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


     <span class="h1">A Hitchhiker's Guide to the Session Initiation Protocol (SIP)</span>

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

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





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<span class="grey"><a href="./rfc5411">RFC 5411</a>                Hitchhiker's Guide to SIP           January 2009</span>


<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   The Session Initiation Protocol (SIP) [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] 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!"  [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</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
   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 [<a href="./rfc2026" title="&quot;The Internet Standards Process -- Revision 3&quot;">RFC2026</a>].  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 [<a href="#ref-SIMPLE" title="&quot;SIMPLE made Simple: An Overview of the IETF Specifications for Instant Messaging and Presence using the Session Initiation Protocol (SIP)&quot;">SIMPLE</a>] for a full
      treatment of SIP for IM and Presence (SIMPLE).

   Emergency Services:  SIP extensions related to emergency services.
      See [<a href="#ref-ECRIT-FRAME" title="&quot;Framework for Emergency Calling using Internet Multimedia&quot;">ECRIT-FRAME</a>] 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 [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>].

   This document itself is not an update to <a href="./rfc3261">RFC 3261</a> 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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<span class="grey"><a href="./rfc5411">RFC 5411</a>                Hitchhiker's Guide to SIP           January 2009</span>


<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Scope of This Document</span>

   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  <a href="./rfc3261">RFC 3261</a> 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 [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] 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 [<a href="./rfc3427" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">RFC3427</a>] 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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<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Core SIP Specifications</span>

   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 [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>].

   <a href="./rfc3261">RFC 3261</a>, The Session Initiation Protocol (S):  [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] is the core
      SIP protocol itself.  <a href="./rfc3261">RFC 3261</a> obsoletes [<a href="./rfc2543" title="&quot;SIP: Session Initiation Protocol&quot;">RFC2543</a>].  It is the
      president of the galaxy [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>] as far as the suite of SIP
      specifications is concerned.

   <a href="./rfc3263">RFC 3263</a>, Locating SIP Servers (S):  [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>] provides DNS
      procedures for taking a SIP URI and determining a SIP server that
      is associated with that SIP URI.  <a href="./rfc3263">RFC 3263</a> is essential for any
      implementation using SIP with DNS.  <a href="./rfc3263">RFC 3263</a> makes use of both DNS
      SRV records [<a href="./rfc2782" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">RFC2782</a>] and NAPTR records [<a href="./rfc3401" title="&quot;Dynamic Delegation Discovery System (DDDS) Part One: The Comprehensive DDDS&quot;">RFC3401</a>].

   <a href="./rfc3264">RFC 3264</a>, An Offer/Answer Model with the Session Description Protocol
   (S):  [<a href="./rfc3264" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">RFC3264</a>] defines how the Session Description Protocol (SDP)
      [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] 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 <a href="./rfc3261">RFC 3261</a>.

   <a href="./rfc3265">RFC 3265</a>, SIP-Specific Event Notification (S):  [<a href="./rfc3265" title="&quot;Session Initiation Protocol (SIP)- Specific Event Notification&quot;">RFC3265</a>] 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 <a href="./rfc3265">RFC 3265</a> implementation is required when any event
      package is used.



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   <a href="./rfc3325">RFC 3325</a>, Private Extensions to SIP for Asserted Identity within
   Trusted Networks (I):  Though its P-header status implies that it has
      limited applicability, [<a href="./rfc3325" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">RFC3325</a>], 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, [<a href="#ref-UPDATE-PAI" title="&quot;Updates to Asserted Identity in the Session Initiation Protocol (SIP)&quot;">UPDATE-PAI</a>], clarifies its usage
      for connected party identification as well.

   <a href="./rfc3327">RFC 3327</a>, SIP Extension Header Field for Registering Non-Adjacent
   Contacts (S):  [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>] 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.

   <a href="./rfc3581">RFC 3581</a>, An Extension to SIP for Symmetric Response Routing (S):
      [<a href="./rfc3581" title="&quot;An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing&quot;">RFC3581</a>] 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
      <a href="#section-6">Section 6</a>).

   <a href="./rfc3840">RFC 3840</a>, Indicating User Agent Capabilities in SIP (S):  [<a href="./rfc3840" title="and P. Kyzivat">RFC3840</a>]
      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.

   <a href="./rfc4320">RFC 4320</a>, Actions Addressing Issues Identified with the Non-INVITE
   Transaction in SIP (S):  [<a href="./rfc4320" title="&quot;Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non- INVITE Transaction&quot;">RFC4320</a>] formally updates <a href="./rfc3261">RFC 3261</a> and
      modifies some of the behaviors associated with non-INVITE
      transactions.  This addresses some problems found in timeout and
      failure cases.

