draft-ietf-sip-hitchhikers-guide-06.txt   rfc5411.txt 
SIP J. Rosenberg Network Working Group J. Rosenberg
Internet-Draft Cisco Request for Comments: 5411 Cisco
Intended status: Informational November 3, 2008 Category: Informational February 2009
Expires: May 7, 2009
A Hitchhiker's Guide to the Session Initiation Protocol (SIP) A Hitchhiker's Guide to the Session Initiation Protocol (SIP)
draft-ietf-sip-hitchhikers-guide-06
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Abstract Abstract
The Session Initiation Protocol (SIP) is the subject of numerous The Session Initiation Protocol (SIP) is the subject of numerous
specifications that have been produced by the IETF. It can be specifications that have been produced by the IETF. It can be
difficult to locate the right document, or even to determine the set difficult to locate the right document, or even to determine the set
of Request for Comments (RFC) about SIP. This specification serves of Request for Comments (RFC) about SIP. This specification serves
as a guide to the SIP RFC series. It lists a current snapshot of the as a guide to the SIP RFC series. It lists a current snapshot of the
specifications under the SIP umbrella, briefly summarizes each, and specifications under the SIP umbrella, briefly summarizes each, and
groups them into categories. groups them into categories.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction ....................................................2
2. Scope of this Document . . . . . . . . . . . . . . . . . . . . 5 2. Scope of This Document ..........................................4
3. Core SIP Specifications . . . . . . . . . . . . . . . . . . . 6 3. Core SIP Specifications .........................................5
4. Public Switched Telephone Network (PSTN) Interworking . . . . 9 4. Public Switched Telephone Network (PSTN) Interworking ...........9
5. General Purpose Infrastructure Extensions . . . . . . . . . . 11 5. General Purpose Infrastructure Extensions ......................11
6. NAT Traversal . . . . . . . . . . . . . . . . . . . . . . . . 13 6. NAT Traversal ..................................................13
7. Call Control Primitives . . . . . . . . . . . . . . . . . . . 14 7. Call Control Primitives ........................................14
8. Event Framework . . . . . . . . . . . . . . . . . . . . . . . 15 8. Event Framework ................................................15
9. Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 16 9. Event Packages .................................................16
10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 17 10. Quality of Service ............................................17
11. Operations and Management . . . . . . . . . . . . . . . . . . 18 11. Operations and Management .....................................18
12. SIP Compression . . . . . . . . . . . . . . . . . . . . . . . 19 12. SIP Compression ...............................................18
13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 19 13. SIP Service URIs ..............................................18
14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 20 14. Minor Extensions ..............................................20
15. Security Mechanisms . . . . . . . . . . . . . . . . . . . . . 22 15. Security Mechanisms ...........................................21
16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 26 16. Conferencing ..................................................25
17. Instant Messaging, Presence and Multimedia . . . . . . . . . . 27 17. Instant Messaging, Presence, and Multimedia ...................26
18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 27 18. Emergency Services ............................................26
19. Security Considerations . . . . . . . . . . . . . . . . . . . 28 19. Security Considerations .......................................26
20. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 28 20. Acknowledgements ..............................................27
21. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 28 21. Informative References ........................................27
22. Informative References . . . . . . . . . . . . . . . . . . . . 28
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 41
Intellectual Property and Copyright Statements . . . . . . . . . . 42
1. Introduction 1. Introduction
The Session Initiation Protocol (SIP) [RFC3261] is the subject of The Session Initiation Protocol (SIP) [RFC3261] is the subject of
numerous specifications that have been produced by the IETF. It can numerous specifications that have been produced by the IETF. It can
be difficult to locate the right document, or even to determine the be difficult to locate the right document, or even to determine the
set of Request for Comments (RFC) about SIP. Don't Panic! [HGTTG] set of Request for Comments (RFC) about SIP. "Don't Panic!" [HGTTG]
This specification serves as a guide to the SIP RFC series. It is a This specification serves as a guide to the SIP RFC series. It is a
current snapshot of the specifications under the SIP umbrella at the current snapshot of the specifications under the SIP umbrella at the
time of publication. It is anticipated that this document itself time of publication. It is anticipated that this document itself
will be regularly updated as SIP specifications mature. Furthermore, will be regularly updated as SIP specifications mature. Furthermore,
it references many specifications, which, at time of publication of it references many specifications, which, at the time of publication
this document, were not yet finalized, and may eventually be of this document, were not yet finalized, and may eventually be
completed or abandoned. Therefore, the enumeration of specifications completed or abandoned. Therefore, the enumeration of specifications
here is a work-in-progress and subject to change. here is a work-in-progress and subject to change.
For each specification, a paragraph or so description is included For each specification, a paragraph or so description is included
that summarizes the purpose of the specification. Each specification that summarizes the purpose of the specification. Each specification
also includes a letter that designates its category in the standards also includes a letter that designates its category in the Standards
track [RFC2026]. These values are: Track [RFC2026]. These values are:
S: Standards Track (Proposed Standard, Draft Standard, or Standard) S: Standards Track (Proposed Standard, Draft Standard, or Standard)
E: Experimental E: Experimental
B: Best Current Practice B: Best Current Practice
I: Informational I: Informational
The specifications are grouped together by topic. The topics are: The specifications are grouped together by topic. The topics are:
Core: The essential SIP specifications that are expected to be Core: The SIP specifications that are expected to be utilized for
utilized for every session or registration. each session or registration an endpoint participates in.
PSTN Interop: Specifications related to interworking with the Public Switched Telephone Network (PSTN) Interop: Specifications
telephone network. related to interworking with the telephone network.
General Purpose Infrastructure: General purpose extensions to SIP, General Purpose Infrastructure: General purpose extensions to SIP,
SDP and MIME, but ones that are not expected to always be used. SDP (Session Description Protocol), and MIME, but ones that are
not expected to always be used.
NAT Traversal: Specifications to deal with firewall and NAT NAT Traversal: Specifications to deal with firewall and NAT
traversal. traversal.
Minor Extensions: Specifications that solve a narrow problem space
or provide an optimization.
Conferencing: Specifications for multimedia conferencing.
Call Control Primitives: Specifications for manipulating SIP dialogs Call Control Primitives: Specifications for manipulating SIP dialogs
and calls. and calls.
Event Framework: Defines the core specifications for the SIP event Event Framework: Definitions of the core specifications for the SIP
framework, providing for pub/sub capability. event framework, providing for pub/sub capability.
Event Packages: Packages that utilize the SIP event framework. Event Packages: Packages that utilize the SIP event framework.
Quality of Service: Specifications related to multimedia quality of Quality of Service: Specifications related to multimedia quality of
service (QoS). service (QoS).
Operations and Management: Specifications related to configuration Operations and Management: Specifications related to configuration
and monitoring of SIP deployments. and monitoring of SIP deployments.
SIP Compression: Specifications to facilitate usage of SIP with the SIP Compression: Specifications to facilitate usage of SIP with the
Signaling Compression (Sigcomp) framework. Signaling Compression (Sigcomp) framework.
SIP Service URIs: Specifications on how to use SIP URIs to address SIP Service URIs: Specifications on how to use SIP URIs to address
multimedia services. multimedia services.
Minor Extensions: Specifications that solve a narrow problem space
or provide an optimization.
Security Mechanisms: Specifications providing security functionality Security Mechanisms: Specifications providing security functionality
for SIP. for SIP.
Conferencing: Specifications for multimedia conferencing.
Instant Messaging, Presence, and Multimedia: SIP extensions related Instant Messaging, Presence, and Multimedia: SIP extensions related
to IM, presence and multimedia. This covers only the SIP to IM, presence, and multimedia. This covers only the SIP
extensions related to these topics. See [I-D.ietf-simple-simple] extensions related to these topics. See [SIMPLE] for a full
for a full treatment of SIP for IM and Presence (SIMPLE). treatment of SIP for IM and Presence (SIMPLE).
Emergency Services: SIP extensions related to emergency services. Emergency Services: SIP extensions related to emergency services.
See [I-D.ietf-ecrit-framework] for a more complete treatment of See [ECRIT-FRAME] for a more complete treatment of additional
additional functionality related to emergency services. functionality related to emergency services.
Typically, SIP extensions fit naturally into topic areas, and Typically, SIP extensions fit naturally into topic areas, and
implementers interested in a particular topic often implement many or implementors interested in a particular topic often implement many or
all of the specifications in that area. There are some all of the specifications in that area. There are some
specifications which fall into multiple topic areas, in which case specifications that fall into multiple topic areas, in which case
they are listed more than once. they are listed more than once.
Do not print all the specs cited here at once, as they might share Do not print all the specs cited here at once, as they might share
the fate of the rules of Brockian Ultracricket when bound together: the fate of the rules of Brockian Ultracricket when bound together:
collapse under their own gravity and form a black hole [HGTTG]. collapse under their own gravity and form a black hole [HGTTG].
This document itself is not an update to RFC 3261 or an extension to This document itself is not an update to RFC 3261 or an extension to
SIP. It is an informational document, meant to guide newcomers, SIP. It is an informational document, meant to guide newcomers,
implementors and deployers to the many of the specifications implementors, and deployers to the many specifications associated
associated with SIP. with SIP.
2. Scope of this Document 2. Scope of This Document
It is very difficult to enumerate the set of SIP specifications. It is very difficult to enumerate the set of SIP specifications.
This is because there are many protocols that are intimately related This is because there are many protocols that are intimately related
to SIP and used by nearly all SIP implementations, but are not to SIP and used by nearly all SIP implementations, but are not
formally SIP extensions. As such, this document formally defines a formally SIP extensions. As such, this document formally defines a
"SIP specification" as: "SIP specification" as:
o RFC 3261 and any specification that defines an extension to it, o RFC 3261 and any specification that defines an extension to it,
where an extension is a mechanism that changes or updates in some where an extension is a mechanism that changes or updates in some
way a behavior specified there. way a behavior specified there.
o The basic SDP specification, RFC 4566 [RFC4566], and any o The basic SDP specification [RFC4566] and any specification that
specification that defines an extension to SDP whose primary defines an extension to SDP whose primary purpose is to support
purpose is to support SIP. SIP.
o Any specification that defines a MIME object whose primary purpose o Any specification that defines a MIME object whose primary purpose
is to support SIP is to support SIP.
Excluded from this list are requirements, architectures, registry Excluded from this list are requirements, architectures, registry
definitions, non-normative frameworks, and processes. Best Current definitions, non-normative frameworks, and processes. Best Current
Practices are included when they normatively define mechanisms for Practices are included when they normatively define mechanisms for
accomplishing a task, or provide significant description of the usage accomplishing a task, or provide significant description of the usage
of the normative specifications, such as call flows. of the normative specifications, such as call flows.
The SIP change process [RFC3427] defines two types of extensions to The SIP change process [RFC3427] defines two types of extensions to
SIP. These are normal extensions and the so-called P-headers (where SIP: normal extensions and the so-called P-headers (where P stands
P stands for "preliminary", "private", or "proprietary", and the "P-" for "preliminary", "private", or "proprietary", and the "P-" prefix
prefix is included in the header field name), which are meant to be is included in the header field name), which are meant to be used in
used in areas of limited applicability. P-headers cannot be defined areas of limited applicability. P-headers cannot be defined in the
in the standards track. For the most part, P-headers are not Standards Track. For the most part, P-headers are not included in
included in the listing here, with the exception of those which have the listing here, with the exception of those that have seen general
seen general usage despite their P-header status. usage despite their P-header status.
This document includes specifications which have already been This document includes specifications, which have already been
approved by the IETF and granted an RFC number, in addition to approved by the IETF and granted an RFC number, in addition to
Internet Drafts which are still under development within IETF and Internet Drafts, which are still under development within the IETF
will eventually finish and get an RFC number. Inclusion of Internet and will eventually finish and get an RFC number. Inclusion of
Drafts here helps encourage early implementation and demonstrations Internet Drafts here helps encourage early implementation and
of interoperability of the protocol, and thus aids in the standards demonstrations of interoperability of the protocol, and thus aids in
setting process. Inclusion of these also identifes where the IETF is the standards-setting process. Inclusion of these also identifies
targetting a solution at a particular problem space. Note that final where the IETF is targeting a solution at a particular problem space.
IANA assignment of codepoints (such as option tags and header field Note that final IANA assignment of codepoints (such as option tags
names) does not take place until shortly before publication as an and header field names) does not take place until shortly before
RFC, and thus codepoint assignments may change. publication as an RFC, and thus codepoint assignments may change.
3. Core SIP Specifications 3. Core SIP Specifications
The core SIP specifications represent the set of specifications whose The core SIP specifications represent the set of specifications whose
functionality is broadly applicable. An extension is broadly functionality is broadly applicable. An extension is broadly
applicable if it fits into one of the following categories: applicable if it fits into one of the following categories:
o For specifications that impact SIP session management, the o For specifications that impact SIP session management, the
extension would be used for almost every session initiated by a extension would be used for almost every session initiated by a
user agent user agent.
o For specifications that impact SIP registrations, the extension o For specifications that impact SIP registrations, the extension
would be used for almost every registration initiated by a user would be used for almost every registration initiated by a user
agent agent.
o For specifications that impact SIP subscriptions, the extension o For specifications that impact SIP subscriptions, the extension
would be used for almost every subscription initiated by a user would be used for almost every subscription initiated by a user
agent agent.
In other words, these are not specifications that are used just for In other words, these are not specifications that are used just for
some requests and not others; they are specifications that would some requests and not others; they are specifications that would
apply to each and every request that the extension is relevant for. apply to each and every request for which the extension is relevant.
In the galaxy of SIP, these specifications are like towels [HGTTG]. In the galaxy of SIP, these specifications are like towels [HGTTG].
RFC 3261, The Session Initiation Protocol (S): [RFC3261] is the core RFC 3261, The Session Initiation Protocol (S): [RFC3261] is the core
SIP protocol itself. RFC 3261 is an update to [RFC2543]. It is SIP protocol itself. RFC 3261 obsoletes [RFC2543]. It is the
the president of the galaxy [HGTTG] as far as the suite of SIP president of the galaxy [HGTTG] as far as the suite of SIP
specifications is concerned. specifications is concerned.
RFC 3263, Locating SIP Servers (S): [RFC3263] provides DNS RFC 3263, Locating SIP Servers (S): [RFC3263] provides DNS
procedures for taking a SIP URI, and determining a SIP server that procedures for taking a SIP URI and determining a SIP server that
is associated with that SIP URI. RFC 3263 is essential for any is associated with that SIP URI. RFC 3263 is essential for any
implementation using SIP with DNS. RFC 3263 makes use of both DNS implementation using SIP with DNS. RFC 3263 makes use of both DNS
SRV records [RFC2782] and NAPTR records [RFC2915]. SRV records [RFC2782] and NAPTR records [RFC3401].
RFC 3264, An Offer/Answer Model with the Session Description Protocol RFC 3264, An Offer/Answer Model with the Session Description Protocol
(S): [RFC3264] defines how the Session Description Protocol (SDP) (S): [RFC3264] defines how the Session Description Protocol (SDP)
[RFC4566] is used with SIP to negotiate the parameters of a media [RFC4566] is used with SIP to negotiate the parameters of a media
session. It is in widespread usage and an integral part of the session. It is in widespread usage and an integral part of the
behavior of RFC 3261. behavior of RFC 3261.
RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the
SUBSCRIBE and NOTIFY methods. These two methods provide a general SUBSCRIBE and NOTIFY methods. These two methods provide a general
event notification framework for SIP. To actually use the event notification framework for SIP. To actually use the
framework, extensions need to be defined for specific event framework, extensions need to be defined for specific event
packages. An event package defines a schema for the event data, packages. An event package defines a schema for the event data
and describes other aspects of event processing specific to that and describes other aspects of event processing specific to that
schema. An RFC 3265 implementation is required when any event schema. An RFC 3265 implementation is required when any event
package is used. package is used.
RFC 3325, Private Extensions to SIP for Asserted Identity within RFC 3325, Private Extensions to SIP for Asserted Identity within
Trusted Networks (I): Though its P-header status implies that it has Trusted Networks (I): Though its P-header status implies that it
limited applicability, [RFC3325], which defines the P-Asserted- has limited applicability, [RFC3325], which defines the
Identity header field, has been widely deployed. It is used as P-Asserted-Identity header field, has been widely deployed. It is
the basic mechanism for providing network asserted caller ID used as the basic mechanism for providing network-asserted caller
services. Its update, [I-D.ietf-sipping-update-pai], clarifies ID services. Its intended update, [UPDATE-PAI], clarifies its
its usage for connected party identification as well. usage for connected party identification as well.
RFC 3327, SIP Extension Header Field for Registering Non-Adjacent RFC 3327, SIP Extension Header Field for Registering Non-Adjacent
Contacts (S): [RFC3327] defines the Path header field. This field Contacts (S): [RFC3327] defines the Path header field. This
is inserted by proxies between a client and their registrar. It field is inserted by proxies between a client and their registrar.
allows inbound requests towards that client to traverse these It allows inbound requests towards that client to traverse these
proxies prior to being delivered to the user agent. It is proxies prior to being delivered to the user agent. It is
essential in any SIP deployment that has edge proxies, which are essential in any SIP deployment that has edge proxies, which are
proxies between the client and the home proxy or SIP registrar. proxies between the client and the home proxy or SIP registrar.
