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RFC 5411

Informational
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A Hitchhiker's Guide to the Session Initiation Protocol (SIP)

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Network Working Group                                       J. Rosenberg
Request for Comments: 5411                                         Cisco
Category: Informational                                     January 2009


     A Hitchhiker's Guide to the Session Initiation Protocol (SIP)

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

Abstract

   The Session Initiation Protocol (SIP) is the subject of numerous
   specifications that have been produced by the IETF.  It can be
   difficult to locate the right document, or even to determine the set
   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
   specifications under the SIP umbrella, briefly summarizes each, and
   groups them into categories.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2.  Scope of This Document . . . . . . . . . . . . . . . . . . . .  4
   3.  Core SIP Specifications  . . . . . . . . . . . . . . . . . . .  5
   4.  Public Switched Telephone Network (PSTN) Interworking  . . . .  8
   5.  General Purpose Infrastructure Extensions  . . . . . . . . . . 10
   6.  NAT Traversal  . . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Call Control Primitives  . . . . . . . . . . . . . . . . . . . 13
   8.  Event Framework  . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 15
   10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 16
   11. Operations and Management  . . . . . . . . . . . . . . . . . . 17
   12. SIP Compression  . . . . . . . . . . . . . . . . . . . . . . . 17
   13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 17
   14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 19
   15. Security Mechanisms  . . . . . . . . . . . . . . . . . . . . . 20
   16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 23
   17. Instant Messaging, Presence, and Multimedia  . . . . . . . . . 24
   18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 25
   19. Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   20. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   21. Informative References . . . . . . . . . . . . . . . . . . . . 26

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1.  Introduction

   The Session Initiation Protocol (SIP) [RFC3261] is the subject of
   numerous specifications that have been produced by the IETF.  It can
   be difficult to locate the right document, or even to determine the
   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
   current snapshot of the specifications under the SIP umbrella at the
   time of publication.  It is anticipated that this document itself
   will be regularly updated as SIP specifications mature.  Furthermore,
   it references many specifications, which, at the time of publication
   of this document, were not yet finalized, and may eventually be
   completed or abandoned.  Therefore, the enumeration of specifications
   here is a work-in-progress and subject to change.

   For each specification, a paragraph or so description is included
   that summarizes the purpose of the specification.  Each specification
   also includes a letter that designates its category in the Standards
   Track [RFC2026].  These values are:

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

   E: Experimental

   B: Best Current Practice

   I: Informational

   The specifications are grouped together by topic.  The topics are:

   Core:  The SIP specifications that are expected to be utilized for
      each session or registration an endpoint participates in.

   Public Switched Telephone Network (PSTN) Interop:  Specifications
      related to interworking with the telephone network.

   General Purpose Infrastructure:  General purpose extensions to SIP,
      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
      traversal.

   Call Control Primitives:  Specifications for manipulating SIP dialogs
      and calls.

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   Event Framework:  Definitions of the core specifications for the SIP
      event framework, providing for pub/sub capability.

   Event Packages:  Packages that utilize the SIP event framework.

   Quality of Service:  Specifications related to multimedia quality of
      service (QoS).

   Operations and Management:  Specifications related to configuration
      and monitoring of SIP deployments.

   SIP Compression:  Specifications to facilitate usage of SIP with the
      Signaling Compression (Sigcomp) framework.

   SIP Service URIs:  Specifications on how to use SIP URIs to address
      multimedia services.

   Minor Extensions:  Specifications that solve a narrow problem space
      or provide an optimization.

   Security Mechanisms:  Specifications providing security functionality
      for SIP.

   Conferencing:  Specifications for multimedia conferencing.

   Instant Messaging, Presence, and Multimedia:  SIP extensions related
      to IM, presence, and multimedia.  This covers only the SIP
      extensions related to these topics.  See [SIMPLE] for a full
      treatment of SIP for IM and Presence (SIMPLE).

   Emergency Services:  SIP extensions related to emergency services.
      See [ECRIT-FRAME] for a more complete treatment of additional
      functionality related to emergency services.

