol (SIP).
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session Description
Protocol (SDP) (S): RFC 3605 [80] defines a way to explicitly signal, within an SDP
message, the IP address and port for RTCP, rather than using the port+1 rule in the
Real Time Transport Protocol (RTP) [3]. It is needed for devices behind NAT and used
by ICE.
RFC 3581, An Extension to SIP for Symmetric Response Routing (S): RFC 3581 [17]
defines the rport parameter of the Via header. It is an essential piece of getting SIP
through NAT. NAT traversal for SIP is considered a core part of the specifications.
2.6 Minor Extensions
2.6.1 Overview
These SIP extensions don't fit easily into a single specific use case. They have
somewhat general applicability, but they solve a relatively small problem or provide an
optimization.
RFC 4488, Suppression of the SIP REFER Implicit Subscription (S): RFC 4488 [38]
defines an enhancement to REFER. REFER normally creates an implicit subscription to
the target of the REFER. This subscription is used to pass back updates on the
progress of the referral. This extension allows that implicit subscription to be
bypassed as an optimization.
RFC 4538, Request Authorization through Dialog Identification in SIP (S): RFC 4538 [39]
provides a mechanism that allows a UAS to authorize a request because the requestor
proves it knows a dialog that is in progress with the UAS. The specification is useful in
conjunction with the SIP application interaction framework [77].
RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S): RFC 4508 [40]
defines a mechanism for carrying RFC 3840 feature tags in REFER. It is useful for
informing the target of the REFER about the characteristics of the REFER target.
RFC 5373, Requesting Answer Modes for SIP (S): RFC 5373 [41] defines an extension
for indicating to the called party whether or not the phone should ring and/or be
answered immediately. This is useful for push-to-talk and for diagnostic applications.
(The Push To Talk folks refer to it as alert mode versus barge mode)
RFC 5079, Rejecting Anonymous Requests in SIP (S): RFC 5079 [43] defines a
mechanism for a called party to indicate to the calling party that a call was rejected since
the caller was anonymous. This is needed for implementation of the Anonymous Call
Rejection (ACR) feature in SIP.
RFC 5368, Referring to Multiple Resources in SIP (S): RFC 5368 [44] allows a UA
sending a REFER to ask the recipient of the REFER to generate multiple SIP requests,
not just one. This is useful for conferencing, where a client would like to ask a
conference server to eject multiple users.
RFC 4483, A Mechanism for Content Indirection in Session Initiation Protocol (SIP)
Messages (S): RFC 4483 [89] defines a mechanism for content indirection. Instead of
carrying an object within a SIP body, a URL reference is carried instead, and the
recipient dereferences the URL to obtain the object. The specification has potential
applicability for sending large instant messages, but has yet to find much actual use.
RFC 3890, A Transport Independent Bandwidth Modifier for the Session Description
Protocol (SDP) (S): RFC 3890 [90] specifies an SDP extension that allows for the
description of the bandwidth for a media session that is independent of the underlying
transport mechanism. It has seen relatively little usage.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor Control Protocol
(BFCP) Streams (S): RFC 4583 [91] defines a mechanism in SDP to signal floor control
streams that use BFCP. It is used for Push-To-Talk and conference floor control.
RFC 5898, Connectivity Preconditions for Session Description Protocol (SDP)

Media Streams, Connectivity Preconditions for Session Description Protocol Media
Streams (S): RFC 5898 [93] defines a usage of the precondition framework [59]. The
connectivity precondition makes sure that the session doesn't get established until actual
packet connectivity is checked.
RFC 4796, The SDP (Session Description Protocol) Content Attribute (S): RFC 4796
[94] defines an SDP attribute for describing the purpose of a media stream. Examples
include a slide view, the speaker, a sign language feed, and so on.
2.7 Conferencing
2.7.1 Overview
Numerous SIP and SDP extensions are aimed at conferencing as their primary
application.
RFC 4574, The SDP (Session Description Protocol) Label Attribute (S): RFC 4574 [95]
defines an SDP attribute for providing an opaque label for media streams. These labels
can be referred to by external documents, and in particular, by conference policy
documents. This allows a UA to tie together documents it may obtain through
conferencing mechanisms to media streams to which they refer.
RFC 3911, The SIP Join Header Field (S): RFC 3911 [49] defines the Join header field.
When sent in an INVITE, it causes the recipient to join the resulting dialog into a
conference with another dialog in progress.
RFC 4575, A SIP Event Package for Conference State (S): RFC 4575 [56] defines a
mechanism for learning about changes in conference state, including group
membership.
RFC 5366, Conference Establishment Using Request-Contained Lists in SIP (S): RFC
5366 [70] is similar to [68]. However, instead of subscribing to the resource, an INVITE
request is sent to the resource, and it will act as a conference focus and generate an
invitation to each recipient in the list.
RFC4579, Session Initiation Protocol (SIP) Call Control-Conferencing for User Agents
(B): RFC4579 [135] defines best practice procedures and call flows for conferencing.
This includes conference creation, joining, and dial out, amongst other capabilities.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor Control Protocol
(BFCP) Streams (S): RFC 4583 [91] defines a mechanism in SDP to signal floor control
streams that use BFCP. It is used for push-to-talk and conference floor control.
2.8 Call Control Primitives

2.8.1 Overview
Numerous SIP extensions provide a toolkit of dialog and call management techniques.
These techniques have been combined together to build many SIP-based services.
RFC 3515, The REFER Method (S): REFER [45] defines a mechanism for asking a user
agent to send a SIP request. It's a form of SIP remote control, and is the primary tool
used for call transfer in SIP. Beware that not all potential uses of REFER (neither for all
methods nor for all URI schemes) are well defined. Implementors should only use the
well-defined ones, and should not second guess or freely assume behavior for the
others to avoid unexpected behavior of remote UAs, interoperability issues, and other
bad surprises
RFC 3725, Best Current Practices for Third Party Call Control (3pcc) (B): RFC 3725 [46]
defines a number of different call flows that allow one SIP entity, called the controller, to
create SIP sessions amongst other SIP user agents.
RFC 3911, The SIP Join Header Field (S): RFC 3911 [49] defines the Join header field.
When sent in an INVITE, it causes the recipient to join the resulting dialog into a
conference with another dialog in progress.
RFC 3891, The SIP Replaces Header (S): RFC 3891 [47] defines a mechanism that
allows a new dialog to replace an existing dialog. It is useful for certain advanced
transfer services.
RFC 3892, The SIP Referred-By Mechanism (S): RFC 3892 [48] defines the Referred-
By header field. It is used in requests triggered by REFER, and provides the identity of
the referring party to the referred-to party.
RFC 4117, Transcoding Services Invocation in SIP Using Third Party Call Control (I):
RFC 4117 [50] defines how to use 3pcc for the purposes of invoking transcoding
services for a call.
2.9 Event Framework and Packages
2.9.1 Overview
See description of RFC 3265 below.
RFC 3265 defines the SUBSCRIBE and NOTIFY methods. These two methods provide
a general event notification framework for SIP. To actually use the framework,
extensions need to be defined for specific event packages. An event package defines a
schema for the event data and describes other aspects of event processing specific to
that schema. An RFC 3265 implementation is required when any event package is
used.
RFC 3903, SIP Extension for Event State Publication (S): RFC 3903 [51] defines the
PUBLISH method. It is not an event package, but is used by all event packages as a
mechanism for pushing an event into the system.
RFC 4662, A Session Initiation Protocol (SIP) Event Notification Extension for Resource
Lists (S): RFC 4662 [67] defines an extension to RFC 3265 that allows a client to
subscribe to a list of resources using a single subscription. The server, called a
Resource List Server (RLS) will "expand" the subscription and subscribe to each
individual member of the list. It has found applicability primarily in the area of presence,
but can be used with any event package.
[This Internet-Draft is no longer active] Draft-ietf-sip-subnot-etags, An Extension to

Session Initiation Protocol (SIP) Events for Conditional Event Notification (S): I-D.ietf-
sip-subnot-etags [111] defines an extension to RFC 3265 to optimize the performance of
notifications. When a client subscribes, it can indicate what version of a document it
has, so that the server can skip sending a notification if the client is up to date. It is
applicable to any event package.
RFC 3680, A SIP Event Package for Registrations (S): RFC 3680 [52] defines an event
package for finding out about changes in registration state.
RFC 3842, A Message Summary and Message Waiting Indication Event Package for
SIP (S): RFC 3842 [65] defines a way for a user agent to find out about voicemails and
other messages that are waiting for it. Its primary purpose is to enable the voicemail
waiting lamp on most business telephones.
RFC 3856, A Presence Event Package for SIP (S): RFC 3856 [53] defines an event
package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S): RFC 3857 [54],
also known as winfo, provides a mechanism for a user agent to find out what
subscriptions are in place for a particular event package. Its primary usage is with
presence, but it can be used with any event package.
RFC 4235, An INVITE Initiated Dialog Event Package for SIP (S): RFC 4235 [55] defines
an event package for learning the state of the dialogs in progress at a user agent.
RFC 4575, A SIP Event Package for Conference State (S): RFC 4575 [56] defines a
mechanism for learning about changes in conference state, including group
membership.
RFC 4730, A SIP Event Package for Key press Stimulus (KPML) (S): RFC 4730 [57]
defines a way for an application in the network to subscribe to the set of keypresses
made on the keypad of a traditional telephone. It, along with RFC 4733 [137], are the
two mechanisms defined for handling DTMF. RFC 4730 is a signaling-path solution, and
RFC 4733 is a media-path solution.
[This Internet-Draft is no longer active] Draft-ietf-sip-session-policy-framework, A

Framework for Session Initiation Protocol (SIP) Session Policies (S): I-D.ietf-sip-session-
policy-framework [121] defines a framework for session policies. In this framework,
policy servers are used to tell user agents about the media characteristics required for a
particular session. The session policy framework has not been widely implemented.
[This Internet-Draft is no longer active] Draft-ietf-sipping-policy-package, A

Session Initiation Protocol (SIP) Event Package for Session-Specific Session Policies
(S): I-D.ietf-sipping-policy-package [96] defines a SIP event package that allows a proxy
to notify a user agent about its desire for the UA to use certain codecs or generally obey
certain media session policies.
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions Event Package (S):
RFC5362 [140] defines a SIP event package that allows a UA to learn whether consent
has been given for the addition of an address to a SIP "mailing list". It is used in
conjunction with the SIP framework for consent [138](RFC5360).
2.10 Quality of Service

2.10.1 Overview
Several specifications concern themselves with the interactions of SIP with network
Quality of Service (QoS) mechanisms.
RFC 3312, Integration of Resource Management and SIP (S): RFC 3312 [59], updated
by RFC 4032 [60] defines a way to make sure that the phone of the called party doesn't
ring until a QoS reservation has been installed in the network. It does so by defining a
general preconditions framework, which defines conditions that must be true in order for
a SIP session to proceed.
RFC 3313, Private SIP Extensions for Media Authorization (I): RFC 3313 [61] defines a
P-header that provides a mechanism for passing an authorization token between SIP
and a network QoS reservation protocol like RSVP. Its purpose is to make sure network
QoS is only granted if a client has made a SIP call through the same providers network.
This specification is sometimes referred to as the SIP walled garden specification by the
truly paranoid androids in the SIP community. This is because it requires coupling of
signaling and the underlying IP network.
RFC 3524, Mapping of Media Streams to Resource Reservation Flows (S): RFC 3524
[97] defines a usage of the SDP grouping framework for indicating that a set of media
streams should be handled by a single resource reservation.
2.11 Operations and Management
2.11.1 Overview
Several specifications have been defined to support operations and management of SIP
systems. These include mechanisms for configuration and network diagnostics.
Draft-ietf-sipping-config-framework, A Framework for SIP User Agent Profile Delivery

(S): I-D.ietf-sipping-config-framework [123] defines a mechanism that allows a SIP user
agent to bootstrap its configuration from the network, and receive updates to its
configuration should it change. This is considered an essential piece of deploying a
usable SIP network.
2.12 SIP Compression
2.12.1 Overview
Sigcomp [6] was defined to allow compression of SIP messages over low bandwidth
links. Sigcomp is not formally part of SIP. However, usage of Sigcomp with SIP has
required extensions to SIP.
RFC3320/RFC4896 (Sigcomp), was defined to allow compression of SIP messages

over low bandwidth links. Sigcomp is not formally part of SIP. However, usage of
Sigcomp with SIP has required extensions to SIP.
RFC 3486, Compressing SIP (S): RFC 3486 [64] defines a SIP URI parameter that can
be used to indicate that a SIP server supports Sigcomp.
RFC 5049, Applying Signaling Compression (SigComp) to the Session Initiation Protocol
(SIP) (S): RFC 5049 [124] defines how to apply Sigcomp to SIP.
2.13 SIP Service URIs