RFC 3581, An Extension to SIP for Symmetric Response Routing (S): RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
[RFC3581] defines the rport parameter of the Via header. It [RFC3581] defines the rport parameter of the Via header. It
allows SIP responses to traverse NAT. It is one of several allows SIP responses to traverse NAT. It is one of several
specifications that are utilized for NAT traversal (see specifications that are utilized for NAT traversal (see
Section 6). Section 6).
RFC 3840, Indicating User Agent Capabilities in SIP (S): [RFC3840] RFC 3840, Indicating User Agent Capabilities in SIP (S): [RFC3840]
defines a mechanism for carrying capability information about a defines a mechanism for carrying capability information about a
user agent in REGISTER requests and in dialog-forming requests user agent in REGISTER requests and in dialog-forming requests
like INVITE. It has found use with conferencing (the isfocus like INVITE. It has found use with conferencing (the isfocus
parameter declares that a user agent is a conference server) and parameter declares that a user agent is a conference server) and
with applications like push-to-talk. with applications like push-to-talk.
RFC 4320, Actions Addressing Issues Identified with the Non-INVITE RFC 4320, Actions Addressing Issues Identified with the Non-INVITE
Transaction in SIP (S): [RFC4320] formally updates RFC 3261, and Transaction in SIP (S): [RFC4320] formally updates RFC 3261 and
modifies some of the behaviors associated with non-INVITE modifies some of the behaviors associated with non-INVITE
transactions. This addresses some problems found in timeout and transactions. This addresses some problems found in timeout and
failure cases. failure cases.
RFC 4474, Enhancements for Authenticated Identity Management in SIP RFC 4474, Enhancements for Authenticated Identity Management in SIP
(S): [RFC4474] defines a mechanism for providing a cryptographically (S): [RFC4474] defines a mechanism for providing a
verifiable identity of the calling party in a SIP request. Known cryptographically verifiable identity of the calling party in a
as "SIP Identity", this mechanism provides an alternative to RFC SIP request. Known as "SIP Identity", this mechanism provides an
3325. It has seen little deployment so far, but its importance as alternative to RFC 3325. It has seen little deployment so far,
a key construct for anti-spam techniques and new security but its importance as a key construct for anti-spam techniques and
mechanisms makes it a core part of the SIP specifications. new security mechanisms makes it a core part of the SIP
specifications.
draft-ietf-sip-gruu, Obtaining and Using Globally Routable User Agent GRUU, Obtaining and Using Globally Routable User Agent Identifiers
Identifiers (GRUU) in SIP (S): [I-D.ietf-sip-gruu] defines a (GRUU) in SIP (S): [GRUU] defines a mechanism for directing
mechanism for directing requests towards a specific UA instance. requests towards a specific UA instance. GRUU is essential for
GRUU is essential for features like transfer and provides another features like transfer and provides another piece of the SIP NAT
piece of the SIP NAT traversal story. traversal story.
draft-ietf-sip-outbound, Managing Client Initiated Connections OUTBOUND, Managing Client Initiated Connections through SIP (S):
through SIP (S): [I-D.ietf-sip-outbound], also known as SIP [OUTBOUND], also known as SIP outbound, defines important changes
outbound, defines important changes to the SIP registration to the SIP registration mechanism that enable delivery of SIP
mechanism which enable delivery of SIP messages towards a UA when messages towards a UA when it is behind a NAT. This specification
it is behind a NAT. This specification is the cornerstone of the is the cornerstone of the SIP NAT traversal strategy.
SIP NAT traversal strategy.
RFC 4566, Session Description Protocol (S): [RFC4566] defines a RFC 4566, Session Description Protocol (S): [RFC4566] defines a
format for representing multimedia sessions. SDP objects are format for representing multimedia sessions. SDP objects are
carried in the body of SIP messages, and based on the offer/answer carried in the body of SIP messages and, based on the offer/answer
model, are used to negotiate the media characteristics of a model, are used to negotiate the media characteristics of a
session between users. session between users.
draft-ietf-mmusic-sdp-capability-negotiation, SDP Capability SDP-CAP, SDP Capability Negotiation (S): [SDP-CAP] defines a set of
Negotiation (S): [I-D.ietf-mmusic-sdp-capability-negotiation] extensions to SDP that allows for capability negotiation within
defines a set of extensions to SDP that allow for capability SDP. Capability negotiation can be used to select between
negotiation within SDP. Capability negotiation can be used to different profiles of RTP (secure vs. unsecure) or to negotiate
select between different profiles of RTP (secure vs. unsecure) or codecs such that an agent has to select one amongst a set of
to negotiate codecs such that an agent has to select one amongst a supported codecs.
set of supported codecs.
draft-ietf-mmusic-ice, Interactive Connectivity Establishment (ICE) ICE, Interactive Connectivity Establishment (ICE) (S): [ICE] defines
(S): [I-D.ietf-mmusic-ice] defines a technique for NAT traversal of a technique for NAT traversal of media sessions for protocols that
media sessions for protocols that make use of the offer/answer make use of the offer/answer model. This specification is the
model. This specification is the IETF recommended mechanism for IETF-recommended mechanism for NAT traversal for SIP media
NAT traversal for SIP media streams, and is meant to be used even streams, and is meant to be used even by endpoints that are
by endpoints which are themselves never behind a NAT. A SIP themselves never behind a NAT. A SIP option tag and media feature
option tag and media feature tag [I-D.ietf-sip-ice-option-tag] tag [OPTION-TAG] (also a core specification) have been defined for
(also a core specification) have been defined for use with ICE. use with ICE.
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
Description Protocol (SDP) (S): [RFC3605] defines a way to Description Protocol (SDP) (S): [RFC3605] defines a way to
explicitly signal, within an SDP message, the IP address and port explicitly signal, within an SDP message, the IP address and port
for RTCP, rather than using the port+1 rule in the Real Time for RTCP, rather than using the port+1 rule in the Real Time
Transport Protocol (RTP) [RFC3550]. It is needed for devices Transport Protocol (RTP) [RFC3550]. It is needed for devices
behind NAT and used by ICE. behind NAT, and the specification is required by ICE.
RFC 4916, Connected Identity in the Session Initiation Protocol (SIP) RFC 4916, Connected Identity in the Session Initiation Protocol (SIP)
(S): [RFC4916] formally updates RFC 3261. It defines an extension (S): [RFC4916] formally updates RFC 3261. It defines an
to SIP that allows a calling user to determine the identity of the extension to SIP that allows a calling user to determine the
final called user (connected party). Due to forwarding and identity of the final called user (connected party). Due to
retargeting services, this may not be the same as the user that forwarding and retargeting services, this may not be the same as
the caller was originally trying to reach. The mechanism works in the user that the caller was originally trying to reach. The
tandem with the SIP identity specification [RFC4474] to provide mechanism works in tandem with the SIP identity specification
signatures over the connected party identity. It can also be used [RFC4474] to provide signatures over the connected party identity.
if a party identity changes mid call due to third party call It can also be used if a party identity changes mid-call due to
control actions or PSTN behavior. third-party call control actions or PSTN behavior.
RFC 3311, The SIP UPDATE Method (S): [RFC3311] defines the UPDATE RFC 3311, The SIP UPDATE Method (S): [RFC3311] defines the UPDATE
method for SIP. This method is meant as a means for updating method for SIP. This method is meant as a means for updating
session information prior to the completion of the initial INVITE session information prior to the completion of the initial INVITE
transaction. It can also be used to update other information, transaction. It can also be used to update other information,
such as the identity of the participant [RFC4916], without such as the identity of the participant [RFC4916], without
involving an updated offer/answer exchange. It was developed involving an updated offer/answer exchange. It was developed
initially to support [RFC3312] but has found other uses. In initially to support [RFC3312], but has found other uses. In
particular, its usage with RFC 4916 means it will typically be particular, its usage with RFC 4916 means it will typically be
used as part of every session, to convey a secure connected used as part of every session, to convey a secure, connected
identity. identity.
draft-ietf-sip-sips, The use of the SIPS URI Scheme in the Session SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
Initiation Protocol (SIP) (S): [I-D.ietf-sip-sips] formally updated Protocol (SIP) (S): [SIPS-URI] is intended to update RFC 3261.
RFC 3261. It revises the processing of the SIPS URI, originally It revises the processing of the SIPS URI, originally defined in
defined in RFC 3261, to fix many errors and problems that have RFC 3261, to fix many errors and problems that have been
been encountered with that mechanism. encountered with that mechanism.
RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples
(B): [RFC3665] contains best practice call flow examples for basic (B): [RFC3665] contains best-practice call flow examples for
SIP interactions - call establishment, termination, and basic SIP interactions -- call establishment, termination, and
registration. registration.
Essential Corrections to SIP: A collection of fixes to SIP that Essential Corrections to SIP: A collection of fixes to SIP that
address important bugs and vulnerabilities. These include a fix address important bugs and vulnerabilities. These include a fix
requiring loop detection in any proxy that forks requiring loop detection in any proxy that forks [LOOP-FIX], a
[I-D.ietf-sip-fork-loop-fix], a clarification on how record- clarification on how record-routing works [RECORD-ROUTE], and a
routing works [I-D.ietf-sip-record-route-fix], and a correction to correction to the IPv6 BNF [ABNF-FIX].
the IPv6 BNF [I-D.ietf-sip-ipv6-abnf-fix].
4. Public Switched Telephone Network (PSTN) Interworking 4. Public Switched Telephone Network (PSTN) Interworking
Numerous extensions and usages of SIP related to interoperability and Numerous extensions and usages of SIP are related to interoperability
communications with or through the PSTN. and communications with or through the PSTN.
RFC 2848, The PINT Service Protocol (S): [RFC2848] is one of the RFC 2848, The PINT Service Protocol (S): [RFC2848] is one of the
earliest extensions to SIP. It defines procedures for using SIP earliest extensions to SIP. It defines procedures for using SIP
to invoke services that actually execute on the PSTN. Its main to invoke services that actually execute on the PSTN. Its main
application is for third party call control, allowing an IP host application is for third-party call control, allowing an IP host
to set up a call between two PSTN endpoints. PINT has a to set up a call between two PSTN endpoints. PINT (PSTN/Internet
relatively narrow focus and has not seen widespread deployment. Interworking) has a relatively narrow focus and has not seen
widespread deployment.
RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming
PSTN related extensions with alcohol references, SPIRITS [RFC3910] PSTN-related extensions with alcohol references, SPIRITS (Services
defines the inverse of PINT. It allows a switch in the PSTN to in PSTN Requesting Internet Services) [RFC3910] defines the
ask an IP element about how to proceed with call waiting. It was inverse of PINT. It allows a switch in the PSTN to ask an IP
developed primarily to support Internet Call Waiting (ICW). element how to proceed with call waiting. It was developed
Perhaps the next specification will be called the Pan Galactic primarily to support Internet Call Waiting (ICW). Perhaps the
Gargle Blaster [HGTTG]. next specification will be called the Pan Galactic Gargle Blaster
[HGTTG].
RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I): RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I):
SIP-T [RFC3372] defines a mechanism for using SIP between pairs of SIP-T [RFC3372] defines a mechanism for using SIP between pairs of
PSTN gateways. Its essential idea is to tunnel ISUP signaling PSTN gateways. Its essential idea is to tunnel ISDN User Part
between the gateways in the body of SIP messages. SIP-T motivated (ISUP) signaling between the gateways in the body of SIP messages.
the development of INFO [RFC2976]. SIP-T has seen widespread SIP-T motivated the development of INFO [RFC2976]. SIP-T has seen
implementation for the limited deployment model that it addresses. widespread implementation for the limited deployment model that it
As ISUP endpoints disappear from the network, the need for this addresses. As ISUP endpoints disappear from the network, the need
mechanism will decrease. for this mechanism will decrease.
RFC 3398, ISUP to SIP Mapping (S): [RFC3398] defines how to do RFC 3398, ISUP to SIP Mapping (S): [RFC3398] defines how to do
protocol mapping from the SS7 ISDN User Part (ISUP) signaling to 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. It is widely used in SS7 to SIP gateways and is part of the
SIP-T framework. SIP-T framework.
RFC 4497, Interworking between the Session Initiation Protocol (SIP) RFC 4497, Interworking between the Session Initiation Protocol (SIP)
and QSIG (B): [RFC4497] defines how to do protocol mapping from and QSIG (B): [RFC4497] defines how to do protocol mapping from
Q.SIG, used for PBX signaling, to SIP. Q.SIG, used for Private Branch Exchange (PBX) signaling, to SIP.
RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): [RFC3578] RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): [RFC3578]
defines a mechanism to map overlap dialing into SIP. This defines a mechanism to map overlap dialing into SIP. This
specification is widely regarded as the ugliest SIP specification, specification is widely regarded as the ugliest SIP specification,
as the introduction to the specification itself advises that it as the introduction to the specification itself advises that it
has many problems. Overlap signaling (the practice of sending has many problems. Overlap signaling (the practice of sending
digits into the network as dialed instead of waiting for complete digits into the network as dialed instead of waiting for complete
collection of the called party number) is largely incompatible collection of the called party number) is largely incompatible
with SIP at some fairly fundamental levels. That said, RFC 3578 with SIP at some fairly fundamental levels. That said, RFC 3578
is mostly harmless and has seen some usage. is mostly harmless and has seen some usage.
RFC 3960, Early Media and Ringtone Generation in SIP (I): [RFC3960] RFC 3960, Early Media and Ringtone Generation in SIP (I): [RFC3960]
defines some guidelines for handling early media - the practice of defines some guidelines for handling early media -- the practice
sending media from the called party or an application server of sending media from the called party or an application server
towards the caller - prior to acceptance of the call. Early media towards the caller prior to acceptance of the call. Early media
is often generated from the PSTN. Early media is a complex topic, is often generated from the PSTN. Early media is a complex topic,
and this specification does not fully address the problems and this specification does not fully address the problems
associated with it. associated with it.
RFC 3959, Early Session Disposition Type for the Session Initiation RFC 3959, Early Session Disposition Type for the Session Initiation
Protocol (SIP) (S): [RFC3959] defines a new session disposition type Protocol (SIP) (S): [RFC3959] defines a new session disposition
for use with early media. It indicates that the SDP in the body type for use with early media. It indicates that the SDP in the
is for a special early media session. This has seen little usage. body is for a special early media session. This has seen little
usage.
RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): [RFC3204] RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): [RFC3204]
defines MIME objects for representing SS7 and QSIG signaling defines MIME objects for representing SS7 and QSIG signaling
messages. SS7 signaling messages are carried in the body of SIP messages. SS7 signaling messages are carried in the body of SIP
messages when SIP-T is used. QSIG signaling messages can be messages when SIP-T is used. QSIG signaling messages can be
carried in a similar way. carried in a similar way.
RFC3666, Session Initiation Protocol (SIP) Public Switched Telephone RFC3666, Session Initiation Protocol (SIP) Public Switched Telephone
Network (PSTN) Call Flows (B): [RFC3666] provides best practice call Network (PSTN) Call Flows (B): [RFC3666] provides best practice
flows around interworking with the PSTN. call flows around interworking with the PSTN.
5. General Purpose Infrastructure Extensions 5. General Purpose Infrastructure Extensions
These extensions are general purpose enhancements to SIP, SDP and These extensions are general purpose enhancements to SIP, SDP, and
MIME that can serve a wide variety of uses. However, they are not MIME that can serve a wide variety of uses. However, they are not
used for every session or registration, as the core specifications used for every session or registration, as the core specifications
are. are.
RFC 3262, Reliability of Provisional Responses in SIP (S): SIP RFC 3262, Reliability of Provisional Responses in SIP (S): SIP
defines two types of responses to a request - final and defines two types of responses to a request: final and
provisional. Provisional responses are numbered from 100 to 199. provisional. Provisional responses are numbered from 100 to 199.
In SIP, these responses are not sent reliably. This choice was In SIP, these responses are not sent reliably. This choice was
made in RFC 2543 since the messages were meant to just be truly made in RFC 2543 since the messages were meant to just be truly
informational, and rendered to the user. However, subsequent work informational and rendered to the user. However, subsequent work
on PSTN interworking demonstrated a need to map provisional on PSTN interworking demonstrated a need to map provisional
responses to PSTN messages that needed to be sent reliably. responses to PSTN messages that needed to be sent reliably.