   Typically, SIP extensions fit naturally into topic areas, and
   implementors interested in a particular topic often implement many or
   all of the specifications in that area.  There are some
   specifications that fall into multiple topic areas, in which case
   they are listed more than once.

   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:
   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
   SIP.  It is an informational document, meant to guide newcomers,
   implementors, and deployers to the many specifications associated
   with SIP.

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2.  Scope of This Document

   It is very difficult to enumerate the set of SIP specifications.
   This is because there are many protocols that are intimately related
   to SIP and used by nearly all SIP implementations, but are not
   formally SIP extensions.  As such, this document formally defines a
   "SIP specification" as:

   o  RFC 3261 and any specification that defines an extension to it,
      where an extension is a mechanism that changes or updates in some
      way a behavior specified there.

   o  The basic SDP specification [RFC4566] and any specification that
      defines an extension to SDP whose primary purpose is to support
      SIP.

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

   Excluded from this list are requirements, architectures, registry
   definitions, non-normative frameworks, and processes.  Best Current
   Practices are included when they normatively define mechanisms for
   accomplishing a task, or provide significant description of the usage
   of the normative specifications, such as call flows.

   The SIP change process [RFC3427] defines two types of extensions to
   SIP: normal extensions and the so-called P-headers (where P stands
   for "preliminary", "private", or "proprietary", and the "P-" prefix
   is included in the header field name), which are meant to be used in
   areas of limited applicability.  P-headers cannot be defined in the
   Standards Track.  For the most part, P-headers are not included in
   the listing here, with the exception of those that have seen general
   usage despite their P-header status.

   This document includes specifications, which have already been
   approved by the IETF and granted an RFC number, in addition to
   Internet Drafts, which are still under development within the IETF
   and will eventually finish and get an RFC number.  Inclusion of
   Internet Drafts here helps encourage early implementation and
   demonstrations of interoperability of the protocol, and thus aids in
   the standards-setting process.  Inclusion of these also identifes
   where the IETF is targetting a solution at a particular problem
   space.  Note that final IANA assignment of codepoints (such as option
   tags and header field names) does not take place until shortly before
   publication as an RFC, and thus codepoint assignments may change.

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3.  Core SIP Specifications

   The core SIP specifications represent the set of specifications whose
   functionality is broadly applicable.  An extension is broadly
   applicable if it fits into one of the following categories:

   o  For specifications that impact SIP session management, the
      extension would be used for almost every session initiated by a
      user agent.

   o  For specifications that impact SIP registrations, the extension
      would be used for almost every registration initiated by a user
      agent.

   o  For specifications that impact SIP subscriptions, the extension
      would be used for almost every subscription initiated by a user
      agent.

   In other words, these are not specifications that are used just for
   some requests and not others; they are specifications that would
   apply to each and every request for which the extension is relevant.
   In the galaxy of SIP, these specifications are like towels [HGTTG].

   RFC 3261, The Session Initiation Protocol (S):  [RFC3261] is the core
      SIP protocol itself.  RFC 3261 obsoletes [RFC2543].  It is the
      president of the galaxy [HGTTG] as far as the suite of SIP
      specifications is concerned.

   RFC 3263, Locating SIP Servers (S):  [RFC3263] provides DNS
      procedures for taking a SIP URI and determining a SIP server that
      is associated with that SIP URI.  RFC 3263 is essential for any
      implementation using SIP with DNS.  RFC 3263 makes use of both DNS
      SRV records [RFC2782] and NAPTR records [RFC3401].

   RFC 3264, An Offer/Answer Model with the Session Description Protocol
   (S):  [RFC3264] defines how the Session Description Protocol (SDP)
      [RFC4566] is used with SIP to negotiate the parameters of a media
      session.  It is in widespread usage and an integral part of the
      behavior of RFC 3261.

   RFC 3265, SIP-Specific Event Notification (S):  [RFC3265] defines the
      SUBSCRIBE and NOTIFY methods.  These two methods provide a general
      event notification framework for SIP.  To actually use the
      framework, extensions need to be defined for specific event
      packages.  An event package defines a schema for the event data
      and describes other aspects of event processing specific to that
      schema.  An RFC 3265 implementation is required when any event
      package is used.