   <a href="./rfc4474">RFC 4474</a>, Enhancements for Authenticated Identity Management in SIP
   (S):  [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] 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 <a href="./rfc3325">RFC</a>
      <a href="./rfc3325">3325</a>.  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):  [<a href="#ref-GRUU" title="&quot;Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)&quot;">GRUU</a>] 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):
      [<a href="#ref-OUTBOUND" title="&quot;Managing Client Initiated Connections in the Session Initiation Protocol (SIP)&quot;">OUTBOUND</a>], 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.

   <a href="./rfc4566">RFC 4566</a>, Session Description Protocol (S):  [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] 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):  [<a href="#ref-SDP-CAP" title="&quot;SDP Capability Negotiation&quot;">SDP-CAP</a>] 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):  [<a href="#ref-ICE" title="&quot;Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;">ICE</a>] 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 [<a href="#ref-OPTION-TAG" title="&quot;Indicating Support for Interactive Connectivity Establishment (ICE) in the Session Initiation Protocol (SIP)&quot;">OPTION-TAG</a>] (also a core specification) have been defined for
      use with ICE.

   <a href="./rfc3605">RFC 3605</a>, Real Time Control Protocol (RTCP) Attribute in the Session
   Description Protocol (SDP) (S):  [<a href="./rfc3605" title="&quot;Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)&quot;">RFC3605</a>] 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) [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].  It is needed for devices
      behind NAT, and the specification is required by ICE.

   <a href="./rfc4916">RFC 4916</a>, Connected Identity in the Session Initiation Protocol (SIP)
   (S):  [<a href="./rfc4916" title="&quot;Connected Identity in the Session Initiation Protocol (SIP)&quot;">RFC4916</a>] formally updates <a href="./rfc3261">RFC 3261</a>.  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 [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] 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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   <a href="./rfc3311">RFC 3311</a>, The SIP UPDATE Method (S):  [<a href="./rfc3311" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">RFC3311</a>] 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 [<a href="./rfc4916" title="&quot;Connected Identity in the Session Initiation Protocol (SIP)&quot;">RFC4916</a>], without
      involving an updated offer/answer exchange.  It was developed
      initially to support [<a href="./rfc3312" title="&quot;Integration of Resource Management and Session Initiation Protocol (SIP)&quot;">RFC3312</a>], but has found other uses.  In
      particular, its usage with <a href="./rfc4916">RFC 4916</a> 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):  [<a href="#ref-SIPS-URI" title="&quot;The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)&quot;">SIPS-URI</a>] is intended to update <a href="./rfc3261">RFC 3261</a>.  It
      revises the processing of the SIPS URI, originally defined in <a href="./rfc3261">RFC</a>
      <a href="./rfc3261">3261</a>, to fix many errors and problems that have been encountered
      with that mechanism.

   <a href="./rfc3665">RFC 3665</a>, Session Initiation Protocol (SIP) Basic Call Flow Examples
   (B):  [<a href="./rfc3665" title="&quot;Session Initiation Protocol (SIP) Basic Call Flow Examples&quot;">RFC3665</a>] 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 [<a href="#ref-LOOP-FIX" title="&quot;Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies&quot;">LOOP-FIX</a>], a
      clarification on how record-routing works [<a href="#ref-RECORD-ROUTE" title="&quot;Addressing Record-Route issues in the Session Initiation Protocol (SIP)&quot;">RECORD-ROUTE</a>], and a
      correction to the IPv6 BNF [<a href="#ref-ABNF-FIX" title="&quot;Essential correction for IPv6 ABNF in RFC3261&quot;">ABNF-FIX</a>].

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Public Switched Telephone Network (PSTN) Interworking</span>

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

   <a href="./rfc2848">RFC 2848</a>, The PINT Service Protocol (S):  [<a href="./rfc2848" title="&quot;The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services&quot;">RFC2848</a>] 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.

   <a href="./rfc3910">RFC 3910</a>, The SPIRITS Protocol (S):  Continuing the trend of naming
      PSTN-related extensions with alcohol references, SPIRITS (Services
      in PSTN Requesting Internet Services) [<a href="./rfc3910" title="&quot;The SPIRITS (Services in PSTN requesting Internet Services) Protocol&quot;">RFC3910</a>] 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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      [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>].

   <a href="./rfc3372">RFC 3372</a>, SIP for Telephones (SIP-T): Context and Architectures (I):
      SIP-T [<a href="./rfc3372" title="&quot;Session Initiation Protocol for Telephones (SIP-T): Context and Architectures&quot;">RFC3372</a>] 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 [<a href="./rfc2976" title="&quot;The SIP INFO Method&quot;">RFC2976</a>].  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.

   <a href="./rfc3398">RFC 3398</a>, ISUP to SIP Mapping (S):  [<a href="./rfc3398" title="&quot;Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping&quot;">RFC3398</a>] 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.