2.13.1 Overview
Several extensions define well-known services that can be invoked by constructing

requests with the specific structures for the Request URI, resulting in specific behaviors
at the UAS.
RFC 3087, Control of Service Context using Request URI (I): RFC 3087 [66] introduced
the context of using Request URIs, encoded appropriately, to invoke services.
RFC 4662, A SIP Event Notification Extension for Resource Lists (S): RFC 4662 [67]
defines a resource called a Resource List Server. A client can send a SUBSCRIBE
request to this server. The server will generate a series of subscriptions, and compile
the resulting information and send it back to the subscriber. The set of resources that
the RLS will subscribe to is a property of the request URI in the SUBSCRIBE request.
RFC 5363, Framework and Security Considerations for Session Initiation Protocol (SIP)
Uniform Resource Identifier (URI)-List Services (S): RFC 5363 [142] defines the
framework for list services in SIP. In this framework, a UA can include an XML list object
in the body of various requests and the server will provide list-oriented services as a
consequence. For example, a SUBSCRIBE with a list subscribes to the URI in the list.
RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP (S): RFC 5367
[143] uses the URI-list framework [RFC 5363] and allows a client to subscribe to a
resource called a Resource List Server. This server will generate subscriptions to the
URI in the list, compile the resulting information, and send it back to the subscriber.
RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S): RFC 5365 [144] uses
the URI-list framework [RFC5363] and allows a client to send a MESSAGE to a number
of recipients.
RFC 5366, Conference Establishment Using Request-Contained Lists in SIP (S): RFC
5366 [145] uses the URI-list framework [RFC5363]. It allows a client to ask the server to
act as a conference focus and send an invitation to each recipient in the list.
RFC 4240, Basic Network Media Services with SIP (I): RFC 4240 [99] defines a way for
SIP application servers to invoke announcement and conferencing services from a
media server. This is accomplished through a set of defined URI parameters which tell
the media server what to do, such as what file to play and what language to render it in.
RFC 4458, Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and
Interactive Voice Response (IVR) (I): RFC4458 [126] defines a way to invoke voicemail
and IVR services by using a SIP URI constructed in a particular way.
2.14 Security Mechanisms
2.14.1 Overview
Several extensions provide additional security features to SIP.
RFC 3853, S/MIME AES Requirement for SIP (S): RFC 3853 [71] is a brief specification
that updates the cryptography mechanisms used in SIP S/MIME. However, SIP S/MIME
has seen very little deployment.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol (SIP) (S): RFC 3323
[14] defines the Privacy header field, used by clients to request anonymity for their
requests. Though it defines numerous privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-ID header field [15].
RFC 4567, Key Management Extensions for Session Description Protocol (SDP) and
Real Time Streaming Protocol (RTSP) (S): RFC4567 [130] defines extensions to SDP
that allow tunneling of an key management protocol, namely MIKEY [RFC3830], through
offer/answer exchanges. This mechanism is one of three SRTP keying techniques
specified for SIP, with DTLS-SRTP [I-D.ietf-sip-dtls-srtp-framework] having been
selected as the final solution.
RFC 4568, Session Description Protocol (SDP) Security Descriptions for Media Streams
(S): RFC4568 [131] defines extensions to SDP that allow for the negotiation of keying
material directly through offer/answer, without a separate key management protocol.
This mechanism, sometimes called sdescriptions, has the drawback that the media keys
are available to any entity that has visibility to the SDP. It is one of three SRTP keying
techniques specified for SIP, with DTLS-SRTP [I-D.ietf-sip-dtls-srtp-framework] having
been selected as the final solution.
Draft-ietf-sip-certs, Certificate Management Service for The Session Initiation Protocol

(SIP) (S): I-D.ietf-sip-certs [100] defines a certificate service for SIP whose purpose is to
facilitate the deployment of S/MIME. The certificate service allows clients to store and
retrieve their own certificates, in addition to obtaining the certificates for other users.
RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity Body (AIB) Format
(S): RFC 3893 [7] defines a SIP message fragment, which can be signed in order to
provide an authenticated identity over a request. It was an early predecessor to [19],
and consequently AIB has seen no deployment.
[This Internet-Draft is no longer active] Draft-ietf-sip-saml, SIP SAML Profile and

Binding (S): I-D.ietf-sip-saml [102] defines the usage of the Security Assertion Markup
Language (SAML) within SIP, and describes how to use it in conjunction with SIP
identity [19] to provide authenticated assertions about a users role or attributes.
RFC 5360, A Framework for Consent-Based Communications in the Session Initiation

Protocol (SIP) (S): RFC 5360 [101] defines several extensions to SIP, including the
Trigger-Consent and Permission-Missing header fields. These header fields, in addition
to the other procedures defined in the document, define a way to manage membership
on "SIP mailing lists" used for instant messaging or conferencing. In particular, it helps
avoid the problem of using such amplification services for the purposes of an attack on
the network, by making sure a user authorizes the addition of their address onto such a
service.
RFC 5361, A Document Format for Requesting Consent (S): I-D.ietf-sipping-consent-

format [139] defines an XML object used by the consent framework. Consent
documents are sent from SIP "mailing list servers" to users to allow them to manage
their membership on lists.
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions Event Package (S):
RFC 5362 [140] defines a SIP event package that allows a UA to learn whether consent
has been given for the addition of an address to a SIP "mailing list". It is used in
conjunction with the SIP framework for consent [101].
RFC 3329, Security Mechanism Agreement for SIP (S): RFC 3329 [72] defines a
mechanism to prevent bid-down attacks in conjunction with SIP authentication. The
mechanism has seen very limited deployment. It was defined as part of the 3gpp IMS
specification suite [109], and is needed only when there are a multiplicity of security
mechanisms deployed at a particular server. In practice, this has not been the case.
RFC 4572, Connection-Oriented Media Transport over the Transport Layer Security
(TLS) Protocol in the Session Description Protocol (SDP) (S): RFC 4572 [104] specifies
a mechanism for signaling TLS-based media streams between endpoints. It expands
the TCP-based media signaling parameters defined in [86] to include fingerprint
information for TLS streams, so that TLS can operate between end hosts using self-
signed certificates.
RFC 5027, Security Preconditions for Session Description Protocol Media Streams (S):
RFC 5027 [92] defines a precondition for use with the preconditions framework [59].
The security precondition prevents a session from being established until a security
media stream is set up.
RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication Using

Authentication and Key Agreement (S): RFC 3310 [147] defines an extension to digest
authentication to allow it to work with the credentials stored in cell phones. Though
technically it is an extension to HTTP digest, its primary application is SIP. This
extension is useful primarily to implementors of IMS.
RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication Using

Authentication and Key Agreement (AKA) Version-2 (S): RFC 4169 [148] is an
enhancement to [RFC3310] that further improves security of the authentication.
2.15 Instant Messaging, Presence and Multimedia
2.15.1 Overview
SIP provides extensions for instant messaging, presence and Multimedia.
RFC 3428, SIP Extension for Instant Messaging (S): RFC 3428 [74] defines the
MESSAGE method, used for sending an instant message without setting up a session
(sometimes called "page mode").
RFC 3856, A Presence Event Package for SIP (S): RFC 3856 [53] defines an event
package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S): RFC 3857 [54],
also known as winfo, provides a mechanism for a user agent to find out what
subscriptions are in place for a particular event package. Its primary usage is with
presence, but it can be used with any event package.
RFC 5547, A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable

File Transfer (S): RFC 5547 [107] defines a mechanism for signaling a file transfer
session with SIP.
2.16 Emergency Services
2.16.1 Overview
Emergency services here cover both emergency calling (for example, 911 in the United
States), and pre-emption services, which allow authorized individuals to gain access to
network resources in time of emergency.
RFC 4411, Extending the SIP Reason Header for Preemption Events (S): RFC 4411 [75]
defines an extension to the Reason header, allowing a UA to know that its dialog was
torn down because a higher priority session came through.
RFC 4412, Communications Resource Priority for SIP (S): RFC 4412 [76] defines a new
header field, Resource-Priority, that allows a session to get priority treatment from the
network.
3. IMS Specifications (developed by 3GPP)
3.1 Scope and release of the IMS specifications listed in this e-book
General concepts of presentation
In order to identify the correct set of IMS specs to be included in this e-book, we used
similar method to the one we used for SIP. In short we covered all key specs that any
IMS application must consider, no matter if your IMS app is written for an endpoint, core
server (e.g. CSCF) or an App server. We did not include secondary specs, which may
distract you from the important ones. For more details please see section 3.1 above.
3GPP current release (R11)
Currently the 3GPP standard group works on developing Release 11. Main items of IMS
standardization for this release are:
Study on IMS Evolution

Study on enhancements to IMS border functions for IMS Interconnection of
services
IMS Services Centralization and Continuity
IMS Service Continuity Enhancements: Service, Policy and Interactions
IMS Centralized Services
IMS aspects of Architecture for Home NodeB
Registration in Densely-populated area
Earthquake and Tsunami Warning System
Support for IMS Emergency Calls over GPRS and EPS
Study on System enhancements for the use of IMS services in local breakout
IMS System enhancements for corporate network access
USSD simulation service
network-provided location information for IMS
SMS submit and delivery without MSISDN in IMS
3.2 Why only 3GPP specs? How about 3GPP2, TISPAN, CableLabs,
Broadband Forum and OMA?
You may refer to section 2.4, which discusses the SIP/IMS standard bodies and how
they relate to each other. In short, 3GPP is the original and main standard organization
for developing the IMS standard. As discussed in section 2.4 above, the special-
environment-access standard organizations related to IMS are: 3GPP2, TISPAN,
CableLabs/PacketCable and OMA. They develop SIP/IMS specifications, which are
tailored to their environments. However, at the end of the day their goal is to incorporate
their requirements into 3GPP (by submitting periodical contributions). Therefore while it
is beneficial to know what is going on in these special environments, it is still a safe bet
to stick to 3GPP for any IMS key specification; and that is what we just did in this e-book.
3.3 IMS Main functionality
TS 21.202, Technical Specifications and Technical Reports relating to the Common IP

Multimedia Subsystem (IMS) (formulating group: SP): TS 21.202 [394] identifies the
3GPP Technical Specifications and Technical Reports specifically relating to the IP
Multimedia Subsystem (IMS), and which are common to all systems based on IMS,
regardless of access technology.
TR 22.800, IP Multimedia Subsystem (IMS) subscription and access scenarios

(formulating group: SA1): TR 22.800 [301] is an introductory stage 1 document covering
various IMS user and operator scenarios. Examples describe functionality for basic and
roaming operation, for non-3GPP access with and without roaming, for multiple
operators, for multiple terminals, etc.
TS 23.218, IP Multimedia (IM) session handling; IM call model; Stage 2 (formulating

group: CT1): TS 23.218 [302] focuses on the functional requirements of the serving
CSCF, the HSS, the MRFC and the AS during a multimedia call (session). This is the
main spec describing the famous ISC interface (between S-CSCF and the AS).
TS 23.228, IP Multimedia Subsystem (IMS); Stage 2 (formulating group: SA2): TS

23.228 [303] describes in detail the main IMS concepts and procedures, showing the
responsibilities of the associated IMS functional entities.
TS 24.228, Signaling flows for the IP multimedia call control based on Session Initiation
Protocol (SIP) and Session Description Protocol (SDP); Stage 3 (formulating group:
CT1): TS 24.228 [304] is an informative spec, which shows the SIP signaling flows and
the exact content of the SIP messages during registration, session origination, session
termination, etc., both in case of configuration hiding (THIG) and non-hiding.
TS 24.229, Internet Protocol (IP) multimedia call control protocol based on Session
Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3 (formulating
group: CT1): TS 24.229 [389] covers in detail the functionality of the UE, of the P/I/S-
CSCF, of the MGCF, BGCF, MRFC and AS. Various SIP methods, headers and
parameters are described, including SIP compression, as well as procedures associated
with the application usage of the SDP content.
TR 24.930, Signalling Flows for the Session Setup in the IM CN Subsystem based on
SIP and SDP - Stage 3 (formulating group: CT1): TR 24.930 [374] gives examples of the
session setup in the IM CN subsystem based on SIP and SDP. These signaling flows
provide detailed signaling flows, which expand on the overview information flows
provided in 3GPP TS 23.228.
3.4 Wide-scope, generic documents (e.g. service brokering)
TR 21.905, Vocabulary for 3GPP Specifications (formulating group SA1): TR 21.905

[305] is a collection of terms, definitions and abbreviations used in the 3GPP
specifications.
TS 23.002, Network architecture (formulating group: SA2): TS 23.002 [306] is a
comprehensive document showing the core network architectures for the circuit-
switched (CS), packet-switched (PS) and IP multimedia (IMS) domains. Functional
entities, reference points and interfaces are identified and described briefly. When
someone throws an interface name (AKA reference point) at you (e.g. Cx, Sh), and you
have no idea what he/she is talking about, 23.002 is the first place you want to go and
look up.
TS 32.200, Telecommunication management; Charging management; Charging

principles (formulating group: SA5): TS 32.200 [309] is a comprehensive document
covering charging for the CS (including CAMEL support), PS and IMS (both online and
offline charging) domains as well as for the MMS services. Charging is a good example
for the practical add-on value, which IMS brings to the IP telephony world.
TR 23.976, Push architecture (formulating group: SA2): TR 23.976 [310] is a wide-scope

document covering push services in PS, CS and IMS domains. For IMS, it envisions a
Push Function AS connected to a Push Initiator and interfacing to S-CSCF via the ISC
interface. The IMS service is SIP-based, using the SUBSCRIBE, MESSAGE and
NOTIFY methods.
TS 22.173, IMS Multimedia Telephony service and Supplementary Services (formulating

group SA1): TS 22.173 [358] defines the IMS Multimedia Telephony service and the
minimum set of capabilities required to secure multi-vendor and multi-operator inter-
operability for Multimedia Telephony and related Supplementary Services.
TS 24.173, IMS Multimedia Telephony Communication Service and Supplementary

Services - Stage 3 (formulating group CT1): TS 24.173 [370] provides the protocol
details for multimedia telephony communication service and associated supplementary
services in the IP Multimedia (IM) Core Network (CN) subsystem based on the
requirements from 3GPP TS 22.173 (see one item above).
TS 22.228, Service Requirements for the IP Multimedia Core Network (IM CN)
Subsystem - Stage 1 (formulating group SA1): TS 22.228 [359] defines the service
requirements from users and operators perspective for the support of IP multimedia
applications through the IMS.
TR 23.810, Study on Architecture Impacts of Service Brokering (formulating group SA2):

The objective of TR 23.810 [366] is to study if there is enhancement needed to the
current service interaction management architecture (e.g. SCIM as part of AS and
Service Broker as part of OSA SCS) in order to satisfy requirements in 3GPP TS 22.228
(see one item above).
TS 24.167, 3GPP IMS Management Object (MO) - Stage 3 (formulating group CT1): TS
24.167 [369] defines a mobile device 3GPP IMS Management Object. The management
object is compatible with OMA Device Management protocol specifications, version 1.1.2
and upwards, and is defined using the OMA DM Device Description Framework.
3.5 Non-SIP Interfaces
TS 29.229, Cx and Dx interfaces based on the Diameter protocol; Protocol details
(formulating group: CT4): TS 29.229 [311] defines extensions/modifications to the
DIAMETER protocol in support of the signaling between CSCF and HSS (directly at the
Cx reference point and through redirection by SLF at the Dx reference point).
TS 29.228, IP Multimedia (IM) Subsystem Cx and Dx interfaces; Signaling flows and

message contents (formulating group: CT4): TS 29.228 [312] addresses the signaling
flows for Cx and Dx interfaces.
TS 29.328, IP Multimedia Subsystem (IMS) Sh interface signaling flows and message

contents (formulating group: CT4): TS 29.328 [313] defines procedures for the Sh
messages and their mapping to the DIAMETER protocol in support of the signaling
between the SIP AS (or OSA SCS) and HSS (directly at the Sh reference point and
through redirection by SLF at the Dh reference point).
TS 29.329, Sh interface based on the Diameter protocol; Protocol details (formulating

group: CT4): TS 29.329 [314] defines extensions/modifications to the DIAMETER
protocol in support of the signaling between the SIP AS (or OSA SCS) and HSS (directly
at the Sh reference point and through redirection by SLF at the Dh reference point).
TS 29.332, Media Gateway Control Function (MGCF) - IM Media Gateway (IM-MGW)