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   RFC 3325, Private Extensions to SIP for Asserted Identity within
   Trusted Networks (I):  Though its P-header status implies that it has
      limited applicability, [RFC3325], which defines the P-Asserted-
      Identity header field, has been widely deployed.  It is used as
      the basic mechanism for providing network-asserted caller ID
      services.  Its intended update, [UPDATE-PAI], clarifies its usage
      for connected party identification as well.

   RFC 3327, SIP Extension Header Field for Registering Non-Adjacent
   Contacts (S):  [RFC3327] defines the Path header field.  This field
      is inserted by proxies between a client and their registrar.  It
      allows inbound requests towards that client to traverse these
      proxies prior to being delivered to the user agent.  It is
      essential in any SIP deployment that has edge proxies, which are
      proxies between the client and the home proxy or SIP registrar.

   RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
      [RFC3581] defines the rport parameter of the Via header.  It
      allows SIP responses to traverse NAT.  It is one of several
      specifications that are utilized for NAT traversal (see
      Section 6).

   RFC 3840, Indicating User Agent Capabilities in SIP (S):  [RFC3840]
      defines a mechanism for carrying capability information about a
      user agent in REGISTER requests and in dialog-forming requests
      like INVITE.  It has found use with conferencing (the isfocus
      parameter declares that a user agent is a conference server) and
      with applications like push-to-talk.

   RFC 4320, Actions Addressing Issues Identified with the Non-INVITE
   Transaction in SIP (S):  [RFC4320] formally updates RFC 3261 and
      modifies some of the behaviors associated with non-INVITE
      transactions.  This addresses some problems found in timeout and
      failure cases.

   RFC 4474, Enhancements for Authenticated Identity Management in SIP
   (S):  [RFC4474] defines a mechanism for providing a cryptographically
      verifiable identity of the calling party in a SIP request.  Known
      as "SIP Identity", this mechanism provides an alternative to RFC
      3325.  It has seen little deployment so far, but its importance as
      a key construct for anti-spam techniques and new security
      mechanisms makes it a core part of the SIP specifications.

   GRUU, Obtaining and Using Globally Routable User Agent Identifiers
   (GRUU) in SIP (S):  [GRUU] defines a mechanism for directing requests
      towards a specific UA instance.  GRUU is essential for features
      like transfer and provides another piece of the SIP NAT traversal
      story.

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   OUTBOUND, Managing Client Initiated Connections through SIP (S):
      [OUTBOUND], also known as SIP outbound, defines important changes
      to the SIP registration mechanism that enable delivery of SIP
      messages towards a UA when it is behind a NAT.  This specification
      is the cornerstone of the SIP NAT traversal strategy.

   RFC 4566, Session Description Protocol (S):  [RFC4566] defines a
      format for representing multimedia sessions.  SDP objects are
      carried in the body of SIP messages and, based on the offer/answer
      model, are used to negotiate the media characteristics of a
      session between users.

   SDP-CAP, SDP Capability Negotiation (S):  [SDP-CAP] defines a set of
      extensions to SDP that allows for capability negotiation within
      SDP.  Capability negotiation can be used to select between
      different profiles of RTP (secure vs. unsecure) or to negotiate
      codecs such that an agent has to select one amongst a set of
      supported codecs.

   ICE, Interactive Connectivity Establishment (ICE) (S):  [ICE] defines
      a technique for NAT traversal of media sessions for protocols that
      make use of the offer/answer model.  This specification is the
      IETF-recommended mechanism for NAT traversal for SIP media
      streams, and is meant to be used even by endpoints that are
      themselves never behind a NAT.  A SIP option tag and media feature
      tag [OPTION-TAG] (also a core specification) have been defined for
      use with ICE.

   RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
   Description Protocol (SDP) (S):  [RFC3605] defines a way to
      explicitly signal, within an SDP message, the IP address and port
      for RTCP, rather than using the port+1 rule in the Real Time
      Transport Protocol (RTP) [RFC3550].  It is needed for devices
      behind NAT, and the specification is required by ICE.

   RFC 4916, Connected Identity in the Session Initiation Protocol (SIP)
   (S):  [RFC4916] formally updates RFC 3261.  It defines an extension
      to SIP that allows a calling user to determine the identity of the
      final called user (connected party).  Due to forwarding and
      retargeting services, this may not be the same as the user that
      the caller was originally trying to reach.  The mechanism works in
      tandem with the SIP identity specification [RFC4474] to provide
      signatures over the connected party identity.  It can also be used
      if a party identity changes mid-call due to third-party call
      control actions or PSTN behavior.

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   RFC 3311, The SIP UPDATE Method (S):  [RFC3311] defines the UPDATE
      method for SIP.  This method is meant as a means for updating
      session information prior to the completion of the initial INVITE
      transaction.  It can also be used to update other information,
      such as the identity of the participant [RFC4916], without
      involving an updated offer/answer exchange.  It was developed
      initially to support [RFC3312], but has found other uses.  In
      particular, its usage with RFC 4916 means it will typically be
      used as part of every session, to convey a secure, connected
      identity.

   SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
   Protocol (SIP) (S):  [SIPS-URI] is intended to update RFC 3261.  It
      revises the processing of the SIPS URI, originally defined in RFC
      3261, to fix many errors and problems that have been encountered
      with that mechanism.

   RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples
   (B):  [RFC3665] contains best-practice call flow examples for basic
      SIP interactions -- call establishment, termination, and
      registration.

   Essential Corrections to SIP:  A collection of fixes to SIP that
      address important bugs and vulnerabilities.  These include a fix
      requiring loop detection in any proxy that forks [LOOP-FIX], a
      clarification on how record-routing works [RECORD-ROUTE], and a
      correction to the IPv6 BNF [ABNF-FIX].

4.  Public Switched Telephone Network (PSTN) Interworking

   Numerous extensions and usages of SIP are related to interoperability
   and communications with or through the PSTN.

   RFC 2848, The PINT Service Protocol (S):  [RFC2848] is one of the
      earliest extensions to SIP.  It defines procedures for using SIP
      to invoke services that actually execute on the PSTN.  Its main
      application is for third-party call control, allowing an IP host
      to set up a call between two PSTN endpoints.  PINT (PSTN/Internet
      Interworking) has a relatively narrow focus and has not seen
      widespread deployment.

   RFC 3910, The SPIRITS Protocol (S):  Continuing the trend of naming
      PSTN-related extensions with alcohol references, SPIRITS (Services
      in PSTN Requesting Internet Services) [RFC3910] defines the
      inverse of PINT.  It allows a switch in the PSTN to ask an IP
      element how to proceed with call waiting.  It was developed
      primarily to support Internet Call Waiting (ICW).  Perhaps the
      next specification will be called the Pan Galactic Gargle Blaster

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      [HGTTG].

   RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I):
      SIP-T [RFC3372] defines a mechanism for using SIP between pairs of
      PSTN gateways.  Its essential idea is to tunnel ISDN User Part
      (ISUP) signaling between the gateways in the body of SIP messages.
      SIP-T motivated the development of INFO [RFC2976].  SIP-T has seen
      widespread implementation for the limited deployment model that it
      addresses.  As ISUP endpoints disappear from the network, the need
      for this mechanism will decrease.

   RFC 3398, ISUP to SIP Mapping (S):  [RFC3398] defines how to do
      protocol mapping from the SS7 ISDN User Part (ISUP) signaling to
      SIP.  It is widely used in SS7 to SIP gateways and is part of the
      SIP-T framework.

   RFC 4497, Interworking between the Session Initiation Protocol (SIP)
   and QSIG (B):  [RFC4497] defines how to do protocol mapping from
      Q.SIG, used for Private Branch Exchange (PBX) signaling, to SIP.

   RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S):  [RFC3578]
      defines a mechanism to map overlap dialing into SIP.  This
      specification is widely regarded as the ugliest SIP specification,
      as the introduction to the specification itself advises that it
      has many problems.  Overlap signaling (the practice of sending
      digits into the network as dialed instead of waiting for complete
      collection of the called party number) is largely incompatible
      with SIP at some fairly fundamental levels.  That said, RFC 3578
      is mostly harmless and has seen some usage.

   RFC 3960, Early Media and Ringtone Generation in SIP (I):  [RFC3960]
      defines some guidelines for handling early media -- the practice
      of sending media from the called party or an application server
      towards the caller prior to acceptance of the call.  Early media
      is often generated from the PSTN.  Early media is a complex topic,
      and this specification does not fully address the problems
      associated with it.

   RFC 3959, Early Session Disposition Type for the Session Initiation
   Protocol (SIP) (S):  [RFC3959] defines a new session disposition type
      for use with early media.  It indicates that the SDP in the body
      is for a special early media session.  This has seen little usage.

   RFC 3204, MIME Media Types for ISUP and QSIG Objects (S):  [RFC3204]
      defines MIME objects for representing SS7 and QSIG signaling
      messages.  SS7 signaling messages are carried in the body of SIP
      messages when SIP-T is used.  QSIG signaling messages can be
      carried in a similar way.

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   RFC3666, Session Initiation Protocol (SIP) Public Switched Telephone
   Network (PSTN) Call Flows (B):  [RFC3666] provides best practice call
      flows around interworking with the PSTN.

5.  General Purpose Infrastructure Extensions

   These extensions are general purpose enhancements to SIP, SDP, and
   MIME that can serve a wide variety of uses.  However, they are not
   used for every session or registration, as the core specifications
   are.

   RFC 3262, Reliability of Provisional Responses in SIP (S):  SIP
      defines two types of responses to a request: final and
      provisional.  Provisional responses are numbered from 100 to 199.
      In SIP, these responses are not sent reliably.  This choice was
      made in RFC 2543 since the messages were meant to just be truly
      informational and rendered to the user.  However, subsequent work
      on PSTN interworking demonstrated a need to map provisional
      responses to PSTN messages that needed to be sent reliably.
      [RFC3262] was developed to allow reliability of provisional
      responses.  The specification defines the PRACK method, used for
      indicating that a provisional response was received.  Though it
      provides a generic capability for SIP, RFC 3262 implementations
      have been most common in PSTN interworking devices.  However,
      PRACK brings a great deal of complication for relatively small
      benefit.  As such, it has seen only moderate levels of deployment.

   RFC 3323, A Privacy Mechanism for the Session Initiation Protocol
   (SIP) (S):  [RFC3323] defines the Privacy header field, used by
      clients to request anonymity for their requests.  Though it
      defines several privacy services, the only one broadly used is the
      one that supports privacy of the P-Asserted-Identity header field
      [RFC3325].

   UA-PRIVACY, UA-Driven Privacy Mechanism for SIP (S):  [UA-PRIVACY]
      defines a mechanism for achieving anonymous calls in SIP.  It is
      an alternative to [RFC3323], and instead places more intelligence
      in the endpoint to craft anonymous messages by directly accessing
      network services.

   RFC 2976, The INFO Method (S):  [RFC2976] was defined as an extension
      to RFC 2543.  It defines a method, INFO, used to transport mid-
      dialog information that has no impact on SIP itself.  Its driving
      application was the transport of PSTN-related information when
      using SIP between a pair of gateways.  Though originally conceived
      for broader use, it only found standardized usage with SIP-T
      [RFC3372].  It has been used to support numerous proprietary and
      non-interoperable extensions due to its poorly defined scope.

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   RFC 3326, The Reason Header Field for SIP (S):  [RFC3326] defines the
      Reason header field.  It is used in requests, such as BYE, to
      indicate the reason that the request is being sent.

   RFC 3388, Grouping of Media Lines in the Session Description Protocol
   (S):  RFC 3388 [RFC3388] defines a framework for grouping together
      media streams in an SDP message.  Such a grouping allows
      relationships between these streams, such as which stream is the
      audio for a particular video feed, to be expressed.