[RFC3262] was developed to allow reliability of provisional [RFC3262] was developed to allow reliability of provisional
responses. The specification defines the PRACK method, used for responses. The specification defines the PRACK method, used for
indicating that a provisional response was received. Though it indicating that a provisional response was received. Though it
provides a generic capability for SIP, RFC 3262 implementations provides a generic capability for SIP, RFC 3262 implementations
have been most common in PSTN interworking devices. However, have been most common in PSTN interworking devices. However,
PRACK brings a great deal of complication for relatively small PRACK brings a great deal of complication for relatively small
benefit. As such, it has seen only moderate levels of deployment. benefit. As such, it has seen only moderate levels of deployment.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol RFC 3323, A Privacy Mechanism for the Session Initiation Protocol
(SIP) (S): [RFC3323] defines the Privacy header field, used by (SIP) (S): [RFC3323] defines the Privacy header field, used by
clients to request anonymity for their requests. Though it clients to request anonymity for their requests. Though it
defines several privacy services, the only one broadly used is the defines several privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-Identity header field one that supports privacy of the P-Asserted-Identity header field
[RFC3325]. [RFC3325].
draft-ietf-sip-ua-privacy, UA-Driven Privacy Mechanism for SIP (S): UA-PRIVACY, UA-Driven Privacy Mechanism for SIP (S): [UA-PRIVACY]
[I-D.ietf-sip-ua-privacy] defines a mechanism for achieving defines a mechanism for achieving anonymous calls in SIP. It is
anonymous calls in SIP. It is an alternative to [RFC3323], and an alternative to [RFC3323], and instead places more intelligence
instead places more intelligence in the endpoint to craft in the endpoint to craft anonymous messages by directly accessing
anonymous messages by directly accessing network services. network services.
RFC 2976, The INFO Method (S): [RFC2976] was defined as an extension RFC 2976, The INFO Method (S): [RFC2976] was defined as an extension
to RFC 2543. It defines a method, INFO, used to transport mid- to RFC 2543. It defines a method, INFO, used to transport mid-
dialog information that has no impact on SIP itself. Its driving dialog information that has no impact on SIP itself. Its driving
application was the transport of PSTN related information when application was the transport of PSTN-related information when
using SIP between a pair of gateways. Though originally conceived using SIP between a pair of gateways. Though originally conceived
for broader use, it only found standardized usage with SIP-T for broader use, it only found standardized usage with SIP-T
[RFC3372]. It has been used to support numerous proprietary and [RFC3372]. It has been used to support numerous proprietary and
non-interoperable extensions due to its poorly defined scope. non-interoperable extensions due to its poorly defined scope.
RFC 3326, The Reason header field for SIP (S): [RFC3326] defines the RFC 3326, The Reason Header Field for SIP (S): [RFC3326] defines the
Reason header field. It is used in requests, such as BYE, to Reason header field. It is used in requests, such as BYE, to
indicate the reason that the request is being sent. indicate the reason that the request is being sent.
RFC 3388, Grouping of Media Lines in the Session Description Protocol RFC 3388, Grouping of Media Lines in the Session Description Protocol
(S): RFC 3388 [RFC3388] defines a framework for grouping together (S): RFC 3388 [RFC3388] defines a framework for grouping together
media streams in an SDP message. Such a grouping allows media streams in an SDP message. Such a grouping allows
relationships between these streams, such as which stream is the relationships between these streams, such as which stream is the
audio for a particular video feed, to be expressed. audio for a particular video feed, to be expressed.
RFC 3420, Internet Media Type message/sipfrag (S): [RFC3420] defines RFC 3420, Internet Media Type message/sipfrag (S): [RFC3420] defines
a MIME object that contains a SIP message fragment. Only certain a MIME object that contains a SIP message fragment. Only certain
header fields and parts of the SIP message are present. For header fields and parts of the SIP message are present. For
example, it is used to report back on the responses received to a example, it is used to report back on the responses received to a
request sent as a consequence of a REFER. request sent as a consequence of a REFER.
RFC 3608, SIP Extension Header Field for Service Route Discovery RFC 3608, SIP Extension Header Field for Service Route Discovery
During Registration (S): [RFC3608] allows a client to determine, During Registration (S): [RFC3608] allows a client to determine,
from a REGISTER response, a path of proxies to use in requests it from a REGISTER response, a path of proxies to use in requests it
sends outside of a dialog. It can also be used by proxies to sends outside of a dialog. It can also be used by proxies to
verify the Route header in client initiated requests. In many verify the Route header in client-initiated requests. In many
respects, it is the inverse of the Path header field, but has seen respects, it is the inverse of the Path header field, but has seen
less usage since default outbound proxies have been sufficient in less usage since default outbound proxies have been sufficient in
many deployments. many deployments.
RFC 3841, Caller Preferences for SIP (S): [RFC3841] defines a set of RFC 3841, Caller Preferences for SIP (S): [RFC3841] defines a set of
headers that a client can include in a request to control the way 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 in which the request is routed downstream. It allows a client to
direct a request towards a UA with specific capabilities, which a direct a request towards a UA with specific capabilities, which a
UA indicates using [RFC3840]. UA indicates using [RFC3840].
RFC 4028, Session Timers in SIP (S): [RFC4028] defines a keepalive RFC 4028, Session Timers in SIP (S): [RFC4028] defines a keepalive
mechanism for SIP signaling. It is primarily meant to provide a mechanism for SIP signaling. It is primarily meant to provide a
way to cleanup old state in proxies that are holding call state way to cleanup old state in proxies that are holding call state
for calls from failed endpoints which were never terminated for calls from failed endpoints that were never terminated
normally. Despite its name, the session timer is not a mechanism normally. Despite its name, the session timer is not a mechanism
for detecting a network failure mid-call. Session timers for detecting a network failure mid-call. Session timers
introduces a fair bit of complexity for relatively little gain, introduce a fair bit of complexity for relatively little gain, and
and have seen moderate deployment. have seen moderate deployment.
RFC 4168, SCTP as a Transport for SIP (S): [RFC4168] defines how to RFC 4168, SCTP as a Transport for SIP (S): [RFC4168] defines how to
carry SIP messages over the Stream Control Transmission Protocol carry SIP messages over the Stream Control Transmission Protocol
(SCTP) [RFC4960]. SCTP has seen very limited usage for SIP (SCTP) [RFC4960]. SCTP has seen very limited usage for SIP
transport. transport.
RFC 4244, An Extension to SIP for Request History Information (S): RFC 4244, An Extension to SIP for Request History Information (S):
[RFC4244] defines the History-Info header field, which indicates [RFC4244] defines the History-Info header field, which indicates
information on how and why a call came to be routed to a information on how and why a call came to be routed to a
particular destination. particular destination.
RFC 4145, TCP-Based Media Transport in the Session Description RFC 4145, TCP-Based Media Transport in the Session Description
Protocol (SDP) (S): [RFC4145] defines an extension to SDP for Protocol (SDP) (S): [RFC4145] defines an extension to SDP for
setting up TCP-based sessions between user agents. It defines who setting up TCP-based sessions between user agents. It defines who
sets up the connection and how its lifecycle is managed. It has sets up the connection and how its lifecycle is managed. It has
seen relatively little usage due to the small number of media seen relatively little usage due to the small number of media
types to date which use TCP. types to date that use TCP.
RFC 4091, The Alternative Network Address Types (ANAT) Semantics for RFC 4091, The Alternative Network Address Types (ANAT) Semantics for
the Session Description Protocol (SDP) Grouping Framework (S): the Session Description Protocol (SDP) Grouping Framework (S):
[RFC4091] defines a mechanism for including both IPv4 and IPv6 [RFC4091] defines a mechanism for including both IPv4 and IPv6
addresses for a media session as alternates. This mechanism has addresses to establish a media stream. This mechanism has been
been deprecated in favor of ICE [I-D.ietf-mmusic-ice]. deprecated in favor of ICE [ICE].
draft-ietf-mmusic-sdp-media-capabilities, SDP Media Capabilities SDP-MEDIA, SDP Media Capabilities Negotiation (S): [SDP-MEDIA]
Negotiation (S): [I-D.ietf-mmusic-sdp-media-capabilities] defines an defines an extension to the SDP capability negotiation framework
extension to the SDP capability negotiation framework [SDP-CAP] for negotiating codecs, codec parameters, and media
[I-D.ietf-mmusic-sdp-capability-negotiation] for negotiating streams.
codecs, codec parameters, and media streams.
draft-ietf-sip-body-handling, Message Body Handling in the Session BODY-HANDLING, Message Body Handling in the Session Initiation
Initiation Protocol (SIP): [I-D.ietf-sip-body-handling] clarifies Protocol (SIP): [BODY-HANDLING] clarifies handling of bodies in
handling of bodies in SIP, focusing primarily on multi-part SIP, focusing primarily on multi-part behavior, which was under-
behavior, which was underspecified in SIP. specified in SIP.
6. NAT Traversal 6. NAT Traversal
These SIP extensions are primarily aimed at addressing NAT traversal These SIP extensions are primarily aimed at addressing NAT traversal
for SIP. for SIP.
draft-ietf-mmusic-ice, Interactive Connectivity Establishment (ICE) ICE, Interactive Connectivity Establishment (ICE) (S): [ICE] defines
(S): [I-D.ietf-mmusic-ice] defines a technique for NAT traversal of a technique for NAT traversal of media sessions for protocols that
media sessions for protocols that make use of the offer/answer make use of the offer/answer model. This specification is the
model. This specification is the IETF recommended mechanism for IETF-recommended mechanism for NAT traversal for SIP media
NAT traversal for SIP media streams, and is meant to be used even streams, and is meant to be used even by endpoints that are
by endpoints which are themselves never behind a NAT. A SIP themselves never behind a NAT. A SIP option tag and media feature
option tag and media feature tag [I-D.ietf-sip-ice-option-tag] tag [OPTION-TAG] have been defined for use with ICE.
have been defined for use with ICE.
draft-ietf-mmusic-ice-tcp, TCP Candidates with Interactive ICE-TCP, TCP Candidates with Interactive Connectivity Establishment
Connectivity Establishment (ICE) (S): [I-D.ietf-mmusic-ice-tcp] (ICE) (S): [ICE-TCP] specifies the usage of ICE for TCP streams.
specifies the usage of ICE for TCP streams. This allows for This allows for selection of RTP-based voice on top of TCP only
selection of RTP-based voice ontop of TCP only when NAT or when NAT or firewalls would prevent UDP-based voice from working.
firewalls would prevent UDP-based voice from working.
RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
Description Protocol (SDP) (S): [RFC3605] defines a way to Description Protocol (SDP) (S): [RFC3605] defines a way to
explicitly signal, within an SDP message, the IP address and port explicitly signal, within an SDP message, the IP address and port
for RTCP, rather than using the port+1 rule in the Real Time for RTCP, rather than using the port+1 rule in the Real Time
Transport Protocol (RTP) [RFC3550]. It is needed for devices Transport Protocol (RTP) [RFC3550]. It is needed for devices
behind NAT and used by ICE. behind NAT, and the specification is required by ICE.
draft-ietf-sip-outbound, Managing Client Initiated Connections OUTBOUND, Managing Client Initiated Connections through SIP (S):
through SIP (S): [I-D.ietf-sip-outbound], also known as SIP [OUTBOUND], also known as SIP outbound, defines important changes
outbound, defines important changes to the SIP registration to the SIP registration mechanism that enable delivery of SIP
mechanism which enable delivery of SIP messages towards a UA when messages towards a UA when it is behind a NAT.
it is behind a NAT.
RFC 3581, An Extension to SIP for Symmetric Response Routing (S): RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
[RFC3581] defines the rport parameter of the Via header. It [RFC3581] defines the rport parameter of the Via header. It
allows SIP responses to traverse NAT. allows SIP responses to traverse NAT.
draft-ietf-sip-gruu, Obtaining and Using Globally Routable User Agent GRUU, Obtaining and Using Globally Routable User Agent Identifiers
Identifiers (GRUU) in SIP (S): [I-D.ietf-sip-gruu] defines a (GRUU) in SIP (S): [GRUU] defines a mechanism for directing
mechanism for directing requests towards a specific UA instance. requests towards a specific UA instance. GRUU is essential for
GRUU is essential for features like transfer and provides another features like transfer and provides another piece of the SIP NAT
piece of the SIP NAT traversal story. traversal story.
7. Call Control Primitives 7. Call Control Primitives
Numerous SIP extensions provide a toolkit of dialog and call Numerous SIP extensions provide a toolkit of dialog- and call-
management techniques. These techniques have been combined together management techniques. These techniques have been combined together
to build many SIP-based services. to build many SIP-based services.
RFC 3515, The REFER Method (S): REFER [RFC3515] defines a mechanism RFC 3515, The REFER Method (S): REFER [RFC3515] defines a mechanism
for asking a user agent to send a SIP request. It's a form of SIP 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 remote control, and is the primary tool used for call transfer in
SIP. Beware that not all potential uses of REFER (neither for all SIP. Beware that not all potential uses of REFER (neither for all
methods nor for all URI schemes) are well defined. Implementors methods nor for all URI schemes) are well defined. Implementors
should only use the well-defined ones, and should not second guess should only use the well-defined ones, and should not second guess
or freely assume behavior for the others to avoid unexpected or freely assume behavior for the others to avoid unexpected
behavior of remote UAs, interoperability issues, and other bad behavior of remote UAs, interoperability issues, and other bad
surprises. surprises.
RFC 3725, Best Current Practices for Third Party Call Control (3pcc) RFC 3725, Best Current Practices for Third Party Call Control (3pcc)
(B): [RFC3725] defines a number of different call flows that allow (B): [RFC3725] defines a number of different call flows that
one SIP entity, called the controller, to create SIP sessions allow one SIP entity, called the controller, to create SIP
amongst other SIP user agents. sessions amongst other SIP user agents.
RFC 3911, The SIP Join Header Field (S): [RFC3911] defines the Join RFC 3911, The SIP Join Header Field (S): [RFC3911] defines the Join
header field. When sent in an INVITE, it causes the recipient to header field. When sent in an INVITE, it causes the recipient to
join the resulting dialog into a conference with another dialog in join the resulting dialog into a conference with another dialog in
progress. progress.
RFC 3891, The SIP Replaces Header (S): [RFC3891] defines a mechanism RFC 3891, The SIP Replaces Header (S): [RFC3891] defines a mechanism
that allows a new dialog to replace an existing dialog. It is that allows a new dialog to replace an existing dialog. It is
useful for certain advanced transfer services. useful for certain advanced transfer services.
RFC 3892, The SIP Referred-By Mechanism (S): [RFC3892] defines the RFC 3892, The SIP Referred-By Mechanism (S): [RFC3892] defines the
Referred-By header field. It is used in requests triggered by Referred-By header field. It is used in requests triggered by
REFER, and provides the identity of the referring party to the REFER, and provides the identity of the referring party to the
referred-to party. referred-to party.
RFC 4117, Transcoding Services Invocation in SIP Using Third Party RFC 4117, Transcoding Services Invocation in SIP Using Third Party
Call Control (I): [RFC4117] defines how to use 3pcc for the purposes Call Control (I): [RFC4117] defines how to use 3pcc for the
of invoking transcoding services for a call. purposes of invoking transcoding services for a call.
8. Event Framework 8. Event Framework
RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the
SUBSCRIBE and NOTIFY methods. These two methods provide a general SUBSCRIBE and NOTIFY methods. These two methods provide a general
event notification framework for SIP. To actually use the event notification framework for SIP. To actually use the
framework, extensions need to be defined for specific event framework, extensions need to be defined for specific event
packages. An event package defines a schema for the event data, packages. An event package defines a schema for the event data
and describes other aspects of event processing specific to that and describes other aspects of event processing specific to that
schema. An RFC 3265 implementation is required when any event schema. An RFC 3265 implementation is required when any event
package is used. package is used.
RFC 3903, SIP Extension for Event State Publication (S): [RFC3903] RFC 3903, SIP Extension for Event State Publication (S): [RFC3903]
defines the PUBLISH method. It is not an event package, but is defines the PUBLISH method. It is not an event package, but is
used by all event packages as a mechanism for pushing an event used by all event packages as a mechanism for pushing an event
into the system. into the system.
RFC 4662, A Session Initiation Protocol (SIP) Event Notification RFC 4662, A Session Initiation Protocol (SIP) Event Notification
Extension for Resource Lists (S): [RFC4662] defines an extension to Extension for Resource Lists (S): [RFC4662] defines an extension
RFC 3265 that allows a client to subscribe to a list of resources to RFC 3265 that allows a client to subscribe to a list of
using a single subscription. The server, called a Resource List resources using a single subscription. The server, called a
Server (RLS) will "expand" the subscription and subscribe to each Resource List Server (RLS), will "expand" the subscription and
individual member of the list. It has found applicability subscribe to each individual member of the list. It has found
primarily in the area of presence, but can be used with any event applicability primarily in the area of presence, but can be used
package. with any event package.
draft-ietf-sip-subnot-etags, An Extension to Session Initiation SUBNOT-ETAGS, An Extension to Session Initiation Protocol (SIP)
Protocol (SIP) Events for Conditional Event Notification (S): Events for Conditional Event Notification (S): [SUBNOT-ETAGS]
[I-D.ietf-sip-subnot-etags] defines an extension to RFC 3265 to defines an extension to RFC 3265 to optimize the performance of
optimize the performance of notifications. When a client notifications. When a client subscribes, it can indicate what
subscribes, it can indicate what version of a document it has, so version of a document it has so that the server can skip sending a
that the server can skip sending a notification if the client is notification if the client is up-to-date. It is applicable to any
up to date. It is applicable to any event package. event package.