Mn interface; Stage 3 (formulating group: CT4): TS 29.332 [315] defines
extensions/modifications to the MEGACO protocol ([11]) in support of the signaling
between the MGCF and IM-MGW (at the Mn reference point).
TS 23.198 Open Service Access (OSA); Stage 2 (formulating group: CT5): TS 23.198
[363] defines an architecture that enables service application developers to make use of
network functionality through open standardized interface, i.e. the OSA APIs and Parlay
X Web Services. The concepts and the functional architecture for the OSA are contained
in this document. The requirements for OSA are contained in 3GPP TS 22.127.
3.6 QoS Support (Policy and Charging Control)
TS 23.203, Policy and Charging Control Architecture (formulating group: SA2): TS

23.203 [364] specifies the overall stage 2 level functionality for Policy and Charging
Control that encompasses the following high level functions for IP-CANs (e.g. GPRS, I-
WLAN, Fixed Broadband, etc.):
- Policy control (e.g. gating control, QoS control, etc.).
- Flow Based Charging, including charging control and online credit control;
TS 23.207, End-to-End Quality of Service (QoS) concept and architecture (formulating

group: SA2): TS 23.207 [308] covers QoS for GPRS systems, end-to-end interactions
with RSVP and IMS aspects. With regard to IMS, the functionality of the PDF and of the
P-CSCF, acting as an AF, is explained. The Gq interface and the COPS based Go
interface are addressed, together with SDP mappings to QoS parameters.
TR 23.802, Architectural enhancements for end-to-end Quality of Service (QoS)

(formulating group: SA2): TR 23.802 [317] covers possible solutions to enhance the end-
to-end QoS architecture to achieve improved end-to-end QoS in the case of interworking
with IP network domains or backbone networks that provide IP QoS mechanisms. The
on-path and off-path QoS signaling alternatives are presented side by side.
Document is currently frozen.
TS 29.207, Policy control over Go Interface (formulating group: CT3): TS 29.207 [318]
covers the interactions between the GGSN and the PDF, including the associated
signaling, based on the COPS protocol.
TS 29.208, End-to-end Quality of Service (QoS) Signalling Flows (formulating group:

CT3): TS 29.208 [379] gives examples of QoS signaling flows for resource reservation to
provide end-to-end QoS. The flows are used as bases of developing QoS related
protocol descriptions for new and existing specifications.
TS 29.209, Policy Control over Gq interface (formulating group: CT3): TS 29.209 [316]
defines procedures and protocol extensions to the DIAMETER protocol for session
based policy set-up information exchange between the Policy Decision Function (PDF)
and the Application Function (AF). In IMS the role of AF is played by the P-CSCF.
TS 29.212, Policy and Charging Control over Gx reference point (formulating group:
CT3): TS 29.212 [390] provides the stage 3 specification of the Gx reference point for
the present release. The functional requirements and the stage 2 specifications of the
Gx reference point are contained in 3GPP TS 23.203. The Gx reference point lies
between the Policy and Charging Rule Function and the Policy and Charging
Enforcement Function.
TS 29.213, Policy and charging control signalling flows and Quality of Service (QoS)
parameter mapping; Stage 3 (formulating group: CT3): TS 29.213 [391] adds detailed
flows of Policy and Charging Control (PCC) over the Rx and Gx reference points and
their relationship with the bearer level signalling flows over the Gn interface.
TS 29.214, Policy and Charging Control over Rx reference point Stage 3 (formulating
group: CT3): TS 29.214 [392] provides the stage 3 specification of the Rx reference
point for the present release. The functional requirements and the stage 2 specifications
of the Rx reference point are contained in 3GPP TS 23.203. The Rx reference point lies
between the Application Function and the Policy and Charging Rule Function.
TS 29.215, Policy and Charging Control over S9 reference point Stage 3 (formulating
group: CT3): TS 29.215 [393] provides the stage 3 specification of the S9 reference
point for the present release. The functional requirements of stage 2 specification for the
S9 reference point are contained in 3GPP TS 23.203. The S9 reference point lies
between the PCRF in the home PLMN (also known as H-PCRF) and the PCRF in the
visited PLMN (also known as V-PCRF).
TR 23.813, Study on Policy solutions and enhancements (formulating group: SA2): TR

23.813 [xxx] is a study an evolved policy solution through enhancement of 3GPP policy
framework. One of the aims of the study is to ensure a policy architecture that provides
an extensible framework for easy reuse with new IP based services by identifying areas
where improvement of specifications are feasible. Gradually its content is being exported
into TS 23.203.
TR 23.854, Enhancements for Multimedia Priority Service (formulating group: SA2): TR
23.854 [xxx] defines frame work for Multi Media Priority Services (MPS). MPS will enable
National Security/Emergency Preparedness (NS/EP) users (herein called Service Users)
to make priority calls/sessions using the public networks. This service needs to be
ensured also under special conditions such as network congestion. Service Users are
the government-authorized personnel, emergency management officials and/or other
authorized users. Effective disaster response and management rely on the Service
User's ability to communicate during congestion conditions. Service Users are expected
to receive priority treatment, in support of mission critical multimedia communications.
The enhancements for MPS evaluated in this document are priority aspects of EPS
packet bearer services and priority related interworking between IMS and EPS packet
bearer services. These enhancements enable the network to support end-to-end priority
treatment for MPS call/session origination/termination, including the Non Access Stratum
(NAS) and Access Stratum (AS) signaling establishment procedures at
originating/terminating network side as well as resource allocation in the core and radio
networks for bearers. Priority treatment will be applicable to IMS based multimedia
services, priority EPS bearer services and CS Fallback.
TR 23.839, Study on Support of BBF Access Interworking (formulating group: SA2): TR

23.839 [xxx] Based on requirements documented in the stage 1 specifications, this
technical report addresses system architecture impacts to support BBF Access
Interworking. The study includes multiple phases and covers aspects such as basic
connectivity, mobility, authentication and authorisation, policy and QoS aspects, IP Flow
mobility, traffic offload, convergence etc.
3.7 Intelligent Networks Support
TS 23.278, Customized Applications for Mobile network Enhanced Logic (CAMEL) - IP

Multimedia System (IMS) interworking; Stage 2 (formulating group: CT4): TS 23.278
[319] defines interworking between the CAMEL IP Multimedia Service Switching
Function (IM-SSF) and IMS functional entities such as HSS (via the MAP-based Si
interface) or S-CSCF (via SIP-based ISC interface).
TS 29.278, customized Applications for Mobile network Enhanced Logic (CAMEL);

CAMEL Application Part (CAP) specification for IP Multimedia Subsystems (IMS)
(formulating group: CT4): TS 29.278 [320] has little to do with IMS, to date, despite its
title. The document focuses on the gsmSCF interface with the IM-SSF. When operating
in an IMS environment, the IM-SSF is supposed to treat the MRFC as a gsmSRF and to
map between CAP messages to/from gsmSCF and SIP-based messages to/from MRFC
(sent /received via an ISC interface to/from S-CSCF).
3.8 Identities, Profile and User data
TS 31.103, Characteristics of the IP Multimedia Services Identity Module (ISIM)

application (formulating group: CT6): TS 31.103 [321] defines (in support of IMS
operation) the structure and content of the Elementary Files (EFs) stored in the ISIM, as
well as the security-related procedure (e.g. authentication) performed by the ISIM.
TS 31.133, ISIM API for Java Card (formulating group: CT6): TS 31.133 [381] defines
the ISIM Application Programming Interface extending the "UICC API for Java Card
(TM)". This API allows to develop an application running together with an ISIM
application. The present document includes information applicable to network operators,
service providers, server -, ISIM - and database manufacturers.
TS 29.240, Generic User Profile (GUP); Stage 3; Network (formulating group: CT4): TS
29.240 [322] contains a normative Appendix A defining the content of the user profile
stored in the HSS.
TS 23.008, Organization of Subscriber Data (formulating group: CT4): TS 23.008 [361]

provides details concerning information to be stored in home subscriber servers (HSS),
visitor location registers, GPRS Support Nodes and Call Session Control Function
(CSCF) concerning mobile subscriber.
TR 23.816, Identification of Communication Services in IMS (formulating group SA2):

The scope of TR 23.816 [367] is to capture the results of a feasibility study in the needs
of identifying IMS communication services with and in the need of the IMS
communication service identifier.
TS 23.003, Numbering, addressing and identification (formulating group: CT4): TS

23.003 [307] is a wide-scope document concentrating information about
identities/addresses used in 3GPP family of specifications: user identities (e.g. IMSI,
TMSI), equipment identities (e.g. IMEI), areas identities (e.g., LAI, RAI), etc. For IMS, it
defines home network domains, private and public user identities and the public service
identities.
3.9 Network Interworking and Combining
TR 23.864, Commonality and interoperability between IP Multimedia System (IMS) core

networks (formulating group: SA2): TR 23.864 [323] addresses interoperability between
IMS and non-IMS IP-connectivity networks (e.g. some WLAN). Issues include the level
of QoS provided by the IP-connectivity network, (local) policy decision and enforcement
functions, user profiles.
TS 29.162, Interworking between the IM CN subsystem and IP networks (formulating

group: CT3): TS 29.162 [324] describes IMS with external IP networks interactions at the
Mb reference point. The external IP network may be IPv4-based (rather than IPv6-
based, like IMS) and may not support (all of) the extensions to SIP prescribed by IMS.
TS 29.163, Interworking between the IP Multimedia (IM) Core Network (CN) subsystem
and Circuit Switched (CS) networks (formulating group: CT3): TS 29.163 [325] is a
detailed document covering interworking between IMS and BICC/ISUP based legacy
CS, with and without 3GPP specific additions, in order to support IMS basic voice calls
and supplementary services. The document describes protocol translations between
ISUP and SIP occurring in the MGCF and MGW as well as codecs and transcoders
handling.
TR 23.981, Interworking aspects and migration scenarios for IPv4-based IP Multimedia
Subsystem (IMS) implementations (formulating group: SA2): TR 23.981 [326] addresses
issues related to IPv4 equipment interworking with IMS. The document looks into P-
CSCF discovery, roaming between IPv4 and IPv6 systems, as well as impacts on the
current GPRS systems, if connected to the IMS core.
TR 23.899, Combining Circuit Switched (CS) bearers with IP Multimedia Subsystem

(IMS) (formulating group: SA2): TR 23.899 [327] looks at using CS bearers and IMS
simultaneously (e.g. mixed circuit voice and VoIP call). The spec is mainly targeted
towards GERAN access networks. (SPEC WITHDRAWN! Here for education purpose
only)
TR 22.979, Feasibility study on combined Circuit Switched (CS) calls and IP Multimedia
Subsystem (IMS) sessions (formulating group: SA1): TR 22.979 [328] is being defined at
high level as part of Release 7, to introduce combinational services, sessions and calls
between UEs, by adding IMS components to CS calls and vice-versa.
TS 22.279, Combined CS and IMS Sessions; stage 1 (formulating group: SA1): TS

22.279 [360] specifies service requirements for Combining CS and IMS services using a
CS speech or CS multimedia call in association with an IMS session.
TS 24.279, Combining Circuit Switched (CS) and IP Multimedia Subsystem (IMS)

Services - Stage 3 (formulating group: CT1): TS 24.279 [372] provides the technical
realization for the combination of Circuit Switched calls and IM sessions when using
them simultaneously between the same two users.
3.10 Accounting/Charging
TR 23.125, Overall high level functionality and architecture impacts of flow based
charging; Stage 2 (formulating group: SA2): TR 23.125 [329] specifies the overall high
level functionality and architecture impacts of Flow Based Charging.
TR 23.815, Charging implications of IMS architecture (formulating group: SA2): TR

23.815 [330] shows architectural impacts of charging, functional entities used for
charging, user cases and requirements.
TS 32.225, Telecommunication management; Charging management; Charging data

description for the IP Multimedia Subsystem (IMS) (formulating group: SA5): TS 32.225
[331] covers online and offline charging for IMS, including formats for the CDR. The
document focuses on the extensions to the DIAMETER protocol in support of IMS
charging.
TS 32.240, Telecommunication management; Charging management; Charging

architecture and principles (formulating group: SA5): TS 32.240 [332] is a general
document (i.e. not IMS-specific) defining the architecture, interfaces and common
charging principles.
TS 32.260, Telecommunication management; Charging management; IP Multimedia

Subsystem (IMS) charging (formulating group: SA5): TS 32.260 [333] covers online and
offline charging for IMS. The document details charging principles, impacts on functional
entities, signaling and the formats of CDRs.
TS 32.296, Telecommunication management; Charging management; Online Charging

System (OCS): Applications and interfaces (formulating group: SA5): TS 32.296 [334]
covers all internal aspects of the Online Charging System (OCS). The document
contains the architecture and functions of the OCS logical components and thereby
derives the functionality of the OCS interfaces. A detailed specification of interfaces
between the logical OCS components is also included. The functionality of the OCS, as
described in the document, applies to all charging domains (bearer, session and
service).
TS 32.297, Charging Management - Charging Data Record (CDR) File Format and
Transfer (formulating group: SA5): TS 23.297 [387] specifies the mechanisms used to
transfer CDR files from the network to the operator's billing domain (e.g. the billing
system or a mediation device). This includes the file transfer procedures and the layout
of the CDR files, as well as file meta information and the encoding of the CDRs within
the files.
TS 32.298, Charging Management - Charging Data Record (CDR) Parameter