   RFC 3420, Internet Media Type message/sipfrag (S):  [RFC3420] defines
      a MIME object that contains a SIP message fragment.  Only certain
      header fields and parts of the SIP message are present.  For
      example, it is used to report back on the responses received to a
      request sent as a consequence of a REFER.

   RFC 3608, SIP Extension Header Field for Service Route Discovery
   During Registration (S):  [RFC3608] allows a client to determine,
      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
      verify the Route header in client-initiated requests.  In many
      respects, it is the inverse of the Path header field, but has seen
      less usage since default outbound proxies have been sufficient in
      many deployments.

   RFC 3841, Caller Preferences for SIP (S):  [RFC3841] defines a set of
      headers that a client can include in a request to control the way
      in which the request is routed downstream.  It allows a client to
      direct a request towards a UA with specific capabilities, which a
      UA indicates using [RFC3840].

   RFC 4028, Session Timers in SIP (S):  [RFC4028] defines a keepalive
      mechanism for SIP signaling.  It is primarily meant to provide a
      way to clean up old state in proxies that are holding call state
      for calls from failed endpoints that were never terminated
      normally.  Despite its name, the session timer is not a mechanism
      for detecting a network failure mid-call.  Session timers
      introduce a fair bit of complexity for relatively little gain, and
      have seen moderate deployment.

   RFC 4168, SCTP as a Transport for SIP (S):  [RFC4168] defines how to
      carry SIP messages over the Stream Control Transmission Protocol
      (SCTP) [RFC4960].  SCTP has seen very limited usage for SIP
      transport.

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   RFC 4244, An Extension to SIP for Request History Information (S):
      [RFC4244] defines the History-Info header field, which indicates
      information on how and why a call came to be routed to a
      particular destination.

   RFC 4145, TCP-Based Media Transport in the Session Description
   Protocol (SDP) (S):  [RFC4145] defines an extension to SDP for
      setting up TCP-based sessions between user agents.  It defines who
      sets up the connection and how its lifecycle is managed.  It has
      seen relatively little usage due to the small number of media
      types to date that use TCP.

   RFC 4091, The Alternative Network Address Types (ANAT) Semantics for
   the Session Description Protocol (SDP) Grouping Framework (S):
      [RFC4091] defines a mechanism for including both IPv4 and IPv6
      addresses for a media session as alternates.  This mechanism has
      been deprecated in favor of ICE [ICE].

   SDP-MEDIA, SDP Media Capabilities Negotiation (S):  [SDP-MEDIA]
      defines an extension to the SDP capability negotiation framework
      [SDP-CAP] for negotiating codecs, codec parameters, and media
      streams.

   BODY-HANDLING, Message Body Handling in the Session Initiation
   Protocol (SIP):  [BODY-HANDLING] clarifies handling of bodies in SIP,
      focusing primarily on multi-part behavior, which was under-
      specified in SIP.

6.  NAT Traversal

   These SIP extensions are primarily aimed at addressing NAT traversal
   for SIP.

   ICE, Interactive Connectivity Establishment (ICE) (S):  [ICE] defines
      a technique for NAT traversal of media sessions for protocols that
      make use of the offer/answer model.  This specification is the
      IETF-recommended mechanism for NAT traversal for SIP media
      streams, and is meant to be used even by endpoints that are
      themselves never behind a NAT.  A SIP option tag and media feature
      tag [OPTION-TAG] have been defined for use with ICE.

   ICE-TCP, TCP Candidates with Interactive Connectivity Establishment
   (ICE) (S):  [ICE-TCP] specifies the usage of ICE for TCP streams.
      This allows for selection of RTP-based voice on top of TCP only
      when NAT or firewalls would prevent UDP-based voice from working.

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   RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session
   Description Protocol (SDP) (S):  [RFC3605] defines a way to
      explicitly signal, within an SDP message, the IP address and port
      for RTCP, rather than using the port+1 rule in the Real Time
      Transport Protocol (RTP) [RFC3550].  It is needed for devices
      behind NAT, and the specification is required by ICE.