9. Event Packages 9. Event Packages
These are event packages defined to utilize the SIP events framework. These are event packages defined to utilize the SIP events framework.
Many of these are also listed elsewhere in their respective areas. Many of these are also listed elsewhere in their respective areas.
RFC 3680, A SIP Event Package for Registrations (S): [RFC3680] RFC 3680, A SIP Event Package for Registrations (S): [RFC3680]
defines an event package for finding out about changes in defines an event package for finding out about changes in
registration state. registration state.
draft-ietf-sipping-gruu-reg-event (S): GRUU-REG (S): [GRUU-REG] is an extension to the registration event
[I-D.ietf-sipping-gruu-reg-event] is an extension to the package [RFC3680] that allows user agents to learn about their
registration event package [RFC3680] that allows user agents to GRUUs. It is particularly useful in helping to synchronize a
learn about their GRUUs. It is particularly useful in helping to client and its registrar with their currently valid temporary
synchronize a client and its registrar with its currently valid GRUU.
temporary GRUU.
RFC 3842, A Message Summary and Message Waiting Indication Event RFC 3842, A Message Summary and Message Waiting Indication Event
Package for SIP (S): [RFC3482] defines a way for a user agent to Package for SIP (S): [RFC3842] defines a way for a user agent to
find out about voicemails and other messages that are waiting for find out about voicemails and other messages that are waiting for
it. Its primary purpose is to enable the voicemail waiting lamp it. Its primary purpose is to enable the voicemail waiting lamp
on most business telephones. on most business telephones.
RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an
event package for indicating user presence through SIP. event package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S): RFC 3857, A Watcher Information Event Template Package for SIP (S):
[RFC3857], also known as winfo, provides a mechanism for a user [RFC3857], also known as winfo, provides a mechanism for a user
agent to find out what subscriptions are in place for a particular agent to find out what subscriptions are in place for a particular
event package. Its primary usage is with presence, but it can be event package. Its primary usage is with presence, but it can be
used with any event package. used with any event package.
RFC 4235, An INVITE Initiated Dialog Event Package for SIP (S): RFC 4235, An INVITE-Initiated Dialog Event Package for SIP (S):
[RFC4235] defines an event package for learning the state of the [RFC4235] defines an event package for learning the state of the
dialogs in progress at a user agent, and is one of several RFCs dialogs in progress at a user agent, and is one of several RFCs
starting with the important number 42 [HGTTG]. starting with the important number 42 [HGTTG].
RFC 4575, A SIP Event Package for Conference State (S): [RFC4575] RFC 4575, A SIP Event Package for Conference State (S): [RFC4575]
defines a mechanism for learning about changes in conference defines a mechanism for learning about changes in conference
state, including conference membership. state, including conference membership.
RFC 4730, A SIP Event Package for Keypress Stimulus (KPML) (S): RFC 4730, A SIP Event Package for Key Press Stimulus (KPML) (S):
[RFC4730] defines a way for an application in the network to [RFC4730] defines a way for an application in the network to
subscribe to the set of keypresses made on the keypad of a subscribe to the set of keypresses made on the keypad of a
traditional telephone. It, along with RFC 2833 [RFC2833], are the traditional telephone. It, along with RFC 4733 [RFC4733], are the
two mechanisms defined for handling DTMF. RFC 4730 is a two mechanisms defined for handling DTMF. RFC 4730 is a
signaling-path solution, and RFC 2833 is a media-path solution. signaling-path solution, and RFC 4733 is a media-path solution.
draft-ietf-sipping-rtcp-summary, SIP Event Package for Voice Quality RTCP-SUM, SIP Event Package for Voice Quality Reporting (S):
Reporting (S): [I-D.ietf-sipping-rtcp-summary] defines a SIP event [RTCP-SUM] defines a SIP event package that enables the collection
package that enables the collection and reporting of metrics that and reporting of metrics that measure the quality for Voice over
measure the quality for Voice over Internet Protocol (VoIP) Internet Protocol (VoIP) sessions.
sessions.
draft-ietf-sip-session-policy-framework, A Framework for Session SESSION-POLICY, A Framework for Session Initiation Protocol (SIP)
Initiation Protocol (SIP) Session Policies (S): Session Policies (S): [SESSION-POLICY] defines a framework for
[I-D.ietf-sip-session-policy-framework] defines a framework for
session policies. In this framework, policy servers are used to session policies. In this framework, policy servers are used to
tell user agents about the media characteristics required for a tell user agents about the media characteristics required for a
particular session. The session policy framework has not been particular session. The session policy framework has not been
widely implemented. widely implemented.
draft-ietf-sipping-policy-package, A Session Initiation Protocol POLICY-PACK, A Session Initiation Protocol (SIP) Event Package for
(SIP) Event Package for Session-Specific Session Policies (S): Session-Specific Session Policies (S): [POLICY-PACK] defines a
[I-D.ietf-sipping-policy-package] defines a SIP event package used SIP event package used in conjunction with the session policy
in conjunction with the session policy framework framework [SESSION-POLICY].
[I-D.ietf-sip-session-policy-framework].
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions RFC 5362, The Session Initiation Protocol (SIP) Pending Additions
Event Package (S): [RFC5362] defines a SIP event package that allows Event Package (S): [RFC5362] defines a SIP event package that
a UA to learn whether consent has been given for the addition of allows a UA to learn whether consent has been given for the
an address to a SIP "mailing list". It is used in conjunction addition of an address to a SIP "mailing list". It is used in
with the SIP framework for consent [RFC5360]. conjunction with the SIP framework for consent [RFC5360].
10. Quality of Service 10. Quality of Service
Several specifications concern themselves with the interactions of Several specifications concern themselves with the interactions of
SIP with network Quality of Service (QoS) mechanisms. SIP with network Quality of Service (QoS) mechanisms.
RFC 3312, Integration of Resource Management and SIP (S): [RFC3312], RFC 3312, Integration of Resource Management and SIP (S): [RFC3312],
updated by [RFC4032] defines a way to make sure that the phone of updated by [RFC4032], defines a way to make sure that the phone of
the called party doesn't ring until a QoS reservation has been the called party doesn't ring until a QoS reservation has been
installed in the network. It does so by defining a general installed in the network. It does so by defining a general
preconditions framework, which defines conditions that must be preconditions framework, which defines conditions that must be
true in order for a SIP session to proceed. true in order for a SIP session to proceed.
draft-ietf-mmusic-qos-identification, Quality of Service (QoS) QoS-ID, Quality of Service (QoS) Mechanism Selection in the Session
Mechanism Selection in the Session Description Protocol (SDP) (S): Description Protocol (SDP) (S): [QoS-ID] defines a way for user
[I-D.ietf-mmusic-qos-identification] defines a way for user agents agents to negotiate what type of end-to-end QoS mechanism to use
to negotiate what type of end-to-end QoS mechanism to use for a for a session. At this time, there are two that can be used: the
session. At this time, there are two that can be used - RSVP and Resource Reservation Protocol (RSVP) and Next Steps in Signaling
NSIS. This negotiation is done through an SDP extension. Due to (NSIS). This negotiation is done through an SDP extension. Due
limited deployment of RSVP and even more limited deployment of to limited deployment of RSVP and even more limited deployment of
NSIS, this extension has not been widely used. NSIS, this extension has not been widely used.
RFC 3313, Private SIP Extensions for Media Authorization (I): RFC 3313, Private SIP Extensions for Media Authorization (I):
[RFC3313] defines a P-header that provides a mechanism for passing [RFC3313] defines a P-header that provides a mechanism for passing
an authorization token between SIP and a network QoS reservation an authorization token between SIP and a network QoS reservation
protocol like RSVP. Its purpose is to make sure network QoS is 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 only granted if a client has made a SIP call through the same
providers network. This specification is sometimes referred to as provider's network. This specification is sometimes referred to
the SIP walled garden specification by the truly paranoid androids as the SIP walled-garden specification by the truly paranoid
in the SIP community. This is because it requires coupling of androids in the SIP community. This is because it requires
signaling and the underlying IP network. coupling of signaling and the underlying IP network.
RFC 3524, Mapping of Media Streams to Resource Reservation Flows RFC 3524, Mapping of Media Streams to Resource Reservation Flows
(S): [RFC3524] defines a usage of the SDP grouping framework for (S): [RFC3524] defines a usage of the SDP grouping framework for
indicating that a set of media streams should be handled by a indicating that a set of media streams should be handled by a
single resource reservation. single resource reservation.
11. Operations and Management 11. Operations and Management
Several specifications have been defined to support operations and Several specifications have been defined to support operations and
management of SIP systems. These include mechanisms for management of SIP systems. These include mechanisms for
configuration and network diagnostics. configuration and network diagnostics.
draft-ietf-sipping-config-framework, A Framework for SIP User Agent CONFIG-FRAME, A Framework for SIP User Agent Profile Delivery (S):
Profile Delivery (S): [I-D.ietf-sipping-config-framework] defines a [CONFIG-FRAME] defines a mechanism that allows a SIP user agent to
mechanism that allows a SIP user agent to bootstrap its bootstrap its configuration from the network and receive updates
configuration from the network, and receive updates to its to its configuration, should it change. This is considered an
configuration should it change. This is considered an essential essential piece of deploying a usable SIP network.
piece of deploying a usable SIP network.
draft-ietf-sipping-rtcp-summary, SIP Event Package for Voice Quality RTCP-SUM, SIP Event Package for Voice Quality Reporting (S):
Reporting (S): [I-D.ietf-sipping-rtcp-summary] defines a SIP event [RTCP-SUM] defines a SIP event package that enables the collection
package that enables the collection and reporting of metrics that and reporting of metrics that measure the quality for Voice over
measure the quality for Voice over Internet Protocol (VoIP) Internet Protocol (VoIP) sessions.
sessions.
12. SIP Compression 12. SIP Compression
Sigcomp [RFC3320] [RFC4896] was defined to allow compression of SIP Sigcomp [RFC3320] [RFC4896] was defined to allow compression of SIP
messages over low bandwidth links. Sigcomp is not formally part of messages over low bandwidth links. Sigcomp is not formally part of
SIP. However, usage of Sigcomp with SIP has required extensions to SIP. However, usage of Sigcomp with SIP has required extensions to
SIP. SIP.
RFC 3486, Compressing SIP (S): [RFC3486] defines a SIP URI parameter RFC 3486, Compressing SIP (S): [RFC3486] defines a SIP URI parameter
that can be used to indicate that a SIP server supports Sigcomp. that can be used to indicate that a SIP server supports Sigcomp.
RFC 5049, Applying Signaling Compression (SigComp) to the Session RFC 5049, Applying Signaling Compression (SigComp) to the Session
Initiation Protocol (SIP) (S): [RFC5049] defines how to apply Initiation Protocol (SIP) (S): [RFC5049] defines how to apply
Sigcomp to SIP. Sigcomp to SIP.
13. SIP Service URIs 13. SIP Service URIs
Several extensions define well-known services that can be invoked by Several extensions define well-known services that can be invoked by
constructing requests with the specific structures for the Request constructing requests with specific structures for the Request URI,
URI, resulting in specific behaviors at the UAS. resulting in specific behaviors at the User Agent Server (UAS).
RFC 3087, Control of Service Context using Request URI (I): RFC 3087, Control of Service Context using Request URI (I):
[RFC3087] introduced the context of using Request URIs, encoded [RFC3087] introduced the context of using Request URIs, encoded
appropriately, to invoke services. appropriately, to invoke services.
RFC 4662, A SIP Event Notification Extension for Resource Lists (S): RFC 4662, A SIP Event Notification Extension for Resource Lists (S):
[RFC4662] defines a resource called a Resource List Server. A [RFC4662] defines a resource called a Resource List Server (RLS).
client can send a subscribe to this server. The server will A client can send a subscribe to this server. The server will
generate a series of subscriptions, and compile the resulting generate a series of subscriptions, compile the resulting
information and send it back to the subscriber. The set of information, and send it back to the subscriber. The set of
resources that the RLS will subscribe to is a property of the resources that the RLS will subscribe to is a property of the
request URI in the SUBSCRIBE request. request URI in the SUBSCRIBE request.
RFC 5363, Framework and Security Considerations for Session RFC 5363, Framework and Security Considerations for Session
Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List
Services (S): [RFC5363] defines the framework for list services in Services (S): [RFC5363] defines the framework for list services
SIP. In this framework, a UA can include an XML list object in in SIP. In this framework, a UA can include an XML list object in
the body of various requests and the server provides list-oriented the body of various requests and the server will provide list-
services as a consequence. For example, a SUBSCRIBE with a list oriented services as a consequence. For example, a SUBSCRIBE with
subscribes to the URI in the list. a list subscribes to the URI in the list.
RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP
(S): [RFC5367] uses the URI list framework [RFC5363] and allows a (S): [RFC5367] uses the URI-list framework [RFC5363] and allows a
client to subscribe to a resource called a Resource List Server. client to subscribe to a resource called a Resource List Server.
This server will generate subscriptions to the URI in the list, This server will generate subscriptions to the URI in the list,
and compile the resulting information and send it back to the compile the resulting information, and send it back to the
subscriber. subscriber.
RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S): [RFC5365] RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S): [RFC5365]
uses the URI list framework [RFC5363] and allows a client to send uses the URI-list framework [RFC5363] and allows a client to send
a MESSAGE to a number of recipients. a MESSAGE to a number of recipients.
RFC 5366, Conference Establishment Using Request-Contained Lists in RFC 5366, Conference Establishment Using Request-Contained Lists in
SIP (S): [RFC5366] uses the URI list framework [RFC5363] and allows SIP (S): [RFC5366] uses the URI-list framework [RFC5363]. It
a client to ask the server to act as a conference focus and send allows a client to ask the server to act as a conference focus and
an invitation to each recipient in the list. send an invitation to each recipient in the list.
RFC 4240, Basic Network Media Services with SIP (I): [RFC4240] RFC 4240, Basic Network Media Services with SIP (I): [RFC4240]
defines a way for SIP application servers to invoke announcement defines a way for SIP application servers to invoke announcement
and conferencing services from a media server. This is and conferencing services from a media server. This is
accomplished through a set of defined URI parameters which tell accomplished through a set of defined URI parameters that tell the
the media server what to do, such as what file to play and what media server what to do, such as what file to play and what
language to render it in. language to render it in.
RFC 4458, Session Initiation Protocol (SIP) URIs for Applications RFC 4458, Session Initiation Protocol (SIP) URIs for Applications
such as Voicemail and Interactive Voice Response (IVR) (I): such as Voicemail and Interactive Voice Response (IVR) (I):
[RFC4458] defines a way to invoke voicemail and IVR services by [RFC4458] defines a way to invoke voicemail and IVR services by
using a SIP URI constructed in a particular way. using a SIP URI constructed in a particular way.
14. Minor Extensions 14. Minor Extensions
These SIP extensions don't fit easily into a single specific use These SIP extensions don't fit easily into a single specific use
skipping to change at page 20, line 40 skipping to change at page 20, line 19
relatively small problem or provide an optimization. relatively small problem or provide an optimization.
RFC 4488, Suppression of the SIP REFER Implicit Subscription (S): RFC 4488, Suppression of the SIP REFER Implicit Subscription (S):
[RFC4488] defines an enhancement to REFER. REFER normally creates [RFC4488] defines an enhancement to REFER. REFER normally creates
an implicit subscription to the target of the REFER. This an implicit subscription to the target of the REFER. This
subscription is used to pass back updates on the progress of the subscription is used to pass back updates on the progress of the
referral. This extension allows that implicit subscription to be referral. This extension allows that implicit subscription to be
bypassed as an optimization. bypassed as an optimization.
RFC 4538, Request Authorization through Dialog Identification in SIP RFC 4538, Request Authorization through Dialog Identification in SIP
(S): [RFC4538] provides a mechanism that allows a UAS to authorize a (S): [RFC4538] provides a mechanism that allows a UAS to
request because the requestor proves it knows a dialog that is in authorize a request because the requestor proves it knows a dialog
progress with the UAS. The specification is useful in conjunction that is in progress with the UAS. The specification is useful in
with the SIP application interaction framework conjunction with the SIP application interaction framework
[I-D.ietf-sipping-app-interaction-framework]. [INTERACT-FRAME].
RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S): RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S):
[RFC4508] defines a mechanism for carrying RFC 3840 feature tags [RFC4508] defines a mechanism for carrying RFC 3840 feature tags
in REFER. It is useful for informing the target of the REFER in REFER. It is useful for informing the target of the REFER
about the characteristics of the intentended target of the about the characteristics of the intended target of the referred
referred request. request.
draft-ietf-sip-answermode, Requesting Answer Modes for SIP (S): RFC 5373, Requesting Answer Modes for SIP (S): [RFC5373] defines an
[I-D.ietf-sip-answermode] defines an extension for indicating to extension for indicating to the called party whether or not the
the called party whether or not the phone should ring and/or be phone should ring and/or be answered immediately. This is useful
answered immediately. This is useful for push-to-talk and for for push-to-talk and for diagnostic applications.
diagnostic applications.