Description (formulating group: SA5): TS 23.298 [388] specifies the CDR parameters,
the abstract syntax and encoding rules for all the CDR types that are defined in the
charging management TSs described above. Therefore, it is only applicable to offline
charging.
3.11 Security
TS 33.102, 3G Security; Security Architecture (formulating group: SA3): TS 33.102 [382]

defines the security architecture, i.e., the security features and the security mechanisms,
for the third generation mobile telecommunication system.
TS 33.203, 3G Security; Access security for IP-based services (formulating group: SA3):
TS 33.203 [335] overviews the IMS security architecture and covers various features
and mechanisms involved in IMS security. The document addresses the protection of
the Gm interface (UE to P-CSCF).
TS 33.210, 3G Security; Network Domain Security; IP network layer security (formulating

group: SA3): TS 33.210 [336] covers at high level Network Domain Security aspects,
such as security gateways, the Za and Zb reference points and various aspects of IKE
and IPSEC.
TS 33.220, Generic Authentication Architecture (GAA); Generic bootstrapping

architecture (formulating group: SA3): TS 33.220 [337] covers the AKA-based generic
bootstrapping architecture. IMS is impacted at the reference point Zh between the HSS
and the bootstrapping function entity.
TS 33.222, Generic Authentication Architecture (GAA); Access to network application

functions using Hypertext Transfer Protocol over Transport Layer Security (HTTPS)
architecture (formulating group: SA3): TS 33.222 [338] specifies secure access methods
to Network Application Functions (NAF) using HTTP over TLS in the Generic
Authentication Architecture (GAA), and provides Stage 2 security requirements,
principles and procedures for the access. The document describes both direct access to
an Application Server (AS) and access to an Application Server through an
Authentication Proxy (AP).
TS 29.109, Generic Authentication Architecture (GAA); Zh and Zn interfaces based on

the Diameter protocol (formulating group: CT4): TS 29.109 [339] defines the Diameter
based implementation for bootstrapping Zh interface (BSF-HSS) and GAA Application
Zn interface (BSF-NAF) in Generic Authentication Architecture (GAA). The document
covers procedures, message contents and coding. The interface is based on reuse of
Cx interface Multimedia-Auth-Request/Answer messages between CSCF and HSS.
TR 33.978, Security aspects of early IP Multimedia Subsystem (IMS) (formulating group:

SA3): TR 33.978 [340] covers a method of providing security without support of IPsec,
through IP comparison for GSN-based systems.
TS 33.310, Network Domain Security (NDS) - Authentication Framework (AF)

(formulating group: SA3): The scope of TS 33.310 [384] is limited to authentication of
network elements, which are using NDS/IP or TLS, and located in the inter-operator
domain.
TR 33.803, Security aspects of early IP Multimedia Subsystem (IMS) (formulating group:

SA3): TR 33.803 [385] studies from a security point of view the coexistence between
TISPAN authentication methods (as specified in TISPAN release 1) and existing 3GPP
authentication schemes, i.e. both the IMS AKA (as specified in 3GPP TS 33.203 and
3GPP TS 24.229) and the early IMS security (as specified in 3GPP TR 33.978). This
document also aims to provide solutions to handle potential compatibility issues. These
issues are listed in detail in section 5 of this document.
3.12 Lawful Intercept
TS 33.107, 3G security; Lawful interception architecture and functions (formulating

group: SA3): TS 33.107 [341] shows the service requirements from a Law Enforcement
point of view. The invocation of Lawful Intercept services is addressed separately for
CS, PS and IMS domains.
TS 33.108, 3G security; Handover Interface for Lawful Interception (formulating group:

SA3): TS 33.108 [342] addresses the handover interfaces for Lawful Interception of
Packet-Data Services, Circuit Switched Services, Multimedia Services (IMS). The
handover interface includes the delivery of Intercept Related Information and Content of
Communication to the Law Enforcement Monitoring Facility.
3.13 Group Management
TS 22.250, IP Multimedia Subsystem (IMS) Group Management; Stage 1 (formulating

group: SA1): TS 22.250 [343] is a brief document covering a couple of use cases, high
level requirements and some security and charging considerations. It does not identify
any impacts on IMS functional entities, architecture, protocols or signaling.
3.14 Presence
TS 22.141, Presence service; Stage 1 (formulating group: SA1): TS 22.141 [344] defines
the requirements for the support of the presence service.
TS 23.141, Presence service; Architecture and functional description; Stage 2

(formulating group: SA2): TS 23.141 [345] describes the presence architecture,
functional entities, attributes, interfaces and reference points involved in presence.
TS 24.141, Presence service using the IP Multimedia (IM) Core Network (CN)
subsystem; Stage 3 (formulating group: CT1): TS 24.141 [346] provides the protocol
details for the presence service within the IMS, based on SIP and SIP Events.
Requirements for manipulation of presence data are defined by use of a protocol at the
Ut reference point (between AS and UE) based on XML Configuration Access Protocol
(XCAP).
TR 24.841, Presence service based on Session Initiation Protocol (SIP);

Functional models, information flows and protocol details; (formulating group: CT1): TR
24.841 [347] Illustrates many Presence service examples; the equivalent of 24.228 for
the Presence service area. Ironically the TR, which nicely illustrates the Presence
service related - SIP PUBLISH method, never got published itself. Still we find it to be a
quite useful informational document.
TS 33.141, Presence service; Security (formulating group: SA3): TS 33.141 [348] is the
Stage 2 specification for the security requirements, security architecture, security
features and security mechanisms for the Presence Service. The main content of this
specification is the security for the Ut reference point, which is HTTPbased, as applied
in presence services.
3.15 Emergency Services
TR 23.867, Internet Protocol (IP) based IP Multimedia Subsystem (IMS) emergency

sessions (formulating group: SA2): TR 23.867 [349] provides requirements and
describes architecture and signaling for emergency procedures. The document identifies
impacts on the IMS, the UE and the GPRS IP-connectivity access network.
TS 23.167, IP Multimedia Subsystem (IMS) Emergency Sessions (formulating group:

SA2): TS 23.167 [362] defines the stage-2 service description for emergency services in
the IP Multimedia Core Network Subsystem (IMS), including the elements necessary to
support IP Multimedia (IM) emergency services. This document covers also the Access
Network aspects that are crucial for the provisioning of IMS emergency services.
3.16 Conferencing
TS 24.147, Conferencing using the IP Multimedia (IM) Core Network (CN) subsystem;
Stage 3 (formulating group: CT1): TS 24.147 [350] provides protocol details for
conferencing within IMS, based on the SIP, SIP Events, SDP, the Conference Policy
Control Protocol (CPCP) and the Binary Floor Control Protocol (BFCP) protocols. The
document is applicable to AS, MRFC, MRFP, MGCF and UE, but the MRFC to MRFP
signaling is not covered.
3.17 Messaging and Push To Talk
TS 22.340, IP Multimedia Subsystem (IMS) messaging; Stage 1 (formulating group: TS

SA1): TS 22.340 [351] contains the requirements for the IMS messaging system.
TR 22.940, IP Multimedia Subsystem (IMS) messaging; Stage 1 (formulating group:

SA1): TR 22.940 [352] covers basic aspects of IMS messaging, includes user scenarios
and addresses interactions with other technologies, whether developed by 3GPP (e.g.
MMS) or other groups (e.g. OMA). Three types of IMS messages are identified:
immediate, deferred delivery and session based.
TR 23.804, Support of SMS and MMS over generic 3GPP IP access (formulating group:
SA2): TR 23.804 [353] covers registration for, and delivery of, short and multimedia
messages. SIP-based signaling in support of IMS is included.
TS 23.204, Support of SMS and MMS over generic 3GPP IP access (formulating group:
SA2): TS 23.204 [365] This TS specifies the new capabilities and enhancements needed
to support SMS over a generic IP Connectivity Access Network (IP-CAN) using IMS
capabilities.
TS 24.247, Messaging service using the IP Multimedia (IM) Core Network (CN)
subsystem Stage 3 (formulating group: CT1): TS 24.247 [354] provides protocol details
for the IMS messaging service (immediate messaging, session-based messaging and
session-based messaging conferences) using SIP, SDP, and Message Session Relay
Protocol (MSRP). The document is applicable to AS, MRFC, MRFP and UEs, but does
not cover signaling between MRFC and MRFP.
TS 24.341, Support of SMS over IP networks - Stage 3 (formulating group: CT1): TS

24.341 [373] provides the protocol details for SMS over IP within the IP Multimedia (IM)
Core Network (CN) subsystem based on the Session Initiation Protocol (SIP) and SIP
Events as defined in 3GPP TS 24.229.
TR 23.979, Push-to-talk over Cellular (PoC) Services - Stage 2 (formulating group: SA2):
TR 23.979 [368] studies the architectural requirements in order to enable services like
PoC over 3GPP systems. The report looks into aspects of using 3GPP PS domain and
radio access technologies (GERAN, UTRAN) for bearer services and IMS for
reachability and connectivity for applications like PoC.
TS 32.272, Push-to-talk over Cellular (PoC) Charging (formulating group: SA5): TS

32.272 [386] specifies the Offline and Online Charging description for the "Push-to-Talk
over Cellular" (PoC) service, based on the functional description of the PoC service in
3GPP TR 23.979 "3GPP enablers for OMA PoC Services", in OMA-AD-POC "Push to
talk over Cellular (PoC) - Architecture", in OMA-CP-POC "OMA PoC Control Plane" and
in OMA-UP-POC: "OMA POC User Plane", respectively. This charging description
includes the offline and online charging architecture and scenarios specific to PoC, as
well as the mapping of the common 3GPP charging architecture specified in 3GPP TS
32.240 onto the PoC service. It further specifies the structure and content of the CDRs
for offline charging, and the charging events for online charging.
3.18 I-WLAN Interworking with IMS
TS 23.234, 3GPP system to Wireless Local Area Network (WLAN) interworking; system
description (formulating group: SA2): TS 23.234 [355] specifies the architecture,
functionality and interfaces necessary for connecting a WLAN access network to a
3GPP system.
TS 29.234, 3GPP System to WLAN Interworking - Stage 3 (formulating group: CT4): TS

29.234 [380] defines the stage-3 protocol description for several reference points in the
WLAN-3GPP Interworking System. The present document is applicable to:
- The Dw reference point between the 3GPP AAA Server and an SLF.
- The Wa reference point between the WLAN AN and the 3GPP AAA Proxy.
- The Wd reference point between the 3GPP AAA Proxy and 3GPP AAA Server.
- The Wx reference point between the 3GPP AAA Server and the HSS.
- The Wm reference point between the 3GPP AAA Server and the PDG.
- The Wg reference point between the 3GPP AAA Server/Proxy and the WAG.
- The Pr reference point between the 3GPP AAA Server and the PNA.
TS 33.234, WLAN Interworking Security (formulating group: SA3): TS 33.234 [383]

specifies the security architecture; trust model and security requirements for the
interworking of the 3GPP System and WLAN Access Networks. Specifications of the
appropriate mechanisms for user and network authentication, key management, service
authorization, confidentiality and integrity protection of user and signaling data are also
provided.
TR 23.836, Quality of Service (QoS) and policy aspects of 3GPP IP Wireless Local
Area Network (WLAN) interworking (formulating group: SA2): TR 23.836 [356] identifies
requirements, functions and procedures for the WLAN AN, PDG and WAG in support of
QoS.
3.19 Seamless Mobility (AKA Voice Call Continuity (VCC))
TS 23.206, Voice Call Continuity (VCC) between Circuit Switched (CS)

and IP Multimedia Subsystem (IMS); (formulating group: SA2): TS 23.206 [357] specifies
the architecture, functionality and interfaces necessary for performing handoff (real
time/seamless transfer) of voice calls between the Circuit Switch (CS) domain and the
IMS domain (and vice versa).
TS 24.206, Voice Call Continuity between the Circuit-Switched (CS) domain and the IP
Multimedia Core Network (CN) (IMS) subsystem - Stage 3; (formulating group: CT1): TS
24.206 [371] provides the protocol details for voice call continuity between the IP
Multimedia (IM) Core Network (CN) subsystem based on the Session Initiation Protocol
(SIP) and the Session Description Protocol (SDP) and the protocols of the 3GPP Circuit-
Switched (CS) domain (CAP, MAP, ISUP, BICC and the NAS call control protocol for the
CS access).
TS 23.216, Single Radio Voice Call Continuity (SRVCC); - Stage 2; (formulating group:
SA2): TS 23.216 [405] specifies the architecture enhancements for Single Radio Voice
Call Continuity (SRVCC) between the following access systems for voice calls that are
anchored in the IMS:
- from E UTRAN to 3GPP2 1xCS;
- from E UTRAN to UTRAN/GERAN;
- from UTRAN (HSPA) to UTRAN/GERAN.
3.20 Media handling and characteristics (codecs)
TS 26.114, IMS - Multimedia Telephony - Media Handling and Interaction (formulating

group: SA4): TS 26.114 [375] specifies a client for the Multimedia Telephony Service for
IMS (MTSI) supporting conversational speech (including DTMF), video and text
transported over RTP with the scope to deliver a user experience equivalent to or better
than that of Circuit Switched (CS) conversational services using the same amount of
network resources. It defines media handling (e.g. signaling, transport, jitter buffer
management, packet-loss handling, adaptation), as well as interactivity (e.g. adding or
dropping media during a call).
TS 26.141, IP Multimedia System (IMS) Messaging and Presence - Media Formats and
Codecs (formulating group: SA4): TS 26.141 [376] specifies the basic media formats and
codecs to be used in the IMS Messaging and Presence services. It defines the
mandatory 'baseline' set of media types for the services. Additionally, it also targets to
allow possible message content type enhancements, either 3GPP-standardized or other
generally used media types, in a flexible way.
TS 26.235, Packet Switched Conversational Multimedia Applications - Default Codecs

(formulating group: SA4): TS 26.235 [377] introduces the set of default codecs for packet
switched conversational multimedia applications within 3GPP IP Multimedia Subsystem.
Visual and sound communications are specifically addressed. The intended applications
are assumed to require low-delay, real-time functionality.
TS 26.236, Packet Switched Conversational Multimedia Applications - Transport