   OUTBOUND, Managing Client Initiated Connections through SIP (S):
      [OUTBOUND], also known as SIP outbound, defines important changes
      to the SIP registration mechanism that enable delivery of SIP
      messages towards a UA when it is behind a NAT.

   RFC 3581, An Extension to SIP for Symmetric Response Routing (S):
      [RFC3581] defines the rport parameter of the Via header.  It
      allows SIP responses to traverse NAT.

   GRUU, Obtaining and Using Globally Routable User Agent Identifiers
   (GRUU) in SIP (S):  [GRUU] defines a mechanism for directing requests
      towards a specific UA instance.  GRUU is essential for features
      like transfer and provides another piece of the SIP NAT traversal
      story.

7.  Call Control Primitives

   Numerous SIP extensions provide a toolkit of dialog- and call-
   management techniques.  These techniques have been combined together
   to build many SIP-based services.

   RFC 3515, The REFER Method (S):  REFER [RFC3515] defines a mechanism
      for asking a user agent to send a SIP request.  It's a form of SIP
      remote control, and is the primary tool used for call transfer in
      SIP.  Beware that not all potential uses of REFER (neither for all
      methods nor for all URI schemes) are well defined.  Implementors
      should only use the well-defined ones, and should not second guess
      or freely assume behavior for the others to avoid unexpected
      behavior of remote UAs, interoperability issues, and other bad
      surprises.

   RFC 3725, Best Current Practices for Third Party Call Control (3pcc)
   (B):  [RFC3725] defines a number of different call flows that allow
      one SIP entity, called the controller, to create SIP sessions
      amongst other SIP user agents.

   RFC 3911, The SIP Join Header Field (S):  [RFC3911] defines the Join
      header field.  When sent in an INVITE, it causes the recipient to
      join the resulting dialog into a conference with another dialog in
      progress.

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   RFC 3891, The SIP Replaces Header (S):  [RFC3891] defines a mechanism
      that allows a new dialog to replace an existing dialog.  It is
      useful for certain advanced transfer services.

   RFC 3892, The SIP Referred-By Mechanism (S):  [RFC3892] defines the
      Referred-By header field.  It is used in requests triggered by
      REFER, and provides the identity of the referring party to the
      referred-to party.

   RFC 4117, Transcoding Services Invocation in SIP Using Third Party
   Call Control (I):  [RFC4117] defines how to use 3pcc for the purposes
      of invoking transcoding services for a call.

8.  Event Framework

   RFC 3265, SIP-Specific Event Notification (S):  [RFC3265] defines the
      SUBSCRIBE and NOTIFY methods.  These two methods provide a general
      event notification framework for SIP.  To actually use the
      framework, extensions need to be defined for specific event
      packages.  An event package defines a schema for the event data
      and describes other aspects of event processing specific to that
      schema.  An RFC 3265 implementation is required when any event
      package is used.

   RFC 3903, SIP Extension for Event State Publication (S):  [RFC3903]
      defines the PUBLISH method.  It is not an event package, but is
      used by all event packages as a mechanism for pushing an event
      into the system.

   RFC 4662, A Session Initiation Protocol (SIP) Event Notification
   Extension for Resource Lists (S):  [RFC4662] defines an extension to
      RFC 3265 that allows a client to subscribe to a list of resources
      using a single subscription.  The server, called a Resource List
      Server (RLS), will "expand" the subscription and subscribe to each
      individual member of the list.  It has found applicability
      primarily in the area of presence, but can be used with any event
      package.

   SUBNOT-ETAGS, An Extension to Session Initiation Protocol  (SIP)
   Events for Conditional Event Notification (S):  [SUBNOT-ETAGS]
      defines an extension to RFC 3265 to optimize the performance of
      notifications.  When a client subscribes, it can indicate what
      version of a document it has so that the server can skip sending a
      notification if the client is up-to-date.  It is applicable to any
      event package.


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