RFC 5079, Rejecting Anonymous Requests in SIP (S): [RFC5079] defines RFC 5079, Rejecting Anonymous Requests in SIP (S): [RFC5079] defines
a mechanism for a called party to indicate to the calling party a mechanism for a called party to indicate to the calling party
that a call was rejected since the caller was anonymous. This is that a call was rejected since the caller was anonymous. This is
needed for implementation of the Anonymous Call Rejection (ACR) needed for implementation of the Anonymous Call Rejection (ACR)
feature in SIP. feature in SIP.
RFC 5368, Referring to Multiple Resources in SIP (S): [RFC5368] RFC 5368, Referring to Multiple Resources in SIP (S): [RFC5368]
allows a UA sending a REFER to ask the recipient of the REFER to 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 generate multiple SIP requests, not just one. This is useful for
skipping to change at page 21, line 32 skipping to change at page 21, line 9
RFC 4483, A Mechanism for Content Indirection in Session Initiation RFC 4483, A Mechanism for Content Indirection in Session Initiation
Protocol (SIP) Messages (S): [RFC4483] defines a mechanism for Protocol (SIP) Messages (S): [RFC4483] defines a mechanism for
content indirection. Instead of carrying an object within a SIP content indirection. Instead of carrying an object within a SIP
body, a URL reference is carried instead, and the recipient body, a URL reference is carried instead, and the recipient
dereferences the URL to obtain the object. The specification has dereferences the URL to obtain the object. The specification has
potential applicability for sending large instant messages, but potential applicability for sending large instant messages, but
has yet to find much actual use. has yet to find much actual use.
RFC 3890, A Transport Independent Bandwidth Modifier for the Session RFC 3890, A Transport Independent Bandwidth Modifier for the Session
Description Protocol (SDP) (S): [RFC3890] specifies an SDP extension Description Protocol (SDP) (S): [RFC3890] specifies an SDP
that allows for the description of the bandwidth for a media extension that allows for the description of the bandwidth for a
session that is independent of the underlying transport mechanism. media session that is independent of the underlying transport
mechanism.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
Control Protocol (BFCP) Streams (S): [RFC4583] defines a mechanism Control Protocol (BFCP) Streams (S): [RFC4583] defines a
in SDP to signal floor control streams that use BFCP. It is used mechanism in SDP to signal floor control streams that use BFCP.
for Push-To-Talk and conference floor control. It is used for push-to-talk and conference floor control.
draft-ietf-mmusic-connectivity-precon, Connectivity Preconditions for CONNECT-PRECON, Connectivity Preconditions for Session Description
Session Description Protocol Media Streams (S): Protocol Media Streams (S): [CONNECT-PRECON] defines a usage of
[I-D.ietf-mmusic-connectivity-precon] defines a usage of the the precondition framework [RFC3312]. The connectivity
precondition framework [RFC3312]. The connectivity precondition precondition makes sure that the session doesn't get established
makes sure that the session doesn't get established until actual until actual packet connectivity is checked.
packet connectivity is checked.
RFC 4796, The SDP (Session Description Protocol) Content Attribute RFC 4796, The SDP (Session Description Protocol) Content Attribute
(S): [RFC4796] defines an SDP attribute for describing the purpose (S): [RFC4796] defines an SDP attribute for describing the
of a media stream. Examples include a slide view, the speaker, a purpose of a media stream. Examples include a slide view, the
sign language feed, and so on. speaker, a sign language feed, and so on.
draft-ietf-sipping-v6-transition, IPv6 Transition in the Session IPv6-TRANS, IPv6 Transition in the Session Initiation Protocol (SIP)
Initiation Protocol (SIP) (S): [I-D.ietf-sipping-v6-transition] (S): [IPv6-TRANS] defines practices for interworking between IPv6
defines practices for interworking between IPv6 and IPv6 user and IPv6 user agents. This is done through multi-homed proxies
agents. This is done through multi-homed proxies which interwork that interwork IPv4 and IPv6, along with ICE [ICE] for media
IPv4 and IPv6, along with ICE [I-D.ietf-mmusic-ice] for media
traversal. The specification includes some minor extensions and traversal. The specification includes some minor extensions and
clarifications to SDP in order to cover some additional cases. clarifications to SDP in order to cover some additional cases.
draft-ietf-sip-connect-reuse, Connection Reuse in the Session CONNECT-REUSE, Connection Reuse in the Session Initiation Protocol
Initiation Protocol (SIP) (S): [I-D.ietf-sip-connect-reuse] defines (SIP) (S): [CONNECT-REUSE] defines an extension to SIP that
an extension to SIP that allows a TLS connection between servers allows a Transport Layer Security (TLS) connection between servers
to be reused for requests in both directions. Normally two to be reused for requests in both directions. Normally, two
connections are set up between a pair of servers, one for requests connections are set up between a pair of servers, one for requests
in each direction. in each direction.
15. Security Mechanisms 15. Security Mechanisms
Several extensions provide additional security features to SIP. Several extensions provide additional security features to SIP.
RFC 4474, Enhancements for Authenticated Identity Management in SIP RFC 4474, Enhancements for Authenticated Identity Management in SIP
(S): [RFC4474] defines a mechanism for providing a cryptographically (S): [RFC4474] defines a mechanism for providing a
verifiable identity of the calling party in a SIP request. Known cryptographically verifiable identity of the calling party in a
as "SIP Identity", this mechanism provides an alternative to RFC SIP request. Known as "SIP Identity", this mechanism provides an
3325. It has seen little deployment so far, but its importance as alternative to RFC 3325. It has seen little deployment so far,
a key construct for anti-spam techniques and new security but its importance as a key construct for anti-spam techniques and
mechanisms makes it a core part of the SIP specifications. new security mechanisms makes it a core part of the SIP
specifications.
RFC 4916, Connected Identity in the Session Initiation Protocol (SIP) RFC 4916, Connected Identity in the Session Initiation Protocol (SIP)
(S): [RFC4916] formally updates RFC 3261. It defines an extension (S): [RFC4916] formally updates RFC 3261. It defines an
to SIP that allows a calling user to determine the identity of the extension to SIP that allows a calling user to determine the
final called user (connected party). Due to forwarding and identity of the final called user (connected party). Due to
retargeting services, this may not be the same as the user that forwarding and retargeting services, this may not be the same as
the caller was originally trying to reach. The mechanism works in the user that the caller was originally trying to reach. The
tandem with the SIP identity specification [RFC4474] to provide mechanism works in tandem with the SIP identity specification
signatures over the connected party identity. It can also be used [RFC4474] to provide signatures over the connected party identity.
if a party identity changes mid call due to third party call It can also be used if a party identity changes mid call due to
control actions or PSTN behavior. third party call control actions or PSTN behavior.
draft-ietf-sip-sips, The use of the SIPS URI Scheme in the Session SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
Initiation Protocol (SIP) (S): [I-D.ietf-sip-sips] formally updated Protocol (SIP) (S): [SIPS-URI] is intended to update RFC 3261.
RFC 3261. It revises the processing of the SIPS URI, originally It revises the processing of the SIPS URI, originally defined in
defined in RFC 3261, to fix many errors and problems that have RFC 3261, to fix many errors and problems that have been
been encountered with that mechanism. encountered with that mechanism.
draft-ietf-sip-domain-certs, Domain Certificates in the Session DOMAIN-CERTS, Domain Certificates in the Session Initiation Protocol
Initiation Protocol (SIP) (B): [I-D.ietf-sip-domain-certs] clarifies (SIP) (B): [DOMAIN-CERTS] clarifies the usage of SIP over TLS
the usage of SIP over TLS with regards to certificate handling, with regards to certificate handling, and defines additional
and defines additional procedures needed for interoperability. procedures needed for interoperability.
RFC 3323, A Privacy Mechanism for the Session Initiation Protocol RFC 3323, A Privacy Mechanism for the Session Initiation Protocol
(SIP) (S): [RFC3323] defines the Privacy header field, used by (SIP) (S): [RFC3323] defines the Privacy header field, used by
clients to request anonymity for their requests. Though it clients to request anonymity for their requests. Though it
defines several privacy services, the only one broadly used is the defines several privacy services, the only one broadly used is the
one that supports privacy of the P-Asserted-Identity header field one that supports privacy of the P-Asserted-Identity header field
[RFC3325]. [RFC3325].
RFC 4567, Key Management Extensions for Session Description Protocol RFC 4567, Key Management Extensions for Session Description Protocol
(SDP) and Real Time Streaming Protocol (RTSP) (S): [RFC4567] defines (SDP) and Real Time Streaming Protocol (RTSP) (S): [RFC4567]
extensions to SDP that allow tunneling of an key management defines extensions to SDP that allow tunneling of a key management
protocol, namely MIKEY [RFC3830], through offer/answer exchanges. protocol, namely MIKEY [RFC3830], through offer/answer exchanges.
This mechanism is one of three SRTP keying techniques specified This mechanism is one of three Secure Realtime Transport Protocol
for SIP, with DTLS-SRTP [I-D.ietf-sip-dtls-srtp-framework] having (SRTP) keying techniques specified for SIP, with Datagram
been selected as the final solution. Transport Layer Security (DTLS)-SRTP [SRTP-FRAME] having been
selected as the final solution.
RFC 4568, Session Description Protocol (SDP) Security Descriptions RFC 4568, Session Description Protocol (SDP) Security Descriptions
for Media Streams (S): [RFC4568] defines extensions to SDP that for Media Streams (S): [RFC4568] defines extensions to SDP that
allow for the negotiation of keying material directly through allow for the negotiation of keying material directly through
offer/answer, without a separate key management protocol. This offer/answer, without a separate key management protocol. This
mechanism, sometimes called sdescriptions, has the drawback that mechanism, sometimes called sdescriptions, has the drawback that
the media keys are available to any entity that has visibility to the media keys are available to any entity that has visibility to
the SDP. It is one of three SRTP keying techniques specified for 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 SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the final
selected as the final solution. solution.
draft-ietf-sip-dtls-srtp-framework, Framework for Establishing an SRTP-FRAME, Framework for Establishing an SRTP Security Context using
SRTP Security Context using DTLS (S): DTLS (S): [SRTP-FRAME] defines the overall framework and SDP and
[I-D.ietf-sip-dtls-srtp-framework] defines the overall framework SIP processing required to perform key management for RTP using
and SDP and SIP processing required to perform key management for Datagram TLS (DTLS) [RFC4347] directly between endpoints, over the
RTP using Datagram TLS (DTLS) [RFC4347] directly between media path. It is one of three SRTP keying techniques specified
endpoints, over the media path. It is one of three SRTP keying for SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the
techniques specified for SIP, with DTLS-SRTP
[I-D.ietf-sip-dtls-srtp-framework] having been selected as the
final solution. final solution.
draft-ietf-mmusic-sdp-dtls, Session Description Protocol (SDP) RFC 3853, S/MIME Advanced Encryption Standard (AES) Requirement for
Indicators for Datagram Transport Layer Security (DTLS) (S): SIP (S): [RFC3853] formally updates RFC 3261. It is a brief
[I-D.ietf-mmusic-sdp-dtls] defines the usage of SDP with DTLS- specification that updates the cryptography mechanisms used in SIP
SRTP. S/MIME. However, SIP S/MIME has seen very little deployment.
RFC 3853, S/MIME AES Requirement for SIP (S): [RFC3853] formally
updates RFC 3261. It is a brief specification that updates the
cryptography mechanisms used in SIP S/MIME. However, SIP S/MIME
has seen very little deployment.
draft-ietf-sip-certs, Certificate Management Service for The Session CERTS, Certificate Management Service for the Session Initiation
Initiation Protocol (SIP) (S): [I-D.ietf-sip-certs] defines a Protocol (SIP) (S): [CERTS] defines a certificate service for SIP
certificate service for SIP whose purpose is to facilitate the whose purpose is to facilitate the deployment of S/MIME. The
deployment of S/MIME. The certificate service allows clients to certificate service allows clients to store and retrieve their own
store and retrieve their own certificates, in addition to certificates, in addition to obtaining the certificates for other
obtaining the certificates for other users. users.
RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity
Body (AIB) Format (S): [RFC3893] defines a SIP message fragment Body (AIB) Format (S): [RFC3893] defines a SIP message fragment
which can be signed in order to provide an authenticated identity that can be signed in order to provide an authenticated identity
over a request. It was an early predecessor to [RFC4474], and over a request. It was an early predecessor to [RFC4474], and
consequently AIB has seen no deployment. consequently AIB has seen no deployment.
draft-ietf-sip-saml, SIP SAML Profile and Binding (S): SAML, SIP SAML Profile and Binding (S): [SAML] defines the usage of
[I-D.ietf-sip-saml] defines the usage of the Security Assertion the Security Assertion Markup Language (SAML) within SIP, and
Markup Language (SAML) within SIP, and describes how to use it in describes how to use it in conjunction with SIP identity [RFC4474]
conjunction with SIP identity [RFC4474] to provide authenticated to provide authenticated assertions about a user's role or
assertions about a users role or attributes. attributes.
RFC 5360, A Framework for Consent-Based Communications in the RFC 5360, A Framework for Consent-Based Communications in the Session
Session Initiation Protocol (SIP) (S): [RFC5360] defines several Initiation Protocol (SIP) (S): [RFC5360] defines several
extensions to SIP, including the Trigger-Consent and Permission- extensions to SIP, including the Trigger-Consent and Permission-
Missing header fields. These header fields, in addition to the Missing header fields. These header fields, in addition to the
other procedures defined in the document, define a way to manage other procedures defined in the document, define a way to manage
membership on "SIP mailing lists" used for instant messaging or membership on "SIP mailing lists" used for instant messaging or
conferencing. In particular, it helps avoid the problem of using conferencing. In particular, it helps avoid the problem of using
such amplification services for the purposes of an attack on the such amplification services for the purposes of an attack on the
network, by making sure a user authorizes the addition of their network by making sure a user authorizes the addition of their
address onto such a service. address onto such a service.
RFC 5361, A Document Format for Requesting Consent (S): [RFC5361] RFC 5361, A Document Format for Requesting Consent (S): [RFC5361]
defines an XML object used by the consent framework. Consent defines an XML object used by the consent framework. Consent
documents are sent from SIP "mailing list servers" to users to documents are sent from SIP "mailing list servers" to users to
allow them to manage their membership on lists. allow them to manage their membership on lists.
RFC 5362, The Session Initiation Protocol (SIP) Pending Additions RFC 5362, The Session Initiation Protocol (SIP) Pending Additions
Event Package (S): [RFC5362] defines a SIP event package that allows Event Package (S): [RFC5362] defines a SIP event package that
a UA to learn whether consent has been given for the addition of allows a UA to learn whether consent has been given for the
an address to a SIP "mailing list". It is used in conjunction addition of an address to a SIP "mailing list". It is used in
with the SIP framework for consent [RFC5360]. conjunction with the SIP framework for consent [RFC5360].
RFC 3329, Security Mechanism Agreement for SIP (S): [RFC3329] RFC 3329, Security Mechanism Agreement for SIP (S): [RFC3329]
defines a mechanism to prevent bid-down attacks in conjunction defines a mechanism to prevent bid-down attacks in conjunction
with SIP authentication. The mechanism has seen very limited with SIP authentication. The mechanism has seen very limited
deployment. It was defined as part of the 3gpp IMS specification deployment. It was defined as part of the 3GPP IP Multimedia
suite [3GPP.24.229], and is needed only when there is a Subsystem (IMS) specification suite [3GPP.24.229], and is needed
multiplicity of security mechanisms deployed at a particular only when there is a multiplicity of security mechanisms deployed
server. In practice, this has not been the case. at a particular server. In practice, this has not been the case.
draft-ietf-sip-e2m-sec, End-to-Middle Security in SIP (S):
[I-D.ietf-sip-e2m-sec] defines mechanisms for providing
confidentiality and integrity for SIP message bodies sent from
user agents to specific network intermediaries.
RFC 4572, Connection-Oriented Media Transport over the Transport RFC 4572, Connection-Oriented Media Transport over the Transport
Layer Security (TLS) Protocol in the Session Description Protocol Layer Security (TLS) Protocol in the Session Description Protocol
(SDP) (S): [RFC4572] specifies a mechanism for signaling TLS-based (SDP) (S): [RFC4572] specifies a mechanism for signaling TLS-
media streams between endpoints. It expands the TCP-based media based media streams between endpoints. It expands the TCP-based
signaling parameters defined in [RFC4145] to include fingerprint media signaling parameters defined in [RFC4145] to include
information for TLS streams, so that TLS can operate between end fingerprint information for TLS streams so that TLS can operate
hosts using self-signed certificates. between end hosts using self-signed certificates.
RFC 5027, Security Preconditions for Session Description Protocol RFC 5027, Security Preconditions for Session Description Protocol
Media Streams (S): [RFC5027] defines a precondition for use with the Media Streams (S): [RFC5027] defines a precondition for use with
preconditions framework [RFC3312]. The security precondition the preconditions framework [RFC3312]. The security precondition
prevents a session from being established until a security media prevents a session from being established until a security media
stream is set up. stream is set up.
RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (S): [RFC3310] defines an Using Authentication and Key Agreement (S): [RFC3310] defines an
extension to digest authentication to allow it to work with the extension to digest authentication to allow it to work with the
credentials stored in cell phones. Though technically it is an credentials stored in cell phones. Though technically it is an
extension to HTTP digest, it's primary application is SIP. This extension to HTTP digest, its primary application is SIP. This
extension is usefully primarily to implementors of IMS. extension is useful primarily to implementors of IMS.
RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication
Using Authentication and Key Agreement (AKA) Version-2 (S): Using Authentication and Key Agreement (AKA) Version-2 (S):
[RFC4169] is an enhancement to [RFC3310] which further improves [RFC4169] is an enhancement to [RFC3310] that further improves
security of the authentication. security of the authentication.
RFC 3329, Security Mechanism Agreement for SIP (S): [RFC3329] is a
SIP security extension meant to be used in conjunction with IMS,
to ensure that there are no bid-down attacks in the negotiation of
an authentication mechanism.
16. Conferencing 16. Conferencing
Numerous SIP and SDP extensions are aimed at conferencing as their Numerous SIP and SDP extensions are aimed at conferencing as their
primary application. primary application.
RFC 4574, The SDP (Session Description Protocol) Label Attribute RFC 4574, The SDP (Session Description Protocol) Label Attribute
(S): [RFC4574] defines an SDP attribute for providing an opaque (S): [RFC4574] defines an SDP attribute for providing an opaque
label for media streams. These labels can be referred to by label for media streams. These labels can be referred to by
external documents, and in particular, by conference policy external documents, and in particular, by conference policy
documents. This allows a UA to tie together documents it may documents. This allows a UA to tie together documents it may
skipping to change at page 26, line 46 skipping to change at page 25, line 46
resource, and it will act as a conference focus and generate an resource, and it will act as a conference focus and generate an
invitation to each recipient in the list. invitation to each recipient in the list.
RFC4579, Session Initiation Protocol (SIP) Call Control - RFC4579, Session Initiation Protocol (SIP) Call Control -
Conferencing for User Agents (B): [RFC4579] defines best practice Conferencing for User Agents (B): [RFC4579] defines best practice
procedures and call flows for conferencing. This includes procedures and call flows for conferencing. This includes
conference creation, joining, and dial out, amongst other conference creation, joining, and dial out, amongst other
capabilities. capabilities.
RFC 4583, Session Description Protocol (SDP) Format for Binary Floor RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
Control Protocol (BFCP) Streams (S): [RFC4583] defines a mechanism Control Protocol (BFCP) Streams (S): [RFC4583] defines a
in SDP to signal floor control streams that use BFCP. It is used mechanism in SDP to signal floor control streams that use BFCP.
for Push-To-Talk and conference floor control. It is used for push-to-talk and conference floor control.
17. Instant Messaging, Presence and Multimedia 17. Instant Messaging, Presence, and Multimedia
SIP provides extensions for instant messaging, presence, and SIP provides extensions for instant messaging, presence, and
multimedia. multimedia.
RFC 3428, SIP Extension for Instant Messaging (S): [RFC3428] defines RFC 3428, SIP Extension for Instant Messaging (S): [RFC3428] defines
the MESSAGE method, used for sending an instant message without the MESSAGE method, used for sending an instant message without
setting up a session (sometimes called "page mode"). setting up a session (sometimes called "page mode").
RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an
event package for indicating user presence through SIP. event package for indicating user presence through SIP.
RFC 3857, A Watcher Information Event Template Package for SIP (S): RFC 3857, A Watcher Information Event Template Package for SIP (S):
[RFC3857], also known as winfo, provides a mechanism for a user [RFC3857], also known as winfo, provides a mechanism for a user
agent to find out what subscriptions are in place for a particular agent to find out what subscriptions are in place for a particular
event package. Its primary usage is with presence, but it can be event package. Its primary usage is with presence, but it can be
used with any event package. used with any event package.
draft-ietf-mmusic-file-transfer-mech, A Session Description Protocol TRANSFER-MECH, A Session Description Protocol (SDP) Offer/Answer
(SDP) Offer/Answer Mechanism to Enable File Transfer (S): Mechanism to Enable File Transfer (S): [TRANSFER-MECH] defines a
[I-D.ietf-mmusic-file-transfer-mech] defines a mechanism for mechanism for signaling a file transfer session with SIP.
signaling a file transfer session with SIP.
18. Emergency Services 18. Emergency Services
Emergency services include pre-emption features, which allow Emergency services include preemption features, which allow
authorized individuals to gain access to network resources in time of authorized individuals to gain access to network resources in time of
emergency, along with traditional emergency calling. emergency, along with traditional emergency calling.
RFC 4411, Extending the SIP Reason Header for Preemption Events (S): RFC 4411, Extending the SIP Reason Header for Preemption Events (S):
[RFC4411] defines an extension to the Reason header, allowing a UA [RFC4411] defines an extension to the Reason header, allowing a UA
to know that its dialog was torn down because a higher priority to know that its dialog was torn down because a higher priority
session came through. session came through.
RFC 4412, Communications Resource Priority for SIP (S): [RFC4412] RFC 4412, Communications Resource Priority for SIP (S): [RFC4412]
defines a new header field, Resource-Priority, that allows a defines a new header field, Resource-Priority, that allows a
session to get priority treatment from the network. session to get priority treatment from the network.
draft-ietf-sip-location-conveyance, Location Conveyance for the LOCATION, Location Conveyance for the Session Initiation Protocol
Session Initiation Protocol (S): [I-D.ietf-sip-location-conveyance] (S): [LOCATION] defines a mechanism for carrying location objects
defines a mechanism for carrying location objects in SIP messages. in SIP messages. This is used to convey location from a UA to an
This is used to convey location from a UA to an emergency call emergency call taker.
taker.
19. Security Considerations 19. Security Considerations
This specification is an overview of existing specifications, and This specification is an overview of existing specifications and does
does not introduce any security considerations on its own. Of not introduce any security considerations on its own. Of course, the
course, the world would be far more secure if everyone would follow world would be far more secure if everyone would follow one simple
one simple rule: "Don't Panic!" [HGTTG]. rule: "Don't Panic!" [HGTTG].
20. IANA Considerations
None.
21. Acknowledgements 20. Acknowledgements
The author would like to thank Spencer Dawkins, Brian Stucker, Keith The author would like to thank Spencer Dawkins, Brian Stucker, Keith
Drage, John Elwell and Avshalom Houri for their comments on this Drage, John Elwell, and Avshalom Houri for their comments on this
document. document.
22. Informative References 21. Informative References
[RFC3261] 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.
[RFC2026] Bradner, S., "The Internet Standards Process -- Revision
3", BCP 9, RFC 2026, October 1996.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [3GPP.24.229] 3GPP, "Internet Protocol (IP) multimedia call
Jacobson, "RTP: A Transport Protocol for Real-Time control protocol based on Session Initiation
Applications", RFC 3550, July 2003. Protocol (SIP) and Session Description Protocol
(SDP); Stage 3", 3GPP TS 24.229 5.22.0,
September 2008.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [ABNF-FIX] Gurbani, V. and B. Carpenter, "Essential correction
with Session Description Protocol (SDP)", RFC 3264, for IPv6 ABNF in RFC3261", Work in Progress,
June 2002. November 2007.
[I-D.ietf-mmusic-ice] [BODY-HANDLING] Camarillo, G., "Message Body Handling in the
Rosenberg, J., "Interactive Connectivity Establishment Session Initiation Protocol (SIP)", Work
(ICE): A Protocol for Network Address Translator (NAT) in Progress, November 2008.
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-19 (work in progress), October 2007.
[RFC3320] Price, R., Bormann, C., Christoffersson, J., Hannu, H., [CERTS] Jennings, C. and J. Fischl, "Certificate Management
Liu, Z., and J. Rosenberg, "Signaling Compression Service for The Session Initiation Protocol (SIP)",
(SigComp)", RFC 3320, January 2003. Work in Progress, November 2008.
[RFC3893] Peterson, J., "Session Initiation Protocol (SIP) [CONFIG-FRAME] Channabasappa, S., "A Framework for Session
Authenticated Identity Body (AIB) Format", RFC 3893, Initiation Protocol User Agent Profile Delivery",
September 2004. Work in Progress, February 2008.
[RFC3427] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., [CONNECT-PRECON] Andreasen, F., Camarillo, G., Oran, D., and D.
and B. Rosen, "Change Process for the Session Initiation Wing, "Connectivity Preconditions for Session
Protocol (SIP)", BCP 67, RFC 3427, December 2002. Description Protocol Media Streams", Work
in Progress, October 2008.
[RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J. [CONNECT-REUSE] Gurbani, V., Mahy, R., and B. Tate, "Connection
Rosenberg, "SIP: Session Initiation Protocol", RFC 2543, Reuse in the Session Initiation Protocol (SIP)",
March 1999. Work in Progress, October 2008.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation [DOMAIN-CERTS] Gurbani, V., Lawrence, S., and B. Laboratories,
Protocol (SIP): Locating SIP Servers", RFC 3263, "Domain Certificates in the Session Initiation
June 2002. Protocol (SIP)", Work in Progress, October 2008.
[RFC2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for [ECRIT-FRAME] Rosen, B., Schulzrinne, H., Polk, J., and A.
specifying the location of services (DNS SRV)", RFC 2782, Newton, "Framework for Emergency Calling using
February 2000. Internet Multimedia", Work in Progress, July 2008.
[RFC2915] Mealling, M. and R. Daniel, "The Naming Authority Pointer [GRUU] Rosenberg, J., "Obtaining and Using Globally
(NAPTR) DNS Resource Record", RFC 2915, September 2000. Routable User Agent (UA) URIs (GRUU) in the Session
Initiation Protocol (SIP)", Work in Progress,
October 2007.
[RFC3265] Roach, A., "Session Initiation Protocol (SIP)-Specific [GRUU-REG] Kyzivat, P., "Registration Event Package Extension
Event Notification", RFC 3265, June 2002. for Session Initiation Protocol (SIP) Globally
Routable User Agent URIs (GRUUs)", Work
in Progress, July 2007.
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session [HGTTG] Adams, D., "The Hitchhiker's Guide to the Galaxy",
Initiation Protocol (SIP)", RFC 3323, November 2002. September 1979.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private [ICE] Rosenberg, J., "Interactive Connectivity
Extensions to the Session Initiation Protocol (SIP) for Establishment (ICE): A Protocol for Network Address
Asserted Identity within Trusted Networks", RFC 3325, Translator (NAT) Traversal for Offer/Answer
November 2002. Protocols", Work in Progress, October 2007.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol [ICE-TCP] Rosenberg, J., "TCP Candidates with Interactive
(SIP) Extension Header Field for Registering Non-Adjacent Connectivity Establishment (ICE)", Work
Contacts", RFC 3327, December 2002. in Progress, July 2008.
[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the [INTERACT-FRAME] Rosenberg, J., "A Framework for Application
Session Initiation Protocol (SIP) for Symmetric Response Interaction in the Session Initiation Protocol
Routing", RFC 3581, August 2003. (SIP)", Work in Progress, July 2005.
[RFC4320] Sparks, R., "Actions Addressing Identified Issues with the [IPv6-TRANS] Camarillo, G., "IPv6 Transition in the Session
Session Initiation Protocol's (SIP) Non-INVITE Initiation Protocol (SIP)", Work in Progress,
Transaction", RFC 4320, January 2006. August 2007.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for [LOCATION] Polk, J. and B. Rosen, "Location Conveyance for the
Authenticated Identity Management in the Session Session Initiation Protocol", Work in Progress,
Initiation Protocol (SIP)", RFC 4474, August 2006. November 2008.
[I-D.ietf-sip-gruu] [LOOP-FIX] Sparks, R., Lawrence, S., Hawrylyshen, A., and B.
Rosenberg, J., "Obtaining and Using Globally Routable User Campen, "Addressing an Amplification Vulnerability
Agent (UA) URIs (GRUU) in the Session Initiation Protocol in Session Initiation Protocol (SIP) Forking
(SIP)", draft-ietf-sip-gruu-15 (work in progress), Proxies", Work in Progress, October 2008.
October 2007.
[I-D.ietf-sip-outbound] [OPTION-TAG] Rosenberg, J., "Indicating Support for Interactive
Jennings, C. and R. Mahy, "Managing Client Initiated Connectivity Establishment (ICE) in the Session
Connections in the Session Initiation Protocol (SIP)", Initiation Protocol (SIP)", Work in Progress,
draft-ietf-sip-outbound-16 (work in progress), June 2007.
October 2008.
[RFC2848] Petrack, S. and L. Conroy, "The PINT Service Protocol: [OUTBOUND] Jennings, C. and R. Mahy, "Managing Client
Extensions to SIP and SDP for IP Access to Telephone Call Initiated Connections in the Session Initiation
Services", RFC 2848, June 2000. Protocol (SIP)", Work in Progress, October 2008.
[RFC3910] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato, J., Lu, [POLICY-PACK] Hilt, V. and G. Camarillo, "A Session Initiation
H., and M. Unmehopa, "The SPIRITS (Services in PSTN Protocol (SIP) Event Package for Session-Specific
requesting Internet Services) Protocol", RFC 3910, Session Policies.", Work in Progress, July 2008.
October 2004.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation Protocol [QoS-ID] Polk, J., Dhesikan, S., and G. Camarillo, "Quality
for Telephones (SIP-T): Context and Architectures", of Service (QoS) Mechanism Selection in the Session
BCP 63, RFC 3372, September 2002. Description Protocol (SDP)", Work in Progress,
November 2008.
[RFC3398] Camarillo, G., Roach, A., Peterson, J., and L. Ong, [RECORD-ROUTE] Froment, T., Lebel, C., and B. Bonnaerens,
"Integrated Services Digital Network (ISDN) User Part "Addressing Record-Route issues in the Session
(ISUP) to Session Initiation Protocol (SIP) Mapping", Initiation Protocol (SIP)", Work in Progress,
RFC 3398, December 2002. October 2008.
[RFC3578] Camarillo, G., Roach, A., Peterson, J., and L. Ong, [RFC2026] Bradner, S., "The Internet Standards Process --
"Mapping of Integrated Services Digital Network (ISDN) Revision 3", BCP 9, RFC 2026, October 1996.
User Part (ISUP) Overlap Signalling to the Session
Initiation Protocol (SIP)", RFC 3578, August 2003.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing [RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J.
Tone Generation in the Session Initiation Protocol (SIP)", Rosenberg, "SIP: Session Initiation Protocol",
RFC 3960, December 2004. RFC 2543, March 1999.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of [RFC2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS
Provisional Responses in Session Initiation Protocol RR for specifying the location of services (DNS
(SIP)", RFC 3262, June 2002. SRV)", RFC 2782, February 2000.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP) [RFC2848] Petrack, S. and L. Conroy, "The PINT Service
UPDATE Method", RFC 3311, October 2002. Protocol: Extensions to SIP and SDP for IP Access
to Telephone Call Services", RFC 2848, June 2000.
[RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976, [RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976,
October 2000. October 2000.
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason [RFC3087] Campbell, B. and R. Sparks, "Control of Service
Header Field for the Session Initiation Protocol (SIP)", Context using SIP Request-URI", RFC 3087,
RFC 3326, December 2002. April 2001.
[RFC3608] Willis, D. and B. Hoeneisen, "Session Initiation Protocol [RFC3204] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L.,
(SIP) Extension Header Field for Service Route Discovery Audet, F., Watson, M., and M. Zonoun, "MIME media
During Registration", RFC 3608, October 2003. types for ISUP and QSIG Objects", RFC 3204,
December 2001.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G.,
"Indicating User Agent Capabilities in the Session Johnston, A., Peterson, J., Sparks, R., Handley,
Initiation Protocol (SIP)", RFC 3840, August 2004. M., and E. Schooler, "SIP: Session Initiation
Protocol", RFC 3261, June 2002.
[RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller [RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Preferences for the Session Initiation Protocol (SIP)", Provisional Responses in Session Initiation
RFC 3841, August 2004. Protocol (SIP)", RFC 3262, June 2002.
[RFC4028] Donovan, S. and J. Rosenberg, "Session Timers in the [RFC3263] Rosenberg, J. and H. Schulzrinne, "Session
Session Initiation Protocol (SIP)", RFC 4028, April 2005. Initiation Protocol (SIP): Locating SIP Servers",
RFC 3263, June 2002.
[RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Stream Control Transmission Protocol (SCTP) as a Transport Model with Session Description Protocol (SDP)",
for the Session Initiation Protocol (SIP)", RFC 4168, RFC 3264, June 2002.
October 2005.
[RFC4244] Barnes, M., "An Extension to the Session Initiation [RFC3265] Roach, A., "Session Initiation Protocol (SIP)-
Protocol (SIP) for Request History Information", RFC 4244, Specific Event Notification", RFC 3265, June 2002.
November 2005.
[RFC4488] Levin, O., "Suppression of Session Initiation Protocol [RFC3310] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext
(SIP) REFER Method Implicit Subscription", RFC 4488, Transfer Protocol (HTTP) Digest Authentication
May 2006. Using Authentication and Key Agreement (AKA)",
RFC 3310, September 2002.