Protocols (formulating group: SA4): TS 26.236 [378] introduces the required protocols
for packet switched conversational multimedia applications within 3GPP IP Multimedia
Subsystem. Visual and sound communications are specifically addressed. The intended
applications are assumed to require low-delay, real-time functionality.
3.21 End Point requirements
TS 21.111, USIM and IC card requirements (formulating group: CT6): TS 21.111 [395]
defines the requirements of the USIM (Universal Subscriber Identity Module) and the IC
card for 3G (UICC). The USIM is a 3G application on an IC card. It inter-operates with a
3G terminal and provides access to 3G services. This document is intended to serve as
a basis for the detailed specification of the USIM and the UICC, and the interface to the
3G terminal.
TS 22.030, Man-Machine Interface (MMI) of the User Equipment (UE) (formulating

group: SA1): TS 22.030 [396] defines the requirements for and gives guidelines on the
MMI on the User Equipment (UE). This includes the requirements of the user procedures
for call control and supplementary service control and the requirements on the physical
input media and the output, such as indications and displayed information.
TS 31.101, UICC-terminal interface; Physical and logical characteristics (formulating

group: CT6): TS 31.101 [397] specifies the interface between the UICC and the Terminal
for 3G telecom network operation.
It specifies:
- the requirements for the physical characteristics of the UICC;
- the electrical interface between the UICC and the Terminal;
- the initial communication establishment and the transport protocols;
- the model which serves as a basis for the logical structure of the UICC;
- the communication commands and the procedures;
- the application independent files and protocols.
The administrative procedures and initial card management are not part of the TS.
TS 31.103, Characteristics of the IP Multimedia Services Identity Module (ISIM)

application (formulating group: CT6): TS 31.103 [398] defines the IM Services Identity
Module (ISIM) application. This application resides on the UICC, an IC card specified in
TS 31.101 [397]. In particular, TS 31.101 [397] specifies the application independent
properties of the UICC/terminal interface such as the physical characteristics and the
logical structure.
TS 31.115, Secured packet structure for (Universal) Subscriber Identity Module (U)SIM
Toolkit applications (formulating group: CT6): TS 31.115 [399] is the result of a split of
TS 23.048 Release 5 between the generic part and the bearers specific application. The
generic part has been transferred to SCP. The present document is the bearers specific
part. It specifies the structure of the Secured Packets in implementations using Short
Message Service Point to Point (SMS-PP), Short Message Service Cell Broadcast
(SMS-CB), Unstructured Supplementary Service Data (USSD) and and Hyper Text
Transfer Protocol (HTTP) based on ETSI TS 102 225.
TS 31.116, Remote APDU Structure for (Universal) Subscriber Identity Module (U)SIM
Toolkit applications (formulating group: CT6): TS 31.116 [400] defines the remote
management of files and applets on the SIM/USIM/ISIM. It describes the APDU format
for remote management.
TS 31.133, IP Multimedia Services Identity Module (ISIM) Application Programming

Interface (API); ISIM API for Java Card (formulating group: CT6): TS 31.133 [401]
defines the ISIM Application Programming Interface extending the "UICC API for Java
Card".
This API allows to develop an application running together with a ISIM application.
The TS includes information applicable to network operators, service providers, server,
,ISIM and database manufactures.
3.22 Location
3GPP TR 23.842, Study on Network Provided Location Information to the IMS

(formulating group: SA2): TS 23.842 [406] investigates architecture solutions for making
the Network Provided Location Information (NPLI) related to the access network that the
UE is camped on available to the IMS nodes whenever the IMS operator needs to record
this information either to fulfill legal obligations, for charging or for other purposes.
3.23 Access Network Aspects (3GPP (LTE, UTRAN, GERAN) and

non-3GPP (3GPP2, WiFi, Wireline, Femto))
3GPP TR 23.861, Multi access PDN connectivity and IP flow mobility (formulating group:
SA2): TR 23.861 [407] study on the scenarios, requirements and solutions for UEs with
multiple interfaces which will simultaneously connect to 3GPP access and one, and only
one, non-3GPP access.
3GPP TS 23.401, General Packet Radio Service (GPRS) enhancements for Evolved
Universal Terrestrial Radio Access Network (E-UTRAN) access (formulating group:
SA2): TS 23.401 [408] defines the Stage 2 service description for the Evolved 3GPP
Packet Switched Domain - also known as the Evolved Packet System (EPS) in this
document. The Evolved 3GPP Packet Switched Domain provides IP connectivity using
the Evolved Universal Terrestrial Radio Access Network (E-UTRAN).
3GPP TS 23.402, Architecture enhancements for non-3GPP accesses (formulating

group: SA2): TS 23.402 [409] specifies the stage 2 service description for providing IP
connectivity using non-3GPP accesses to the Evolved 3GPP Packet Switched domain.
In addition, for E-UTRAN (LTE) and non-3GPP accesses, the specification describes the
Evolved 3GPP PS Domain where the protocols between its Core Network elements are
IETF-based (mostly variants of IPSEC and MIP).
3GPP TS 23.261, IP flow mobility and seamless Wireless Local Area Network (WLAN)
offload; Stage 2 (formulating group: SA2): TS 23.402 [410] specifies the Stage 2 system
description for IP flow mobility between a 3GPP and a WLAN. The technical solution is
based on the working principles of DSMIPv6 and it is applicable to both the Evolved
Packet System and the I-WLAN mobility architecture.
4. References
4.1 IETF References
[1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks,
R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[2] Bradner, S., "The Internet Standards Process -- Revision 3", BCP 9, RFC 2026,
October 1996.
[3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport
Protocol for Real-Time Applications", RFC 3550, July 2003.
[4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session

Description Protocol (SDP)", RFC 3264, June 2002.
[5] Rosenberg, J., "Interactive Connectivity Establishment (ICE):

A Methodology for Network Address Translator (NAT) Traversal
for Offer/Answer Protocols", draft-ietf-mmusic-ice-19 (work in progress), Oct
2007.
[6] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu, Z., and J. Rosenberg,
"Signaling Compression (SigComp)", RFC 3320, January 2003.
[7] Peterson, J., "Session Initiation Protocol (SIP) Authenticated

Identity Body (AIB) Format", RFC 3893, September 2004.
[8] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., and B.
Rosen, "Change Process for the Session Initiation Protocol
(SIP)", BCP 67, RFC 3427, December 2002.
[9] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,

"SIP: Session Initiation Protocol", RFC 2543, March 1999.
[10] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol

(SIP): Locating SIP Servers", RFC 3263, June 2002.
[11] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[12] Mealling, M. and R. Daniel, "The Naming Authority Pointer

(NAPTR) DNS Resource Record", RFC 2915, September 2000.
[13] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[14] Peterson, J., "A Privacy Mechanism for the Session Initiation
Protocol (SIP)", RFC 3323, November 2002.
[15] Jennings, C., Peterson, J., and M. Watson, "Private Extensions

to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks", RFC 3325, November 2002.
[16] Willis, D. and B. Hoeneisen, "Session Initiation Protocol

(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
[17] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session

Initiation Protocol (SIP) for Symmetric Response Routing",
RFC 3581, August 2003.
[18] Sparks, R., "Actions Addressing Identified Issues with the

Session Initiation Protocol's (SIP) Non-INVITE Transaction",
RFC 4320, January 2006.
[19] Peterson, J. and C. Jennings, "Enhancements for Authenticated

Identity Management in the Session Initiation Protocol (SIP)",
[20] Rosenberg, J., "Obtaining and Using Globally Routable User

Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", RFC 5627, Oct 2009.
[21] Jennings, C. and R. Mahy, "Managing Client Initiated

Connections in the Session Initiation Protocol (SIP)",
RFC 5626, Oct 2009.
[22] Petrack, S. and L. Conroy, "The PINT Service Protocol:

Extensions to SIP and SDP for IP Access to Telephone Call
Services", RFC 2848, June 2000.
[23] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato, J., Lu, H.,
and M. Unmehopa, "The SPIRITS (Services in PSTN requesting
Internet Services) Protocol", RFC 3910, October 2004.
[24] Vemuri, A. and J. Peterson, "Session Initiation Protocol for

Telephones (SIP-T): Context and Architectures", BCP 63,
RFC 3372, September 2002.
[25] Camarillo, G., Roach, A., Peterson, J., and L. Ong,

"Integrated Services Digital Network (ISDN) User Part (ISUP)
to Session Initiation Protocol (SIP) Mapping", RFC 3398,
December 2002.
[26] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Mapping
of Integrated Services Digital Network (ISDN) User Part (ISUP)
Overlap Signalling to the Session Initiation Protocol (SIP)",
[27] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing

Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[28] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional

Responses in Session Initiation Protocol (SIP)", RFC 3262,
June 2002.
[29] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE

Method", RFC 3311, October 2002.
[30] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
[31] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason

Header Field for the Session Initiation Protocol (SIP)",
[32] Willis, D. and B. Hoeneisen, "Session Initiation Protocol

(SIP) Extension Header Field for Service Route Discovery
During Registration", RFC 3608, October 2003.
[33] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating

User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[34] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller

Preferences for the Session Initiation Protocol (SIP)",
[35] Donovan, S. and J. Rosenberg, "Session Timers in the Session

Initiation Protocol (SIP)", RFC 4028, April 2005.
[36] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The Stream

Control Transmission Protocol (SCTP) as a Transport for the
Session Initiation Protocol (SIP)", RFC 4168, October 2005.
[37] Barnes, M., "An Extension to the Session Initiation Protocol

(SIP) for Request History Information", RFC 4244, November 2005.
[38] Levin, O., "Suppression of Session Initiation Protocol (SIP)

REFER Method Implicit Subscription", RFC 4488, May 2006.
[39] Rosenberg, J., "Request Authorization through Dialog

Identification in the Session Initiation Protocol (SIP)",
RFC 4538, June 2006.
[40] Levin, O. and A. Johnston, "Conveying Feature Tags with the
Session Initiation Protocol (SIP) REFER Method", RFC 4508, May 2006.
[41] Willis, D. and A. Allen, "Requesting Answering Modes for the Session Initiation
Protocol (SIP)", RFC 5373, November 2008.
[42] Rosenberg, J., "A Hitchhikers Guide to the Session Initiation Protocol (SIP),
RFC 5411, Jan 2009.
[43] Rosenberg, J., "Rejecting Anonymous Requests in the Session

Initiation Protocol (SIP)", RFC 5079, Dec 2007.
[44] Camarillo, G., Niemi, A., Isomaki, M., Garcia-Martin, M., and H. Khartabil,
"Referring to Multiple Resources in the Session Initiation Protocol (SIP)", RFC
5368, October 2008.
[45] Sparks, R., "The Session Initiation Protocol (SIP) Refer

Method", RFC 3515, April 2003.
[46] Rosenberg, J., Peterson, J., Schulzrinne, H., and G.

Camarillo, "Best Current Practices for Third Party Call
Control (3pcc) in the Session Initiation Protocol (SIP)",
BCP 85, RFC 3725, April 2004.
[47] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.
[48] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By

Mechanism", RFC 3892, September 2004.
[49] Mahy, R. and D. Petrie, "The Session Initiation Protocol (SIP)

"Join" Header", RFC 3911, October 2004.
[50] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk,
"Transcoding Services Invocation in the Session Initiation
Protocol (SIP) Using Third Party Call Control (3pcc)",
RFC 4117, June 2005.
[51] Niemi, A., "Session Initiation Protocol (SIP) Extension for

Event State Publication", RFC 3903, October 2004.
[52] Rosenberg, J., "A Session Initiation Protocol (SIP) Event

Package for Registrations", RFC 3680, March 2004.
[53] Rosenberg, J., "A Presence Event Package for the Session
Initiation Protocol (SIP)", RFC 3856, August 2004.
[54] Rosenberg, J., "A Watcher Information Event Template-Package

for the Session Initiation Protocol (SIP)", RFC 3857, August 2004.
[55] Santesson, S. and R. Housley, "Internet X.509 Public Key
Infrastructure Authority Information Access Certificate
Revocation List (CRL) Extension", RFC 4325, December 2005.
[56] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session

Initiation Protocol (SIP) Event Package for Conference State",
[57] Burger, E. and M. Dolly, "A Session Initiation Protocol (SIP)

Event Package for Key Press Stimulus (KPML)",
RFC 4730, November 2006.
[58] Pendleton, A., "Session Initiation Protocol Package for Voice

Quality Reporting Event", draft-ietf-sipping-rtcp-summary-02
(work in progress), May 2007.
[59] Camarillo, G., Marshall, W., and J. Rosenberg, "Integration of

Resource Management and Session Initiation Protocol (SIP)",
RFC 3312, October 2002.
[60] Camarillo, G. and P. Kyzivat, "Update to the Session

Initiation Protocol (SIP) Preconditions Framework", RFC 4032,
March 2005.
[61] Marshall, W., "Private Session Initiation Protocol (SIP)

Extensions for Media Authorization", RFC 3313, January 2003.
[62] Petrie, D. and S. Channabasappa, "A Framework for Session Initiation Protocol
User Agent Profile Delivery", draft-ietf-sipping-config-framework-12 (work in
progress), June 2007.
[63] Petrie, D., "Extensions to the Session Initiation Protocol

(SIP) User Agent Profile Delivery Change Notification Event
Package for the Extensible Markup Language Language
Configuration Access Protocol (XCAP)",
draft-ietf-sip-xcap-config-00 (work in progress),
October 2006.
[64] Camarillo, G., "Compressing the Session Initiation Protocol

(SIP)", RFC 3486, February 2003.
[65] Foster, M., McGarry, T., and J. Yu, "Number Portability in the
Global Switched Telephone Network (GSTN): An Overview",
RFC 3482, February 2003.
[66] Campbell, B. and R. Sparks, "Control of Service Context using

SIP Request-URI", RFC 3087, April 2001.
[67] Roach, A., Campbell, B., and J. Rosenberg, "A Session
Initiation Protocol (SIP) Event Notification Extension for
Resource Lists", RFC 4662, August 2006.
[68] Camarillo, G., "Subscriptions to Request-Contained Resource

Lists in the Session Initiation Protocol (SIP)",
draft-ietf-sip-uri-list-subscribe-01 (work in progress),
January 2007.
[69] Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient

MESSAGE Requests in the Session Initiation Protocol (SIP)",
draft-ietf-sip-uri-list-message-01 (work in progress),
January 2007.
[70] Camarillo, G. and A. Johnston, "Conference Establishment Using Request-

Contained Lists in the Session Initiation Protocol (SIP)", RFC 5366,
October 2008.
[71] Peterson, J., "S/MIME Advanced Encryption Standard (AES)

Requirement for the Session Initiation Protocol (SIP)",
RFC 3853, July 2004.
[72] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T.
Haukka, "Security Mechanism Agreement for the Session
Initiation Protocol (SIP)", RFC 3329, January 2003.
[73] Ono, K. and S. Tachimoto, "End-to-middle Security in the

Session Initiation Protocol (SIP)", draft-ietf-sip-e2m-sec-05
(work in progress), March 2007.
[74] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and
D. Gurle, "Session Initiation Protocol (SIP) Extension for
Instant Messaging", RFC 3428, December 2002.
[75] Polk, J., "Extending the Session Initiation Protocol (SIP)

Reason Header for Preemption Events", RFC 4411, February 2006.
[76] Schulzrinne, H. and J. Polk, "Communications Resource Priority

for the Session Initiation Protocol (SIP)", RFC 4412,
February 2006.
[77] Rosenberg, J., "A Framework for Application Interaction in the

Session Initiation Protocol (SIP)",
draft-ietf-sipping-app-interaction-framework-05 (work in progress), July 2005.
[78] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session

Description Protocol", RFC 4566, July 2006.
[79] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,