[RFC4538] Rosenberg, J., "Request Authorization through Dialog [RFC3311] Rosenberg, J., "The Session Initiation Protocol
Identification in the Session Initiation Protocol (SIP)", (SIP) UPDATE Method", RFC 3311, October 2002.
RFC 4538, June 2006.
[RFC4508] Levin, O. and A. Johnston, "Conveying Feature Tags with [RFC3312] Camarillo, G., Marshall, W., and J. Rosenberg,
the Session Initiation Protocol (SIP) REFER Method", "Integration of Resource Management and Session
RFC 4508, May 2006. Initiation Protocol (SIP)", RFC 3312, October 2002.
[I-D.ietf-sip-answermode] [RFC3313] Marshall, W., "Private Session Initiation Protocol
Willis, D. and A. Allen, "Requesting Answering Modes for (SIP) Extensions for Media Authorization",
the Session Initiation Protocol (SIP)", RFC 3313, January 2003.
draft-ietf-sip-answermode-07 (work in progress), May 2008.
[HGTTG] Adams, D., "The Hitchhiker's Guide to the Galaxy", [RFC3320] Price, R., Bormann, C., Christoffersson, J., Hannu,
September 1979. H., Liu, Z., and J. Rosenberg, "Signaling
Compression (SigComp)", RFC 3320, January 2003.
[RFC5079] Rosenberg, J., "Rejecting Anonymous Requests in the [RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Session Initiation Protocol (SIP)", RFC 5079, Initiation Protocol (SIP)", RFC 3323,
December 2007. November 2002.
[RFC5368] Camarillo, G., Niemi, A., Isomaki, M., Garcia-Martin, M., [RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
and H. Khartabil, "Referring to Multiple Resources in the Extensions to the Session Initiation Protocol (SIP)
Session Initiation Protocol (SIP)", RFC 5368, for Asserted Identity within Trusted Networks",
October 2008. RFC 3325, November 2002.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer [RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The
Method", RFC 3515, April 2003. Reason Header Field for the Session Initiation
Protocol (SIP)", RFC 3326, December 2002.
[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. [RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation
Camarillo, "Best Current Practices for Third Party Call Protocol (SIP) Extension Header Field for
Control (3pcc) in the Session Initiation Protocol (SIP)", Registering Non-Adjacent Contacts", RFC 3327,
BCP 85, RFC 3725, April 2004. December 2002.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation [RFC3329] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A.,
Protocol (SIP) "Replaces" Header", RFC 3891, and T. Haukka, "Security Mechanism Agreement for
September 2004. the Session Initiation Protocol (SIP)", RFC 3329,
January 2003.
[RFC3892] Sparks, R., "The Session Initiation Protocol (SIP) [RFC3372] Vemuri, A. and J. Peterson, "Session Initiation
Referred-By Mechanism", RFC 3892, September 2004. Protocol for Telephones (SIP-T): Context and
Architectures", BCP 63, RFC 3372, September 2002.
[RFC3911] Mahy, R. and D. Petrie, "The Session Initiation Protocol [RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
(SIP) "Join" Header", RFC 3911, October 2004. Schulzrinne, "Grouping of Media Lines in the
Session Description Protocol (SDP)", RFC 3388,
December 2002.
[RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A. van [RFC3398] Camarillo, G., Roach, A., Peterson, J., and L. Ong,
Wijk, "Transcoding Services Invocation in the Session "Integrated Services Digital Network (ISDN) User
Initiation Protocol (SIP) Using Third Party Call Control Part (ISUP) to Session Initiation Protocol (SIP)
(3pcc)", RFC 4117, June 2005. Mapping", RFC 3398, December 2002.
[RFC3903] Niemi, A., "Session Initiation Protocol (SIP) Extension [RFC3401] Mealling, M., "Dynamic Delegation Discovery System
for Event State Publication", RFC 3903, October 2004. (DDDS) Part One: The Comprehensive DDDS", RFC 3401,
October 2002.
[RFC3680] Rosenberg, J., "A Session Initiation Protocol (SIP) Event [RFC3420] Sparks, R., "Internet Media Type message/sipfrag",
Package for Registrations", RFC 3680, March 2004. RFC 3420, November 2002.
[RFC3856] Rosenberg, J., "A Presence Event Package for the Session [RFC3427] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott,
Initiation Protocol (SIP)", RFC 3856, August 2004. J., and B. Rosen, "Change Process for the Session
Initiation Protocol (SIP)", BCP 67, RFC 3427,
December 2002.
[RFC3857] Rosenberg, J., "A Watcher Information Event Template- [RFC3428] Campbell, B., Rosenberg, J., Schulzrinne, H.,
Package for the Session Initiation Protocol (SIP)", Huitema, C., and D. Gurle, "Session Initiation
RFC 3857, August 2004. Protocol (SIP) Extension for Instant Messaging",
RFC 3428, December 2002.
[RFC4235] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE- [RFC3482] Foster, M., McGarry, T., and J. Yu, "Number
Initiated Dialog Event Package for the Session Initiation Portability in the Global Switched Telephone
Protocol (SIP)", RFC 4235, November 2005. Network (GSTN): An Overview", RFC 3482,
February 2003.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session [RFC3486] Camarillo, G., "Compressing the Session Initiation
Initiation Protocol (SIP) Event Package for Conference Protocol (SIP)", RFC 3486, February 2003.
State", RFC 4575, August 2006.
[RFC4730] Burger, E. and M. Dolly, "A Session Initiation Protocol [RFC3515] Sparks, R., "The Session Initiation Protocol (SIP)
(SIP) Event Package for Key Press Stimulus (KPML)", Refer Method", RFC 3515, April 2003.
RFC 4730, November 2006.
[I-D.ietf-sipping-rtcp-summary] [RFC3524] Camarillo, G. and A. Monrad, "Mapping of Media
Clark, A., Pendleton, A., Johnston, A., and H. Sinnreich, Streams to Resource Reservation Flows", RFC 3524,
"Session Initiation Protocol Package for Voice Quality April 2003.
Reporting Event", draft-ietf-sipping-rtcp-summary-05 (work
in progress), October 2008.
[RFC3312] Camarillo, G., Marshall, W., and J. Rosenberg, [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
"Integration of Resource Management and Session Initiation Jacobson, "RTP: A Transport Protocol for Real-Time
Protocol (SIP)", RFC 3312, October 2002. Applications", STD 64, RFC 3550, July 2003.
[RFC4032] Camarillo, G. and P. Kyzivat, "Update to the Session [RFC3578] Camarillo, G., Roach, A., Peterson, J., and L. Ong,
Initiation Protocol (SIP) Preconditions Framework", "Mapping of Integrated Services Digital Network
RFC 4032, March 2005. (ISDN) User Part (ISUP) Overlap Signalling to the
Session Initiation Protocol (SIP)", RFC 3578,
August 2003.
[RFC3313] Marshall, W., "Private Session Initiation Protocol (SIP) [RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to
Extensions for Media Authorization", RFC 3313, the Session Initiation Protocol (SIP) for Symmetric
January 2003. Response Routing", RFC 3581, August 2003.
[I-D.ietf-sipping-config-framework] [RFC3605] Huitema, C., "Real Time Control Protocol (RTCP)
Channabasappa, S., "A Framework for Session Initiation attribute in Session Description Protocol (SDP)",
Protocol User Agent Profile Delivery", RFC 3605, October 2003.
draft-ietf-sipping-config-framework-15 (work in progress),
February 2008.
[RFC3486] Camarillo, G., "Compressing the Session Initiation [RFC3608] Willis, D. and B. Hoeneisen, "Session Initiation
Protocol (SIP)", RFC 3486, February 2003. Protocol (SIP) Extension Header Field for Service
Route Discovery During Registration", RFC 3608,
October 2003.
[RFC3482] Foster, M., McGarry, T., and J. Yu, "Number Portability in [RFC3665] Johnston, A., Donovan, S., Sparks, R., Cunningham,
the Global Switched Telephone Network (GSTN): An C., and K. Summers, "Session Initiation Protocol
Overview", RFC 3482, February 2003. (SIP) Basic Call Flow Examples", BCP 75, RFC 3665,
December 2003.
[RFC3087] Campbell, B. and R. Sparks, "Control of Service Context [RFC3666] Johnston, A., Donovan, S., Sparks, R., Cunningham,
using SIP Request-URI", RFC 3087, April 2001. C., and K. Summers, "Session Initiation Protocol
(SIP) Public Switched Telephone Network (PSTN) Call
Flows", BCP 76, RFC 3666, December 2003.
[RFC4662] Roach, A., Campbell, B., and J. Rosenberg, "A Session [RFC3680] Rosenberg, J., "A Session Initiation Protocol (SIP)
Initiation Protocol (SIP) Event Notification Extension for Event Package for Registrations", RFC 3680,
Resource Lists", RFC 4662, August 2006. March 2004.
[RFC5367] Camarillo, G., Roach, A., and O. Levin, "Subscriptions to [RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and
Request-Contained Resource Lists in the Session Initiation G. Camarillo, "Best Current Practices for Third
Protocol (SIP)", RFC 5367, October 2008. Party Call Control (3pcc) in the Session Initiation
Protocol (SIP)", BCP 85, RFC 3725, April 2004.
[RFC5365] Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M.,
MESSAGE Requests in the Session Initiation Protocol and K. Norrman, "MIKEY: Multimedia Internet
(SIP)", RFC 5365, October 2008. KEYing", RFC 3830, August 2004.
[RFC5366] Camarillo, G. and A. Johnston, "Conference Establishment [RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
Using Request-Contained Lists in the Session Initiation "Indicating User Agent Capabilities in the Session
Protocol (SIP)", RFC 5366, October 2008. Initiation Protocol (SIP)", RFC 3840, August 2004.
[RFC3853] Peterson, J., "S/MIME Advanced Encryption Standard (AES) [RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
Requirement for the Session Initiation Protocol (SIP)", "Caller Preferences for the Session Initiation
RFC 3853, July 2004. Protocol (SIP)", RFC 3841, August 2004.
[RFC3329] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T. [RFC3842] Mahy, R., "A Message Summary and Message Waiting
Haukka, "Security Mechanism Agreement for the Session Indication Event Package for the Session Initiation
Initiation Protocol (SIP)", RFC 3329, January 2003. Protocol (SIP)", RFC 3842, August 2004.
[I-D.ietf-sip-e2m-sec] [RFC3853] Peterson, J., "S/MIME Advanced Encryption Standard
Ono, K. and S. Tachimoto, "End-to-middle Security in the (AES) Requirement for the Session Initiation
Session Initiation Protocol (SIP)", Protocol (SIP)", RFC 3853, July 2004.
draft-ietf-sip-e2m-sec-06 (work in progress), July 2007.
[RFC3428] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., [RFC3856] Rosenberg, J., "A Presence Event Package for the
and D. Gurle, "Session Initiation Protocol (SIP) Extension Session Initiation Protocol (SIP)", RFC 3856,
for Instant Messaging", RFC 3428, December 2002. August 2004.
[RFC4411] Polk, J., "Extending the Session Initiation Protocol (SIP) [RFC3857] Rosenberg, J., "A Watcher Information Event
Reason Header for Preemption Events", RFC 4411, Template-Package for the Session Initiation
February 2006. Protocol (SIP)", RFC 3857, August 2004.
[RFC4412] Schulzrinne, H. and J. Polk, "Communications Resource [RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Priority for the Session Initiation Protocol (SIP)", Modifier for the Session Description Protocol
RFC 4412, February 2006. (SDP)", RFC 3890, September 2004.
[I-D.ietf-sipping-app-interaction-framework] [RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session
Rosenberg, J., "A Framework for Application Interaction in Initiation Protocol (SIP) "Replaces" Header",
the Session Initiation Protocol (SIP)", RFC 3891, September 2004.
draft-ietf-sipping-app-interaction-framework-05 (work in
progress), July 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC3892] Sparks, R., "The Session Initiation Protocol (SIP)
Description Protocol", RFC 4566, July 2006. Referred-By Mechanism", RFC 3892, September 2004.
[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H. [RFC3893] Peterson, J., "Session Initiation Protocol (SIP)
Schulzrinne, "Grouping of Media Lines in the Session Authenticated Identity Body (AIB) Format",
Description Protocol (SDP)", RFC 3388, December 2002. RFC 3893, September 2004.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute [RFC3903] Niemi, A., "Session Initiation Protocol (SIP)
in Session Description Protocol (SDP)", RFC 3605, Extension for Event State Publication", RFC 3903,
October 2003. October 2004.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation [RFC3910] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato,
Protocol (SIP)", RFC 4916, June 2007. J., Lu, H., and M. Unmehopa, "The SPIRITS (Services
in PSTN requesting Internet Services) Protocol",
RFC 3910, October 2004.
[I-D.ietf-sip-fork-loop-fix] [RFC3911] Mahy, R. and D. Petrie, "The Session Initiation
Sparks, R., Lawrence, S., Hawrylyshen, A., and B. Campen, Protocol (SIP) "Join" Header", RFC 3911,
"Addressing an Amplification Vulnerability in Session October 2004.
Initiation Protocol (SIP) Forking Proxies",
draft-ietf-sip-fork-loop-fix-08 (work in progress),
October 2008.
[RFC3959] Camarillo, G., "The Early Session Disposition Type for the [RFC3959] Camarillo, G., "The Early Session Disposition Type
Session Initiation Protocol (SIP)", RFC 3959, for the Session Initiation Protocol (SIP)",
December 2004. RFC 3959, December 2004.
[RFC3204] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, [RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and
F., Watson, M., and M. Zonoun, "MIME media types for ISUP Ringing Tone Generation in the Session Initiation
and QSIG Objects", RFC 3204, December 2001. Protocol (SIP)", RFC 3960, December 2004.
[RFC3420] Sparks, R., "Internet Media Type message/sipfrag", [RFC4028] Donovan, S. and J. Rosenberg, "Session Timers in
RFC 3420, November 2002. the Session Initiation Protocol (SIP)", RFC 4028,
April 2005.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in [RFC4032] Camarillo, G. and P. Kyzivat, "Update to the
the Session Description Protocol (SDP)", RFC 4145, Session Initiation Protocol (SIP) Preconditions
September 2005. Framework", RFC 4032, March 2005.
[RFC4091] Camarillo, G. and J. Rosenberg, "The Alternative Network [RFC4091] Camarillo, G. and J. Rosenberg, "The Alternative
Address Types (ANAT) Semantics for the Session Description Network Address Types (ANAT) Semantics for the
Protocol (SDP) Grouping Framework", RFC 4091, June 2005. Session Description Protocol (SDP) Grouping
Framework", RFC 4091, June 2005.
[I-D.ietf-mmusic-ice-tcp] [RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A.
Rosenberg, J., "TCP Candidates with Interactive van Wijk, "Transcoding Services Invocation in the
Connectivity Establishment (ICE)", Session Initiation Protocol (SIP) Using Third Party
draft-ietf-mmusic-ice-tcp-07 (work in progress), Call Control (3pcc)", RFC 4117, June 2005.
July 2008.
[RFC4483] Burger, E., "A Mechanism for Content Indirection in [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media
Session Initiation Protocol (SIP) Messages", RFC 4483, Transport in the Session Description Protocol
May 2006. (SDP)", RFC 4145, September 2005.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth [RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo,
Modifier for the Session Description Protocol (SDP)", "The Stream Control Transmission Protocol (SCTP) as
RFC 3890, September 2004. a Transport for the Session Initiation Protocol
(SIP)", RFC 4168, October 2005.
[RFC4583] Camarillo, G., "Session Description Protocol (SDP) Format [RFC4169] Torvinen, V., Arkko, J., and M. Naslund, "Hypertext
for Binary Floor Control Protocol (BFCP) Streams", Transfer Protocol (HTTP) Digest Authentication
RFC 4583, November 2006. Using Authentication and Key Agreement (AKA)
Version-2", RFC 4169, November 2005.
[RFC5027] Andreasen, F. and D. Wing, "Security Preconditions for [RFC4235] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An
Session Description Protocol (SDP) Media Streams", INVITE-Initiated Dialog Event Package for the
RFC 5027, October 2007. Session Initiation Protocol (SIP)", RFC 4235,
November 2005.
[I-D.ietf-mmusic-connectivity-precon] [RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic
Andreasen, F., Camarillo, G., Oran, D., and D. Wing, Network Media Services with SIP", RFC 4240,
"Connectivity Preconditions for Session Description December 2005.
Protocol Media Streams",
draft-ietf-mmusic-connectivity-precon-05 (work in
progress), October 2008.
[RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description [RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SDP) Content Attribute", RFC 4796, Protocol (SIP) for Request History Information",
February 2007. RFC 4244, November 2005.
[RFC4574] Levin, O. and G. Camarillo, "The Session Description [RFC4320] Sparks, R., "Actions Addressing Identified Issues
Protocol (SDP) Label Attribute", RFC 4574, August 2006. with the Session Initiation Protocol's (SIP) Non-
INVITE Transaction", RFC 4320, January 2006.
[I-D.ietf-sipping-policy-package] [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport
Hilt, V. and G. Camarillo, "A Session Initiation Protocol Layer Security", RFC 4347, April 2006.