"Grouping of Media Lines in the Session Description Protocol
(SDP)", RFC 3388, December 2002.
[80] Huitema, C., "Real Time Control Protocol (RTCP) attribute in
Session Description Protocol (SDP)", RFC 3605, October 2003.
[81] Elwell, J., "Connected Identity in the Session Initiation

Protocol (SIP)", RFC 4916, June 2007.
[82] Sparks, R., "Addressing an Amplification Vulnerability in

Session Initiation Protocol (SIP) Forking Proxies",
draft-ietf-sip-fork-loop-fix-05 (work in progress),
March 2007.
[83] Camarillo, G., "The Early Session Disposition Type for the
Session Initiation Protocol (SIP)", RFC 3959, December 2004.
[84] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F.,
Watson, M., and M. Zonoun, "MIME media types for ISUP and QSIG
Objects", RFC 3204, December 2001.
[85] Sparks, R., "Internet Media Type message/sipfrag", RFC 3420,

November 2002.
[86] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the

Session Description Protocol (SDP)", RFC 4145, September 2005.
[87] Camarillo, G. and J. Rosenberg, "The Alternative Network

Address Types (ANAT) Semantics for the Session Description
Protocol (SDP) Grouping Framework", RFC 4091, June 2005.
[88] Rosenberg, J., "TCP Candidates with Interactive Connectivity

Establishment (ICE)", draft-ietf-mmusic-ice-tcp-11 (work in
progress), Nov 2010.
[89] Burger, E., "A Mechanism for Content Indirection in Session

Initiation Protocol (SIP) Messages", RFC 4483, May 2006.
[90] Westerlund, M., "A Transport Independent Bandwidth Modifier

for the Session Description Protocol (SDP)", RFC 3890,
September 2004.
[91] Camarillo, G., "Session Description Protocol (SDP) Format for

Binary Floor Control Protocol (BFCP) Streams",
RFC 4583, November 2006.
[92] Andreasen, F. and D. Wing, "Security Preconditions for Session

Description Protocol (SDP) Media Streams",
[93] Andreasen, F., RFC 5898 "Connectivity Preconditions for Session Description
Protocol (SDP) Media Streams", July 2010.
[94] Hautakorpi, J. and G. Camarillo, "The SDP (Session Description
Protocol) Content Attribute", RFC 4796, February 2007.
[95] Levin, O. and G. Camarillo, "The Session Description Protocol

(SDP) Label Attribute", RFC 4574, August 2006.
[96] Hilt, V. and G. Camarillo, "A Session Initiation Protocol

(SIP) Event Package for Session-Specific Session Policies",
draft-ietf-sipping-policy-package (work in progress),
Mar 2010.
[97] Camarillo, G. and A. Monrad, "Mapping of Media Streams to

Resource Reservation Flows", RFC 3524, April 2003.
[98] Lawrence, S., "Diagnostic Responses for Session Initiation

Protocol Hop Limit Errors",
draft-ietf-sip-hop-limit-diagnostics-03 (work in progress), June 2006.
[99] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network Media
Services with SIP", RFC 4240, December 2005.
[100] Jennings, C., "Certificate Management Service for The Session

Initiation Protocol (SIP)", draft-ietf-sip-certs (work in progress), Sep 2010.
[101] Rosenberg, J., Camarillo, G., and D. Willis, "A Framework for Consent-Based
Communications in the Session Initiation Protocol (SIP)", RFC 5360,
October 2008.
[102] Tschofenig, H., "SIP SAML Profile and Binding", draft-ietf-sip-saml (work in
progress), Mar 2010.
[103] Camarillo, G., "The Session Initiation Protocol (SIP) Pending

Additions Event Package", draft-ietf-sipping-pending-additions-02 (work in
progress), April 2007.
[104] Lennox, J., "Connection-Oriented Media Transport over the

Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006
[105] Andreasen, F., "SDP Capability Negotiation",

RFC 5939, Sep 2010.
[106] F. Andreasen, Session Description Protocol (SDP) Simple Capability

Declaration, RFC 3407, OCT 2002.
[107] Garcia-Martin, M., "A Session Description Protocol (SDP)

Offer/Answer Mechanism to Enable File Transfer", RFC 5547, May 2009.
[108] 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.
[109] 3GPP, "Internet Protocol (IP) multimedia call control protocol

based on Session Initiation Protocol (SIP) and Session
Description Protocol (SDP); Stage 3", 3GPP TS 24.229 5.19.0,
June 2007.
[110] Froment, T. and C. Lebel, "Addressing Record-Route issues in

the Session Initiation Protocol (SIP)",
draft-ietf-sip-record-route-fix-00 (work in progress),
July 2007.
[111] Niemi, A., "An Extension to Session Initiation Protocol (SIP)

Events for Conditional Event Notification",
Draft-ietf-sip-subnot-etags-04 (work in progress), Jan 2010.
[112] Audet, F., "The use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630, OCT 2009.
[113] Johnston, A. et al., "Session Initiation Protocol (SIP) Basic Call Flow Examples",
RFC 3665, Dec 2003.
[114] Elwell, J., Derks, F., Mourot, P., and O. Rousseau,

"Interworking between the Session Initiation Protocol
(SIP) and QSIG", BCP 117, RFC 4497, May 2006.
[115] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and
K. Summers, "Session Initiation Protocol (SIP) Public
Switched Telephone Network (PSTN) Call Flows", BCP 76,
[116] Gurbani, V., Carpenter, B., and B. Tate, "Essential

correction for IPv6 ABNF and URI comparison in RFC3261",
draft-ietf-sip-ipv6-abnf-fix-00 (work in progress),
February 2008.
[117] Elwell, J., "Updates to Asserted Identity in the Session Initiation Protocol (SIP)",
draft-ietf-sipping-update-pai-00 (work in progress), February 2008.
[118] Munakata, M., Schubert, S., and T. Ohba, "UA-Driven Privacy Mechanism for
SIP", draft-ietf-sip-ua-privacy-00 (work in progress), November 2007.
[119] Message Body Handling in the Session Initiation Protocol (SIP), RFC 5621, Sep
2009.
[120] Kyzivat, P., "Registration Event Package

Extension for Session Initiation Protocol (SIP) Globally Routable User
Agent URIs (GRUUs)", draft-ietf-sipping-gruu-reg-event-09 (work in progress),
July 2007.
[121] Hilt, V., "A Framework for Session

Initiation Protocol (SIP) Session Policies", draft-ietf-sip-session-policy-framework-
02 (work in progress), Feb 2010.
[122] Polk, J., Dhesikan, S., and G. Camarillo,

Quality of Service (QoS) Mechanism Selection in the Session Description
Protocol (SDP)", draft-ietf-mmusic-qos-identification-01 (work in
progress), January 2008.
[123] Channabasappa, S., "A Framework for

Session Initiation Protocol User Agent Profile Delivery", draft-ietf-sipping-config-
framework-18 (work in progress), OCT 2010.
[124] Bormann, C., Liu, Z., Price, R., and G. Camarillo, "Applying Signaling
Compression (SigComp) to the Session Initiation Protocol (SIP)", RFC 5049,
December 2007.
[125] Garcia-Martin, M. and G. Camarillo, "Multiple-

Recipient MESSAGE Requests in the Session Initiation Protocol (SIP)", draft-ietf-
sip-uri-list-message-03 (work in progress), December 2007.
[126] Jennings, C., Audet, F., and J. Elwell, "Session Initiation Protocol (SIP)
URIs for Applications such as Voicemail and Interactive Voice Response (IVR)",
RFC 4458, April 2006.
[127] Camarillo, G., "IPv6 Transition in the Session

Initiation Protocol (SIP)", draft-ietf-sipping-v6-transition-07 (work in progress),
August 2007.
[128] Mahy, R., Gurbani, V., and B. Tate, "Connection Reuse in the Session Initiation
Protocol (SIP)", draft-ietf-sip-connect-reuse-09 (work in progress), February
2008.
[129] Gurbani, V., Lawrence, S., and B. Laboratories,

"Domain Certificates in the Session Initiation Protocol (SIP)", draft-ietf-sip-
domain-certs-00 (work in progress), November 2007.
[130] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. Carrara, "Key
Management Extensions for Session Description Protocol (SDP)
and Real Time Streaming Protocol (RTSP)", RFC 4567, July 2006.
[131] Andreasen, F., Baugher, M., and D. Wing, "Session

Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568,
July 2006.
[132] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing an
SRTP Security Context using DTLS", draft-ietf-sip-dtls-srtp-framework-00 (work
in progress), November 2007.
[133] Fischl, J. and H. Tschofenig, "Session Description Protocol (SDP) Indicators for
Datagram Transport Layer Security (DTLS)", draft-ietf-mmusic-sdp-dtls-00 (work
In progress), January 2008.
[134] Camarillo, G., "A Document Format for Requesting Consent", draft-ietf-sipping-
consent-format-05 (work in progress), November 2007.
[135] Johnston, A. and O. Levin, "Session Initiation Protocol (SIP) Call Control
Conferencing for User Agents", BCP 119, RFC 4579, August 2006.
[136] Polk, J. and B. Rosen, "Location Conveyance for the Session Initiation Protocol",
draft-ietf-sip-location-conveyance-09 (work in progress), November 2007.
[137] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF Digits, Telephony Tones,
and Telephony Signals", RFC 4733, December 2006.
[138] Rosenberg, J., Camarillo, G., and D. Willis, "A Framework for Consent-Based
Communications in the Session Initiation Protocol (SIP)", RFC 5360,
October 2008.
[139] Camarillo, G., "A Document Format for Requesting Consent", RFC 5361,
October 2008.
[140] Camarillo, G., "The Session Initiation Protocol (SIP) Pending Additions Event
Package", RFC 5362, October 2008.
[141] Clark, A., Pendleton, A., Johnston, A., and H. Sinnreich, "Session Initiation
Protocol Package for Voice Quality Reporting Event", Work in Progress,
October 2008.
[142] Camarillo, G. and A. Roach, "Framework and Security

Considerations for Session Initiation Protocol (SIP) URI-List Services", RFC
5363, October 2008.
[143] Camarillo, G., Roach, A., and O. Levin,"Subscriptions to Request-

Contained Resource Lists in the Session Initiation Protocol (SIP)",
[144] Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient MESSAGE Requests in

the Session Initiation Protocol (SIP)", RFC 5365, October 2008.
[145] Camarillo, G. and A. Johnston, "Conference Establishment Using Request-

Contained Lists in the Session Initiation Protocol (SIP)", RFC 5366,
October 2008.
[146] Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol
(SIP)", Work in Progress, November 2008.
[147] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext Transfer Protocol (HTTP) Digest
Authentication Using Authentication and Key Agreement (AKA)",
RFC 3310, September 2002.
[148] Torvinen, V., Arkko, J., and M. Naslund, "Hypertext Transfer Protocol (HTTP)
Digest Authentication Using Authentication and Key Agreement (AKA)
Version-2", RFC 4169, November 2005.
[149] Session Initiation Protocol Event Package for Voice Quality Reporting
draft-ietf-sipping-rtcp-summary-10, Sep 2010.
[150] J. Rosenberg, Interactive Connectivity Establishment (ICE): A Protocol for

Network Address Translator (NAT) Traversal for Offer/Answer Protocols, Apr
2010.
[151] Simple SIP Usage Scenario for Applications in the Endpoints, Sep 2009.
4.2 3GPP References
[301] 3GPP TR 22.800,IP Multimedia Subsystem (IMS) subscription and access

scenarios
[302] 3GPP TS 23.218, IP Multimedia (IM) session handling;

IM call model; Stage 2
[303] 3GPP TS 23.228, IP Multimedia Subsystem (IMS); Stage 2
[304] 3GPP TS 24.228, Signalling flows for the IP multimedia call control based on
Session Initiation Protocol (SIP) and Session Description Protocol (SDP), Stage 3
[305] 3GPP TR 21.905, Vocabulary for 3GPP Specifications
[306] 3GPP TS 23.002, Network architecture
[307] 3GPP TS 23.003, Numbering, addressing and identification
[308] 3GPP TS 23.207, End-to-end Quality of Service (QoS) concept and architecture
[309] 3GPP TS 32.200, Telecommunication management; Charging management;

Charging principles
[310] 3GPP TR 23.976, Push architecture
[311] 3GPP TS 29.229, Cx and Dx interfaces based on the Diameter protocol; Protocol
details
[312] 3GPP TS 29.228, IP Multimedia (IM) Subsystem Cx and Dx interfaces; Signalling

flows and message contents
[313] 3GPP TS 29.328, IP Multimedia (IM) Subsystem Sh interface;
Signalling flows and message contents
[314] 3GPP TS 29.329, Sh Interface based on the Diameter protocol;

Protocol details
[315] 3GPP TS 29.332, Media Gateway Control Function (MGCF) IM Media Gateway;
Mn Interface
[316] 3GPP TS 29.209, Policy control over Gq interface
[317] 3GPP TR 23.802, Architectural enhancements for end-to-end Quality of Service

(QoS)
[318] 3GPP TS 29.207, Policy control over Go interface
[319] 3GPP TS 23.278, Customised Applications for Mobile network Enhanced Logic
(CAMEL) Phase 4; Stage 2; IM CN Interworking
[320] 3GPP TS 29.278, Customised Applications for Mobile network Enhanced Logic
(CAMEL) Phase 4; CAMEL Application Part (CAP) specification for IP Multimedia
Subsystems (IMS)
[321] 3GPP TS 31.103, Characteristics of the IP Multimedia Services Identity Module

(ISIM) application
[322] 3GPP TS 29.240, 3GPP Generic User Profile (GUP);

Stage 3; Network
[323] 3GPP TR 23.864, Commonality and Interoperability between IMSs, (Release 6)

(since 2007-08 RELEASE WITHDRAWN)
[324] 3GPP TS 29.162, Interworking between the IM CN subsystem and IP networks
[325] 3GPP TS 29.163, Interworking between the IP Multimedia (IM) Core Network
(CN) subsystem and Circuit Switched (CS) networks
[326] 3GPP TR 23.981, Interworking aspects and migration scenarios for IPv4 based
IMS Implementations
[327] 3GPP TR 23.899, Report on alternative architectures for combining CS Bearers

with IMS, (Release 6)(RELEASE WITHDRAWN)
[328] 3GPP TR 22.979, Feasibility study on combined Circuit Switched (CS) calls and
IP Multimedia Subsystem (IMS) sessions
[329] 3GPP TS 23.125, Overall high level functionality and architecture impacts of flow
based charging; Stage 2
[330] 3GPP TR 23.815, Charging implications of IMS architecture
[331] 3GPP TS 32.225, Charging management; Charging data description for the IP
Multimedia Subsystem (IMS)
[332] 3GPP TS 32.240, Charging management; Charging architecture and principles
[333] 3GPP TS 32.260, Charging management; IP Multimedia Subsystem (IMS)

charging
[334] 3GPP TS 32.296, Charging management; Online Charging System (OCS):

Applications and interfaces
[335] 3GPP TS 33.203, 3G security; Access security for IP-based services
[336] 3GPP TS 33.210, Network Domain Security; IP network layer security
[337] 3GPP TS 33.220, Generic Authentication Architecture (GAA); Generic

bootstrapping architecture
[338] 3GPP TS 33.222, Generic Authentication Architecture (GAA);

Access to network application functions using Hypertext Transfer Protocol over
Transport Layer Security (HTTPS)
[339] 3GPP TS 29.109, Generic Authentication Architecture (GAA);

Zh and Zn Interfaces based on the Diameter protocol; Stage 3
[340] 3GPP TR 33.978, Security aspects of early IP Multimedia Subsystem (IMS)
[341] 3GPP TS 33.107, 3G security; Lawful interception architecture and functions
[342] 3GPP TS 33.108, 3G security; Handover interface for Lawful Interception (LI)
[343] 3GPP TS 22.250, IP Multimedia Subsystem (IMS) group management; Stage 1
[344] 3GPP TS 22.141, Presence Service; Stage 1
[345] 3GPP TS 23.141, Presence Service; Architecture and functional description
[346] 3GPP TS 24.141, Presence service using the IP Multimedia (IM)

Core Network (CN) subsystem; Stage 3
[347] 3GPP TR 24.841, Presence service based on Session Initiation Protocol (SIP);
Functional models, information flows and protocol details
[348] 3GPP TS 33.141, Presence service; Security
[349] 3GPP TR 23.867, Internet Protocol (IP) based IP Multimedia Subsystem (IMS)
emergency sessions, (Release 7)(RELEASE WITHDRAWN)
[350] 3GPP TS 24.147, Conferencing using the IP Multimedia (IM)

[351] 3GPP TS 22.340, IP Multimedia System (IMS) messaging; Stage 1
[352] 3GPP TR 22.940, IP Multimedia Subsystem (IMS) messaging
[353] 3GPP TR 23.804, Support of SMS and MMS over generic 3GPP IP access
[354] 3GPP TS 24.247, Messaging service using the IP Multimedia (IM) Core Network
(CN) subsystem; Stage 3
[355] 3GPP TS 23.234, 3GPP system to Wireless Local Area Network (WLAN)
interworking; System description
[356] 3GPP TR 23.836, Quality of Service (QoS) and policy aspects of

3GPP - Wireless Local Area Network (WLAN) interworking
[357] 3GPP TS 23.206, Voice Call Continuity (VCC) between Circuit Switched (CS) and
IP Multimedia Subsystem (IMS)
[358] 3GPP TS 22.173, IMS Multimedia Telephony service and Supplementary

Services
[359] 3GPP TS 22.228, Service requirements for the

Internet Protocol (IP) multimedia core network subsystem
[360] 3GPP TS 22.279, Combined Circuit Switched (CS) and IP Multimedia Subsystem
(IMS) sessions; stage 1
[361] 3GPP TS 23.008, Organization of Subscriber Data
[362] 3GPP TS 23.167, IP Multimedia Subsystem (IMS) Emergency Sessions
[363] 3GPP TS 23.198, Open Service Access (OSA); Stage 2
[364] 3GPP TS 23.203, Policy and charging control architecture
[365] 3GPP TS 23.204, Support of SMS over generic 3GPP IP access - Stage 2
[366] 3GPP TR 23.810, on Architecture Impacts of Service Brokering
[367] 3GPP TR 23.816, Identification of Communication Services in IMS
[368] 3GPP TR 23.979, Push-to-talk over Cellular (PoC) services;

Stage 2
[369] 3GPP TS 24.167, 3GPP IMS Management Object (MO); Stage 3
[370] 3GPP TS 24.173, IMS Multimedia Telephony Communication Service and

Supplementary Services; Stage 3
[371] 3GPP TS 24.206, Voice Call Continuity between the Circuit-Switched (CS)
domain and the IP Multimedia Core Network (CN) (IMS) subsystem - Stage 3
[372] 3GPP TS 24.279, Combining Circuit Switched (CS) and IP Multimedia Subsystem
(IMS) Services - Stage 3
[373] 3GPP TS 24.341, Support of SMS over IP networks - Stage 3
[374] 3GPP TR 24.930, Signalling Flows for the Session Setup in the IM CN Subsystem
based on SIP and SDP - Stage 3
[375] 3GPP TS 26.114, IMS - Multimedia Telephony - Media Handling and Interaction
[376] 3GPP TS 26.141, IP Multimedia System (IMS) Messaging and Presence - Media
Formats and Codecs
[377] 3GPP TS 26.235, Packet Switched Conversational Multimedia Applications -

Default Codecs
[378] 3GPP TS 26.236, Packet Switched Conversational Multimedia Applications -

Transport Protocols
[379] 3GPP TS 29.208, End-to-end Quality of Service (QoS) Signalling Flows
[380] 3GPP TS 29.234, 3GPP System to WLAN Interworking - Stage 3
[381] 3GPP TS 31.133, ISIM API for Java Card
[382] 3GPP TS 33.102, Security Architecture
[383] 3GPP TS 33.234, WLAN Interworking Security
[384] 3GPP TS 33.310, Network Domain Security (NDS) - Authentication Framework

(AF)
[385] 3GPP TR 33.803, Coexistence between TISPAN and 3GPP Authentication

Schemes
[386] 3GPP TS 32.272, Push-to-talk over Cellular (PoC) Charging
[387] 3GPP TS 32.297, Charging Management - Charging Data Record (CDR) File
Format and Transfer
[388] 3GPP TS 32.298, Charging Management - Charging Data Record (CDR)

Parameter Description
[389] 3GPP TS 24.229, Internet Protocol (IP) multimedia call control protocol based on
Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3
[390] 3GPP TS 29.212, Policy and Charging Control over Gx reference point; Stage 3
[391] 3GPP TS 29.213, Policy and charging control signalling flows and Quality of
Service (QoS) parameter mapping; Stage 3
[392] 3GPP TS 29.214, Policy and Charging Control over Rx reference point; Stage 3
[393] 3GPP TS 29.215, Policy and Charging Control over S9 reference point; Stage 3
[394] 3GPP TS 21.202, Technical Specifications and Technical Reports relating to the
Common IP Multimedia Subsystem (IMS); Stage 1
[395] 3GPP TS 21.111, Technical Specification Group Core Network and Terminals;
USIM and IC card requirements
[396] 3GPP TS 22.030, Man-Machine Interface (MMI) of the User Equipment (UE)
[397] 3GPP TS 31.101, UICC-terminal interface; Physical and logical characteristics
[398] 3GPP TS 31.103, Characteristics of the IP Multimedia Services Identity Module

(ISIM) application
[399] 3GPP TS 31.115, Secured packet structure for (Universal) Subscriber Identity
Module (U)SIM Toolkit applications
[400] 3GPP TS 31.116, Remote APDU Structure for (Universal) Subscriber Identity
Module (U)SIM Toolkit applications
[401] 3GPP TS 31.133, IP Multimedia Services Identity Module (ISIM) Application

Programming Interface (API); ISIM API for Java Card
[402] 3GPP TR 23.813, Study on Policy solutions and enhancements
[403] 3GPP TR 23.854, Enhancements for Multimedia Priority Service
[404] 3GPP TR 23.839, Study on Support of BBF Access Interworking
[405] 3GPP TS 23.216, Single Radio Voice Call Continuity (SRVCC)
[406] 3GPP TR 23.842, Study on Network Provided Location Information to the IMS
[407] 3GPP TR 23.861, Multi access PDN connectivity and IP flow mobility
[408] 3GPP TR 23.401, General Packet Radio Service (GPRS) enhancements for
Evolved Universal Terrestrial Radio Access Network (E-UTRAN) access
[409] 3GPP TR 23.402, Architecture enhancements for non-3GPP accesses
[410] 3GPP TS 23.261, IP flow mobility and seamless Wireless Local Area Network
(WLAN) offload
Appendix A a brief summary of all Common IMS specs
(3GPP specs, which are common across 3GPP, 3GPP2 and TISPAN).
Note: C6 means CT6, S1 means SA1 etc.
3GPP For
Type Number Title
Group publication?
TS 21.111 USIM and IC card requirements C6 Yes
TR 21.905 Vocabulary for 3GPP Specifications S1 Yes
TS 22.030 Man-Machine Interface (MMI) of the User S1 Yes
Equipment (UE)
TS 22.041 Operator Determined Call Barring (ODB) S1 Yes
TS 22.071 Location Services (LCS); Service description; S1 Yes
Stage 1
TS 22.101 Service aspects; Service principles S1 Yes
TS 22.105 Services and service capabilities S1 Yes
TS 22.115 Service aspects; Charging and billing S1 Yes
TS 22.127 Service requirement for the Open Services S1 Yes
Access (OSA); Stage 1
TS 22.140 Multimedia Messaging Service (MMS); Stage 1 S1 Yes
TS 22.141 Presence service; Stage 1 S1 Yes
TS 22.153 Multimedia priority service S1 Yes
TS 22.173 IP Multimedia Core Network Subsystem (IMS) S1 Yes
Multimedia Telephony Service and supplementary
services; Stage 1
TS 22.174 Push service; Stage 1 S1 Yes
TS 22.182 Customized Alerting Tone (CAT) requirements; S1 Yes
Stage 1
TS 22.228 Service requirements for the Internet Protocol (IP) S1 Yes
multimedia core network subsystem (IMS); Stage
1
TS 22.250 IP Multimedia Subsystem (IMS) Group S1 Yes
Management; Stage 1
TS 22.279 Combined Circuit Switched (CS) and IP S1 Yes
Multimedia Subsystem (IMS) sessions; Stage 1
TS 22.340 IP Multimedia Subsystem (IMS) messaging; S1 Yes
Stage 1
TR 22.979 Feasibility study on combined Circuit Switched S1 Yes
(CS) calls and IP Multimedia Subsystem (IMS)
sessions
TS 23.141 Presence service; Architecture and functional S2 Yes
description; Stage 2
TS 23.167 IP Multimedia Subsystem (IMS) emergency S2 Yes
3GPP For
Type Number Title
Group publication?
sessions
TS 23.204 Support of Short Message Service (SMS) over S2 Yes
generic 3GPP Internet Protocol (IP) access;
Stage 2
TS 23.218 IP Multimedia (IM) session handling; IM call C1 Yes
model; Stage 2
TS 24.141 Presence service using the IP Multimedia (IM) C1 Yes
TS 24.147 Conferencing using the IP Multimedia (IM) Core C1 Yes
Network (CN) subsystem; Stage 3
TS 24.173 IMS Multimedia telephony service and C1 Yes
supplementary services; Stage 3
TS 24.182 IP Multimedia Subsystem (IMS) Customized C1 Yes
Alerting Tones (CAT); Protocol specification
TS 24.229 Internet Protocol (IP) multimedia call control C1 Yes
protocol based on Session Initiation Protocol
(SIP) and Session Description Protocol (SDP);
Stage 3
TS 24.238 Session Initiation Protocol (SIP) based user C1 Yes
configuration; Stage 3
TS 24.239 Flexible Alerting (FA) using IP Multimedia (IM) C1 Yes
Core Network (CN) subsystem; Protocol
specification
TS 24.247 Messaging service using the IP Multimedia (IM) C1 Yes
TS 24.341 Support of SMS over IP networks; Stage 3 C1 Yes
TS 24.604 Communication Diversion (CDIV) using IP C1 Yes
Multimedia (IM)Core Network (CN) subsystem;
Protocol specification
TS 24.605 Conference (CONF) using IP Multimedia (IM) C1 Yes
Core Network (CN) subsystem; Protocol
specification
TS 24.606 Message Waiting Indication (MWI)using IP C1 Yes
Multimedia (IM) Core Network (CN) subsystem;
TS 24.607 Originating Identification Presentation (OIP) and C1 Yes
Originating Identification Restriction (OIR) using
IP Multimedia (IM) Core Network (CN)
subsystem; Protocol specification
TS 24.608 Terminating Identification Presentation (TIP) and C1 Yes
Terminating Identification Restriction (TIR)using
IP Multimedia (IM) Core Network (CN)
subsystem; Protocol specification
3GPP For
Type Number Title
Group publication?
TS 24.610 Communication HOLD (HOLD) using IP C1 Yes
TS 24.611 Anonymous Communication Rejection (ACR) and C1 Yes
Communication Barring (CB)using IP Multimedia
(IM) Core Network (CN) subsystem; Protocol
specification
TS 24.615 Communication Waiting (CW) using IP Multimedia C1 Yes
(IM) Core Network (CN) subsystem; Protocol
Specfication
TS 24.616 Malicious Communication Identification C1 Yes
(MCID)using IP Multimedia (IM) Core Network
(CN) subsystem; Protocol specification
TS 24.628 Common Basic Communication procedures using C1 Yes
IP Multimedia (IM)Core Network (CN) subsystem;
TS 24.629 Explicit Communication Transfer (ECT) using IP C1 Yes
TS 24.642 Completion of Communications to Busy C1 Yes
Subscriber (CCBS) and Completion of
Communications by No Reply (CCNR) using IP
Multimedia (IM)Core Network (CN) subsystem;
Protocol Specification
TS 24.647 Advice Of Charge (AOC) using IP Multimedia C1 Yes
(IM)Core Network (CN) subsystem; Protocol
Specification
TS 24.654 Closed User Group (CUG) using IP Multimedia C1 Yes
(IM) Core Network (CN) subsystem, Protocol
Specification
TR 24.930 Signalling flows for the session setup in the IP C1 Yes
Multimedia core network Subsystem (IMS) based
on Session Initiation Protocol (SIP) and Session
Description Protocol (SDP); Stage 3
TS 29.162 Interworking between the IM CN subsystem and C3 Yes
IP networks
TS 29.163 Interworking between the IP Multimedia (IM) Core C3 Yes
Network (CN) subsystem and Circuit Switched
(CS) networks
TS 29.165 Inter-IMS Network to Network Interface (NNI) C3 Yes
TS 29.212 Policy and charging control over Gx reference C3 Yes
point
TS 29.213 Policy and charging control signalling flows and C3 Yes
Quality of Service (QoS) parameter mapping
3GPP For
Type Number Title
Group publication?
TS 29.214 Policy and charging control over Rx reference C3 Yes
point
TS 29.215 Policy and Charging Control (PCC) over S9 C3 Yes
reference point
TS 29.228 IP Multimedia (IM) Subsystem Cx and Dx C4 Yes
Interfaces; Signalling flows and message contents
TS 29.229 Cx and Dx interfaces based on the Diameter C4 Yes
protocol; Protocol details
TS 29.292 Interworking between the IP Multimedia (IM) Core C3 Yes
Network (CN) subsystem (IMS) and MSC Server
for IMS Centralized Services (ICS)
TS 29.311 Service Level Interworking for Messaging C3 Yes
Services
TS 29.328 IP Multimedia Subsystem (IMS) Sh interface; C4 Yes
Signalling flows and message contents
TS 29.329 Sh interface based on the Diameter protocol; C4 Yes
Protocol details
TS 29.658 TISPAN; SIP Transfer of IP Multimedia Service C3 Yes
Tariff Information; Protocol specification
TS 31.101 UICC-terminal interface; Physical and logical C6 Yes
characteristics
TS 31.103 Characteristics of the IP Multimedia Services C6 Yes
Identity Module (ISIM) application
TS 31.115 Secured packet structure for (Universal) C6 Yes
Subscriber Identity Module (U)SIM Toolkit
applications
TS 31.116 Remote APDU Structure for (Universal) C6 Yes
Subscriber Identity Module (U)SIM Toolkit
applications
TS 31.133 IP Multimedia Services Identity Module (ISIM) C6 Yes
Application Programming Interface (API); ISIM
API for Java Card
TS 32.240 Telecommunication management; Charging S5 Yes
management; Charging architecture and
principles
management; IP Multimedia Subsystem (IMS)
charging
management; Diameter charging applications
TS 32.824 Telecommunication management; Service S5 No
Oriented Architecture (SOA) Integration
Reference Point (IRP) study
3GPP For
Type Number Title
Group publication?
TS 33.141 Presence service; Security S3 Yes
TS 33.203 3G security; Access security for IP-based S3 Yes
services
TS 33.210 3G security; Network Domain Security (NDS); IP S3 Yes
network layer security
Appendix B - Core IMS Specifications transferred from
ETSI TISPAN to 3GPP
Note: Notes and explanation about the color coding are provided under the table. This
table will soon be updated for 3GPP R9.
TISPAN doc topic TISPAN R1 frz R2 frz 3GPP 3GPP WG