(SIP) Event Package for Session-Specific Session
Policies.", draft-ietf-sipping-policy-package-05 (work in
progress), July 2008.
[RFC3524] Camarillo, G. and A. Monrad, "Mapping of Media Streams to [RFC4411] Polk, J., "Extending the Session Initiation
Resource Reservation Flows", RFC 3524, April 2003. Protocol (SIP) Reason Header for Preemption
Events", RFC 4411, February 2006.
[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network [RFC4412] Schulzrinne, H. and J. Polk, "Communications
Media Services with SIP", RFC 4240, December 2005. Resource Priority for the Session Initiation
Protocol (SIP)", RFC 4412, February 2006.
[I-D.ietf-sip-certs] [RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session
Jennings, C. and J. Fischl, "Certificate Management Initiation Protocol (SIP) URIs for Applications
Service for The Session Initiation Protocol (SIP)", such as Voicemail and Interactive Voice Response
draft-ietf-sip-certs-06 (work in progress), April 2008. (IVR)", RFC 4458, April 2006.
[RFC5360] Rosenberg, J., Camarillo, G., and D. Willis, "A Framework [RFC4474] Peterson, J. and C. Jennings, "Enhancements for
for Consent-Based Communications in the Session Initiation Authenticated Identity Management in the Session
Protocol (SIP)", RFC 5360, October 2008. Initiation Protocol (SIP)", RFC 4474, August 2006.
[I-D.ietf-sip-saml] [RFC4483] Burger, E., "A Mechanism for Content Indirection in
Tschofenig, H., Hodges, J., Peterson, J., Polk, J., and D. Session Initiation Protocol (SIP) Messages",
RFC 4483, May 2006.
Sicker, "SIP SAML Profile and Binding", [RFC4488] Levin, O., "Suppression of Session Initiation
draft-ietf-sip-saml-04 (work in progress), July 2008. Protocol (SIP) REFER Method Implicit Subscription",
RFC 4488, May 2006.
[RFC5362] Camarillo, G., "The Session Initiation Protocol (SIP) [RFC4497] Elwell, J., Derks, F., Mourot, P., and O. Rousseau,
Pending Additions Event Package", RFC 5362, October 2008. "Interworking between the Session Initiation
Protocol (SIP) and QSIG", BCP 117, RFC 4497,
May 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the [RFC4508] Levin, O. and A. Johnston, "Conveying Feature Tags
Transport Layer Security (TLS) Protocol in the Session with the Session Initiation Protocol (SIP) REFER
Description Protocol (SDP)", RFC 4572, July 2006. Method", RFC 4508, May 2006.
[I-D.ietf-mmusic-sdp-capability-negotiation] [RFC4538] Rosenberg, J., "Request Authorization through
Andreasen, F., "SDP Capability Negotiation", Dialog Identification in the Session Initiation
draft-ietf-mmusic-sdp-capability-negotiation-09 (work in Protocol (SIP)", RFC 4538, June 2006.
progress), July 2008.
[I-D.ietf-mmusic-sdp-media-capabilities] [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP:
Gilman, R., Even, R., and F. Andreasen, "SDP media Session Description Protocol", RFC 4566, July 2006.
capabilities Negotiation",
draft-ietf-mmusic-sdp-media-capabilities-05 (work in
progress), July 2008.
[I-D.ietf-mmusic-file-transfer-mech] [RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K.,
Garcia-Martin, M., Isomaki, M., Camarillo, G., Loreto, S., and E. Carrara, "Key Management Extensions for
and P. Kyzivat, "A Session Description Protocol (SDP) Session Description Protocol (SDP) and Real Time
Offer/Answer Mechanism to Enable File Transfer", Streaming Protocol (RTSP)", RFC 4567, July 2006.
draft-ietf-mmusic-file-transfer-mech-08 (work in
progress), May 2008.
[I-D.ietf-sip-ice-option-tag] [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Rosenberg, J., "Indicating Support for Interactive Description Protocol (SDP) Security Descriptions
Connectivity Establishment (ICE) in the Session for Media Streams", RFC 4568, July 2006.
Initiation Protocol (SIP)",
draft-ietf-sip-ice-option-tag-02 (work in progress),
June 2007.
[3GPP.24.229] [RFC4572] Lennox, J., "Connection-Oriented Media Transport
3GPP, "Internet Protocol (IP) multimedia call control over the Transport Layer Security (TLS) Protocol in
protocol based on Session Initiation Protocol (SIP) and the Session Description Protocol (SDP)", RFC 4572,
Session Description Protocol (SDP); Stage 3", 3GPP July 2006.
TS 24.229 5.22.0, September 2008.
[I-D.ietf-sip-record-route-fix] [RFC4574] Levin, O. and G. Camarillo, "The Session
Froment, T., Lebel, C., and B. Bonnaerens, "Addressing Description Protocol (SDP) Label Attribute",
Record-Route issues in the Session Initiation Protocol RFC 4574, August 2006.
(SIP)", draft-ietf-sip-record-route-fix-05 (work in
progress), October 2008.
[I-D.ietf-sip-subnot-etags] [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A
Niemi, A., "An Extension to Session Initiation Protocol Session Initiation Protocol (SIP) Event Package for
(SIP) Events for Conditional Event Notification", Conference State", RFC 4575, August 2006.
draft-ietf-sip-subnot-etags-03 (work in progress),
July 2008.
[I-D.ietf-sip-sips] [RFC4579] Johnston, A. and O. Levin, "Session Initiation
Audet, F., "The use of the SIPS URI Scheme in the Session Protocol (SIP) Call Control - Conferencing for User
Initiation Protocol (SIP)", draft-ietf-sip-sips-08 (work Agents", BCP 119, RFC 4579, August 2006.
in progress), February 2008.
[RFC4896] Surtees, A., West, M., and A. Roach, "Signaling [RFC4583] Camarillo, G., "Session Description Protocol (SDP)
Compression (SigComp) Corrections and Clarifications", Format for Binary Floor Control Protocol (BFCP)
RFC 4896, June 2007. Streams", RFC 4583, November 2006.
[RFC5049] Bormann, C., Liu, Z., Price, R., and G. Camarillo, [RFC4662] Roach, A., Campbell, B., and J. Rosenberg, "A
"Applying Signaling Compression (SigComp) to the Session Session Initiation Protocol (SIP) Event
Initiation Protocol (SIP)", RFC 5049, December 2007. Notification Extension for Resource Lists",
RFC 4662, August 2006.
[I-D.ietf-simple-simple] [RFC4730] Burger, E. and M. Dolly, "A Session Initiation
Rosenberg, J., "SIMPLE made Simple: An Overview of the Protocol (SIP) Event Package for Key Press Stimulus
IETF Specifications for Instant Messaging and Presence (KPML)", RFC 4730, November 2006.
using the Session Initiation Protocol (SIP)",
draft-ietf-simple-simple-03 (work in progress), July 2008.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for
RFC 4960, September 2007. DTMF Digits, Telephony Tones, and Telephony
Signals", RFC 4733, December 2006.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session
Carrara, "Key Management Extensions for Session Description Protocol (SDP) Content Attribute",
Description Protocol (SDP) and Real Time Streaming RFC 4796, February 2007.
Protocol (RTSP)", RFC 4567, July 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4896] Surtees, A., West, M., and A. Roach, "Signaling
Description Protocol (SDP) Security Descriptions for Media Compression (SigComp) Corrections and
Streams", RFC 4568, July 2006. Clarifications", RFC 4896, June 2007.
[I-D.ietf-sip-dtls-srtp-framework] [RFC4916] Elwell, J., "Connected Identity in the Session
Fischl, J., Tschofenig, H., and E. Rescorla, "Framework Initiation Protocol (SIP)", RFC 4916, June 2007.
for Establishing an SRTP Security Context using DTLS",
draft-ietf-sip-dtls-srtp-framework-05 (work in progress),
October 2008.
[I-D.ietf-ecrit-framework] [RFC4960] Stewart, R., "Stream Control Transmission
Rosen, B., Schulzrinne, H., Polk, J., and A. Newton, Protocol", RFC 4960, September 2007.
"Framework for Emergency Calling using Internet
Multimedia", draft-ietf-ecrit-framework-06 (work in
progress), July 2008.
[RFC2833] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF [RFC5027] Andreasen, F. and D. Wing, "Security Preconditions
Digits, Telephony Tones and Telephony Signals", RFC 2833, for Session Description Protocol (SDP) Media
May 2000. Streams", RFC 5027, October 2007.
[RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session [RFC5049] Bormann, C., Liu, Z., Price, R., and G. Camarillo,
Initiation Protocol (SIP) URIs for Applications such as "Applying Signaling Compression (SigComp) to the
Voicemail and Interactive Voice Response (IVR)", RFC 4458, Session Initiation Protocol (SIP)", RFC 5049,
April 2006. December 2007.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. [RFC5079] Rosenberg, J., "Rejecting Anonymous Requests in the
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, Session Initiation Protocol (SIP)", RFC 5079,
August 2004. December 2007.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC5360] Rosenberg, J., Camarillo, G., and D. Willis, "A
Security", RFC 4347, April 2006. Framework for Consent-Based Communications in the
Session Initiation Protocol (SIP)", RFC 5360,
October 2008.
[I-D.ietf-sipping-v6-transition] [RFC5361] Camarillo, G., "A Document Format for Requesting
Camarillo, G., "IPv6 Transition in the Session Initiation Consent", RFC 5361, October 2008.
Protocol (SIP)", draft-ietf-sipping-v6-transition-07 (work
in progress), August 2007.
[I-D.ietf-sipping-update-pai] [RFC5362] Camarillo, G., "The Session Initiation Protocol
Elwell, J., "Updates to Asserted Identity in the Session (SIP) Pending Additions Event Package", RFC 5362,
Initiation Protocol (SIP)",
draft-ietf-sipping-update-pai-07 (work in progress),
October 2008. October 2008.
[RFC3665] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and [RFC5363] Camarillo, G. and A. Roach, "Framework and Security
K. Summers, "Session Initiation Protocol (SIP) Basic Call Considerations for Session Initiation Protocol
Flow Examples", BCP 75, RFC 3665, December 2003. (SIP) URI-List Services", RFC 5363, October 2008.
[RFC3666] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and [RFC5365] Garcia-Martin, M. and G. Camarillo, "Multiple-
K. Summers, "Session Initiation Protocol (SIP) Public Recipient MESSAGE Requests in the Session
Switched Telephone Network (PSTN) Call Flows", BCP 76, Initiation Protocol (SIP)", RFC 5365, October 2008.
RFC 3666, December 2003.
[I-D.ietf-sip-ipv6-abnf-fix] [RFC5366] Camarillo, G. and A. Johnston, "Conference
Gurbani, V., Carpenter, B., and B. Tate, "Essential Establishment Using Request-Contained Lists in the
correction for IPv6 ABNF and URI comparison in RFC3261", Session Initiation Protocol (SIP)", RFC 5366,
draft-ietf-sip-ipv6-abnf-fix-02 (work in progress), October 2008.
May 2008.
[RFC4497] Elwell, J., Derks, F., Mourot, P., and O. Rousseau, [RFC5367] Camarillo, G., Roach, A., and O. Levin,
"Interworking between the Session Initiation Protocol "Subscriptions to Request-Contained Resource Lists
(SIP) and QSIG", BCP 117, RFC 4497, May 2006. in the Session Initiation Protocol (SIP)",
RFC 5367, October 2008.
[I-D.ietf-sip-ua-privacy] [RFC5368] Camarillo, G., Niemi, A., Isomaki, M., Garcia-
Munakata, M., Schubert, S., and T. Ohba, "UA-Driven Martin, M., and H. Khartabil, "Referring to
Privacy Mechanism for SIP", draft-ietf-sip-ua-privacy-03 Multiple Resources in the Session Initiation
(work in progress), October 2008. Protocol (SIP)", RFC 5368, October 2008.
[I-D.ietf-sip-body-handling] [RFC5373] Willis, D. and A. Allen, "Requesting Answering
Camarillo, G., "Message Body Handling in the Session Modes for the Session Initiation Protocol (SIP)",
Initiation Protocol (SIP)", RFC 5373, November 2008.
draft-ietf-sip-body-handling-04 (work in progress),
October 2008.
[I-D.ietf-sip-domain-certs] [RTCP-SUM] Clark, A., Pendleton, A., Johnston, A., and H.
Gurbani, V., Lawrence, S., and B. Laboratories, "Domain Sinnreich, "Session Initiation Protocol Package for
Certificates in the Session Initiation Protocol (SIP)", Voice Quality Reporting Event", Work in Progress,
draft-ietf-sip-domain-certs-02 (work in progress),
October 2008. October 2008.
[I-D.ietf-sipping-gruu-reg-event] [SAML] Tschofenig, H., Hodges, J., Peterson, J., Polk, J.,
Kyzivat, P., "Registration Event Package Extension for and D. Sicker, "SIP SAML Profile and Binding", Work
Session Initiation Protocol (SIP) Globally Routable User in Progress, November 2008.
Agent URIs (GRUUs)", draft-ietf-sipping-gruu-reg-event-09
(work in progress), July 2007.
[I-D.ietf-sip-session-policy-framework] [SDP-CAP] Andreasen, F., "SDP Capability Negotiation", Work
Hilt, V., Camarillo, G., and J. Rosenberg, "A Framework in Progress, July 2008.
for Session Initiation Protocol (SIP) Session Policies",
draft-ietf-sip-session-policy-framework-04 (work in
progress), August 2008.
[I-D.ietf-mmusic-qos-identification] [SDP-MEDIA] Gilman, R., Even, R., and F. Andreasen, "SDP media
Polk, J., Dhesikan, S., and G. Camarillo, "Quality of capabilities Negotiation", Work in Progress,
Service (QoS) Mechanism Selection in the Session July 2008.
Description Protocol (SDP)",
draft-ietf-mmusic-qos-identification-02 (work in
progress), October 2008.
[RFC5363] Camarillo, G. and A. Roach, "Framework and Security [SESSION-POLICY] Hilt, V., Camarillo, G., and J. Rosenberg, "A
Considerations for Session Initiation Protocol (SIP) URI- Framework for Session Initiation Protocol (SIP)
List Services", RFC 5363, October 2008. Session Policies", Work in Progress, November 2008.
[I-D.ietf-sip-connect-reuse] [SIMPLE] Rosenberg, J., "SIMPLE made Simple: An Overview of
Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse in the IETF Specifications for Instant Messaging and
the Session Initiation Protocol (SIP)", Presence using the Session Initiation Protocol
draft-ietf-sip-connect-reuse-12 (work in progress), (SIP)", Work in Progress, October 2008.
October 2008.
[I-D.ietf-mmusic-sdp-dtls] [SIPS-URI] Audet, F., "The Use of the SIPS URI Scheme in the
Fischl, J. and H. Tschofenig, "Session Description Session Initiation Protocol (SIP)", Work
Protocol (SDP) Indicators for Datagram Transport Layer in Progress, November 2008.
Security (DTLS)", draft-ietf-mmusic-sdp-dtls-00 (work in
progress), January 2008.
[RFC5361] Camarillo, G., "A Document Format for Requesting Consent", [SRTP-FRAME] Fischl, J., Tschofenig, H., and E. Rescorla,
RFC 5361, October 2008. "Framework for Establishing an SRTP Security
Context using DTLS", Work in Progress,
October 2008.
[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol [SUBNOT-ETAGS] Niemi, A., "An Extension to Session Initiation
(SIP) Call Control - Conferencing for User Agents", Protocol (SIP) Events for Conditional Event
BCP 119, RFC 4579, August 2006. Notification", Work in Progress, July 2008.
[I-D.ietf-sip-location-conveyance] [TRANSFER-MECH] Garcia, M., Isomaki, M., Camarillo, G., Loreto, S.,
Polk, J. and B. Rosen, "Location Conveyance for the and P. Kyzivat, "A Session Description Protocol
Session Initiation Protocol", (SDP) Offer/Answer Mechanism to Enable File
draft-ietf-sip-location-conveyance-10 (work in progress), Transfer", Work in Progress, November 2008.
September 2008.
[RFC3310] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext Transfer [UA-PRIVACY] Munakata, M., Schubert, S., and T. Ohba, "UA-Driven
Protocol (HTTP) Digest Authentication Using Authentication Privacy Mechanism for SIP", Work in Progress,
and Key Agreement (AKA)", RFC 3310, September 2002. October 2008.
[RFC4169] Torvinen, V., Arkko, J., and M. Naslund, "Hypertext [UPDATE-PAI] Elwell, J., "Updates to Asserted Identity in the
Transfer Protocol (HTTP) Digest Authentication Using Session Initiation Protocol (SIP)", Work
Authentication and Key Agreement (AKA) Version-2", in Progress, October 2008.
RFC 4169, November 2005.
Author's Address Author's Address
Jonathan Rosenberg Jonathan Rosenberg
Cisco Cisco
Iselin, NJ Iselin, NJ
US US
Email: jdrosen@cisco.com EMail: jdrosen@cisco.com
URI: http://www.jdrosen.net URI: http://www.jdrosen.net
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