WG v (Note 2) v (Note 2) Rel-7 frz v (Note 9) Rel-8 frz
ETSI TS 181 001 videotelephony 1 1.1.1 22.401 1.1.2 Note 1 S1
ETSI TS 181 002 PSTN/ISDN simulation 1 1.2.8 2.2.4 22.273 2.0.0 Note 8 S1
ETSI TS 181 005 Services, capabilities 1 1.1.1 2.4.1 S1
ETSI ES 282 007 IMS architecture 2 1.1.1 2.0.0 23.417 1.1.2 23.517 S2
ETSI TS 182 006 modified 23.228 2 1.1.1 2.0.3 23.406 1.1.2 23.506 S2
ETSI TS 182 007 VCC 2 0.3.2 23.507 S2
ETSI TS 182 008 presence 2 1.1.1 23.508 1.1.2 Note 1 S2
ETSI TS 182 009 emergency comms 2 1.1.1 23.509 1.1.2 Note 1 S2
ETSI TS 182 011 XML doc mgt 2 1.1.1 23.511 1.1.2 Note 1 S2
ETSI TS 182 021 SMS over IMS 2 0.0.7 23.521 S2
ETSI TS 183 004 CDIV 3 1.5.0 2.4.0 24.404 1.5.1 24.504 C1
ETSI TS 183 005 CONF 3 1.1.3 2.5.0 24.405 1.1.4 24.505 C1
ETSI TS 183 006 MWI 3 1.2.1 24.406 1.2.2 Note 1 C1
ETSI TS 183 007 OIP / OIR 3 1.3.0 24.407 1.3.1 Note 1 C1
ETSI TS 183 008 TIP / TIR 3 1.5.0 2.7.0 24.408 1.5.1 24.508 C1
ETSI TS 183 010 HOLD 3 1.2.2 24.410 1.2.3 Note 1 C1
ETSI TS 183 011 ACR / CB 3 1.3.0 24.411 1.3.1 Note 1 C1
ETSI TS 183 015 CW 3 1.0.1 C1
ETSI TS 183 016 MCID 3 1.2.1 2.5.0 24.416 1.2.2 24.516 C1
ETSI TS 183 021 endorse 29.162 3 1.1.1 29.421 1.1.2 Note 1 C3
ETSI TS 183 023 XCAP 3 1.4.0 24.423 1.4.1 Note 1 C1
ETSI TS 183 028 basic comm 3 1.6.0 2.4.0 24.428 1.6.1 24.528 C1
ETSI TS 183 029 ECT 3 1.3.2 2.5.0 24.429 1.3.3 24.529 C1
ETSI TS 183 033 endorse 29.228/'229 3 1.2.0 29.433 1.2.1 Note 1 C4
ETSI TS 183 041 endorse 24.247 3 1.1.1 24.441 1.1.2 Note 1 C1
ETSI TS 183 042 CCBS/CCNR 3 0.0.18 C1
ETSI TS 183 047 AOC 3 2.1.1 24.447 C1
ETSI TS 183 051 endorse 24.341 3 2.1.0 24.451 C1
ETSI TS 183 054 CUG 3 2.1.0 24.454 C1
ETSI TS 183 058 SIP tariff transfer 3 2.0.0 29.458 C3
ETSI ES 283 003 endorse 24.229 3 1.8.0 2.5.1 24.403 1.8.1 24.503 C1
ETSI ES 283 012 trunking gateway 3 1.1.1 29.412 1.1.2 Note 1 C3
ETSI ES 283 024 Trunking MGW 3 1.1.4 29.424 1.1.5 Note 1 C3
ETSI ES 283 027 endorse 29.163 3 1.4.0 2.4.0 29.427 1.4.1 29.527 C3
ETSI ES 283 030 endorse 24.141 3 1.1.1 24.430 1.1.2 Note 1 C1
Light green: need new spec to continue work within Rel-8 time frame?
Salmon: conversion done, 3GPP spec available
Yellow: conversion proposed (3GPP number not yet allocated)
Note 1: If no R2, use R1; no separate 3GPP Spec number required.

First instance transferred to 3GPP is 2x.4aa where aa is the same as the last two digits of the
TISPAN parent publication.
Where no R2 exists, R1 document is transferred to Rel-7, and is immediately approved and frozen.
Version 7.0.0 is then immediately upgraded to Rel-8 (v8.0.0) unchanged.
All transferred specs with one exception: see note 5) are immediately frozen after transfer, at both
Rel-7 and Rel-8. (A few do not exist at Rel-7.)
Note 2: Version in Italics = TISPAN not yet available for txfer (ie being processed towards ETSI publication
at a higher version).
Note 3: May be unnecessary to transfer this spec, since already superseded by 29.162. Nevertheless, CT3
decided to transfer to be on the safe side.
Note 4: (deleted)
Note 5: Not yet frozen at R2. therefore too early to create "post R2" 3GPP version; further development to
take place within 24.503.
Note 6: 3GPP version numbers take over where TISPAN left off. Thus they do not necessarily abide by the
rules of 21.900.
The version number in each 3GPP Release is stepped up one "editorial" place from the TISPAN
version number used as the basis.
Note 7: For TISPAN endorsements, continuing development will take place within the orignal endorsed
spec, so no new 2x.6xx spec is needed.
Note 8: TS 181 002 not frozen at R2, so work continues in 22.173 (Rel-8).
Note 9: The versions shown are for the first instance of the 3GPP spec concerned. Obviously these will be
raised to v7.0.0 / 8.0.0 as soon as they are approved. There is no intention to maintain this page to
show the most current version.
Note
Some elements of original TISPAN document to be transferred, others to remain.
10:
Appendix C - IMS related Specifications and Reports in
3GPP2
Note: The table below shows the 3GPP2 publications relating to core functions of the IP
Multimedia Subsystem (IMS) used by the 3GPP2. Also shown in the table is the
mapping between the replaced 3GPP2 MMD specifications and the corresponding
3GPP IMS specifications which replace them and the final revision of the document that
was published by 3GPP2.
3GPP2 Document 3GPP2 Document Title 3GPP Rel-8 TS/TR

Number and
Revision
X.S0013-000-B v1.0 Overview TS 23.002
TS 23.228
X.S0013-002-B v1.0 IP Multimedia Subsystem TS 23.228
Stage 2 TS 23.002
X.S0013-003-B v1.0 IP Multimedia Session TS 23.218
Handling; IP Multimedia Call
Model Stage 2
X.S0013-004-B v1.0 IP Multimedia Call Control TS 24.229
Protocol Based on SIP and
SDP - Stage 3
X.S0013-005-B v1.0 IP Multimedia Subsystem Cx TS 29.228
Interface Signaling flows and
Message Contents
X.S0013-006-B v1.0 Cx Interface Based on the TS 29.229
Diameter Protocol; Protocol
Details
X.S0013-007-A v1.0 IP Multimedia Subsystem - TS 32.240
Charging Architecture TS 32.260
X.S0013-008-A v1.0 IP Multimedia Subsystem - TS 32.260
Offline Accounting Information TS 32.299
Flows and Protocol
X.S0013-009-0 v1.0 IMS/MMD Call Flow Examples TR 24.930
X.S0013-010-B v1.0 IP Multimedia Subsystem Sh TS 29.328
interface; Signaling flows and
message contents Stage 2
X.S0013-011-B v1.0 Sh Interface based on TS 29.329
Diameter Protocols Protocol
Details Stage 3
X.S0013-012-0 v1.0 Service Based Bearer Control TS 23.203

Stage 2
Tx Interface Stage 3 TS 29.214
Ty Interface Stage 3 TS 29.215
X.S0013-016-0 v1.0 Messaging Service Using the TS 24.247
IP Multimedia Subsystem
X.S0027-000-A v1.0 Presence Overview No 3GPP equivalent
X.S0027-001-0 v1.0 Presence Service: Architecture TS 23.141
and Functional Description
X.S0027-002-0 v1.0 Presence Security TS 33.141
X.S0027-003-0 v1.0 Presence Stage 3 TS 24.141
X.S0027-004-0 v1.0 Network Presence No 3GPP equivalent
X.S0029-0 v1.0 Conferencing Using the IP TS 24.147
Multimedia (IM) Core Network
(CN) Subsystem
X.S0049-0 v1.0 All-IP Network Emergency Call TS 23.167
Support TS 24.229
X.S0055-0 v1.0 MMD Supplementary Services TS 24.173
TS 24.182
TS 24.238
TS 24.239
TS 24.604
TS 24.605
TS 24.606
TS 24.607
TS 24.608
TS 24.610
TS 24.611
TS 24.615
TS 24.628
TS 24.629
S.S0086-B v2.0 IMS Security Framework TS 33.203
TS 33.210
S.R0058 IP Multimedia Domain TS 22.228
System Requirements
S.R0062 Presence for Wireless TS 22.141
Systems Stage 1
Requirements
S.R0125 VoIP Supplementary Services TS 22.173
Feature Description
X.R0052-0 All-IP System MMD No 3GPP equivalent
Roaming Technical Report
X.S0042 Voice Call Continuity between TS 23.206
IMS and Circuit Switched TS 24.206
Systems
C.S0069 ISIM Application on UICC for TS 31.103
cdma2000 Spread Spectrun
Systems

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SIPKnowledge

SIP/IMS Specifications For Dummies

Document ID: SK_500195

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SIPKnowledge, 2012 Copyright

1.1 Purpose of this e-book

1.2 Acronyms, Abbreviations and Symbols

IETF Standards-track-category (per section 4.1)

(S) = Standards Track (Proposed Standard, Draft Standard, or Standard)

(B) = Best Current Practice

3GPP (www.3gpp.org) The 3rd Generation Partnership Project (3GPP) is a

3GPP2 (http://www.3gpp2.org CDMA2000), TISPAN (http://www.etsi.org/tispan/ -

1.5 SIP/VoIP related IETF Work Groups (WGs)

- The XCON WG - Centralized Conferencing - http://www.ietf.org/html.charters/xcon-

Data for Reachability/Routing

- The DRINKS WG - Data for Reachability of Inter/tra-NetworK SIP -

- The BLISS WG - Basic Level of Interoperability for SIP Services -

- The IMPP WG - Instant Messaging and Presence Protocol

IP Telephony (VoIP and beyond)

- The AVT WG - Audio/Video Transport (RTP) -

Interop with Circuit domain

- The ENUM WG - Telephone Number Mapping -

- The MEDIACTRL WG - Media Server Control -

- The GEOPRIV WG - Geographic Location/Privacy

- The DIFFSERV WG Differentiated Services (QoS in backbone) -

Peer-to-Peer paradigm for SIP

- The P2PSIP WG - Peer-to-Peer Session Initiation Protocol -

Security and Emergency Calls (911)

- The MSEC WG - Multicast Security - http://www.ietf.org/html.charters/msec-

1.6 IMS related 3GPP Specification Groups (SGs)

Call Control, Session Management, Mobility Management, SMS

OSA - Open Service Access (transferred to OMA -

Smart Card Application Aspects

1.7 Download Tips:

Download Tips: You can download an IETF RFC by going to

2.1 Scope of the SIP specifications listed in this e-book.

Any specification that defines an extension to SIP itself, where an extension is a

Any specification that defines an extension to SDP whose primary purpose is to

Any specification that defines a MIME object whose primary purpose is to

We did not include requirements, architectures, registry, definitions, non-normative

1. Standards Track (Proposed Standard, Draft Standard, or Standard)

3. Best Current Practice

2.2.2 List of specifications

RFC 5939, SDP Capability Negotiation (S): I-D.ietf-mmusic-sdp-capability-negotiation

Numerous extensions and usages of SIP related to interoperability and

2.3.2 List of specifications

2.4 General Purpose Infrastructure Extensions

2.4.2 List of specifications

2.5 NAT Traversal

2.5.2 List of specifications

Draft-ietf-mmusic-ice-tcp, TCP Candidates with Interactive Connectivity Establishment

2.6 Minor Extensions

RFC 5898, Connectivity Preconditions for Session Description Protocol (SDP)

2.7.2 List of specifications

2.8 Call Control Primitives

2.8.2 List of specifications

2.9 Event Framework and Packages

See description of RFC 3265 below.