   <a href="./rfc4497">RFC 4497</a>, Interworking between the Session Initiation Protocol (SIP)
   and QSIG (B):  [<a href="./rfc4497" title="&quot;Interworking between the Session Initiation Protocol (SIP) and QSIG&quot;">RFC4497</a>] defines how to do protocol mapping from
      Q.SIG, used for Private Branch Exchange (PBX) signaling, to SIP.

   <a href="./rfc3578">RFC 3578</a>, Mapping of ISUP Overlap Signaling to SIP (S):  [<a href="./rfc3578" title="&quot;Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)&quot;">RFC3578</a>]
      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, <a href="./rfc3578">RFC 3578</a>
      is mostly harmless and has seen some usage.

   <a href="./rfc3960">RFC 3960</a>, Early Media and Ringtone Generation in SIP (I):  [<a href="./rfc3960" title="&quot;Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)&quot;">RFC3960</a>]
      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.

   <a href="./rfc3959">RFC 3959</a>, Early Session Disposition Type for the Session Initiation
   Protocol (SIP) (S):  [<a href="./rfc3959" title="&quot;The Early Session Disposition Type for the Session Initiation Protocol (SIP)&quot;">RFC3959</a>] 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.

   <a href="./rfc3204">RFC 3204</a>, MIME Media Types for ISUP and QSIG Objects (S):  [<a href="./rfc3204" title="&quot;MIME media types for ISUP and QSIG Objects&quot;">RFC3204</a>]
      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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   <a href="./rfc3666">RFC3666</a>, Session Initiation Protocol (SIP) Public Switched Telephone
   Network (PSTN) Call Flows (B):  [<a href="./rfc3666" title="&quot;Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows&quot;">RFC3666</a>] provides best practice call
      flows around interworking with the PSTN.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  General Purpose Infrastructure Extensions</span>

   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.

   <a href="./rfc3262">RFC 3262</a>, 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 <a href="./rfc2543">RFC 2543</a> 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.
      [<a href="./rfc3262" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">RFC3262</a>] 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, <a href="./rfc3262">RFC 3262</a> 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.

   <a href="./rfc3323">RFC 3323</a>, A Privacy Mechanism for the Session Initiation Protocol
   (SIP) (S):  [<a href="./rfc3323" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">RFC3323</a>] 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
      [<a href="./rfc3325" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">RFC3325</a>].

   UA-PRIVACY, UA-Driven Privacy Mechanism for SIP (S):  [<a href="#ref-UA-PRIVACY" title="&quot;UA-Driven Privacy Mechanism for SIP&quot;">UA-PRIVACY</a>]
      defines a mechanism for achieving anonymous calls in SIP.  It is
      an alternative to [<a href="./rfc3323" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">RFC3323</a>], and instead places more intelligence
      in the endpoint to craft anonymous messages by directly accessing
      network services.

   <a href="./rfc2976">RFC 2976</a>, The INFO Method (S):  [<a href="./rfc2976" title="&quot;The SIP INFO Method&quot;">RFC2976</a>] was defined as an extension
      to <a href="./rfc2543">RFC 2543</a>.  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
      [<a href="./rfc3372" title="&quot;Session Initiation Protocol for Telephones (SIP-T): Context and Architectures&quot;">RFC3372</a>].  It has been used to support numerous proprietary and
      non-interoperable extensions due to its poorly defined scope.



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   <a href="./rfc3326">RFC 3326</a>, The Reason Header Field for SIP (S):  [<a href="./rfc3326" title="&quot;The Reason Header Field for the Session Initiation Protocol (SIP)&quot;">RFC3326</a>] defines the
      Reason header field.  It is used in requests, such as BYE, to
      indicate the reason that the request is being sent.

   <a href="./rfc3388">RFC 3388</a>, Grouping of Media Lines in the Session Description Protocol
   (S):  <a href="./rfc3388">RFC 3388</a> [<a href="./rfc3388" title="&quot;Grouping of Media Lines in the Session Description Protocol (SDP)&quot;">RFC3388</a>] 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.

   <a href="./rfc3420">RFC 3420</a>, Internet Media Type message/sipfrag (S):  [<a href="./rfc3420" title="&quot;Internet Media Type message/sipfrag&quot;">RFC3420</a>] 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.

   <a href="./rfc3608">RFC 3608</a>, SIP Extension Header Field for Service Route Discovery
   During Registration (S):  [<a href="./rfc3608" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration&quot;">RFC3608</a>] 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.

   <a href="./rfc3841">RFC 3841</a>, Caller Preferences for SIP (S):  [<a href="./rfc3841" title="&quot;Caller Preferences for the Session Initiation Protocol (SIP)&quot;">RFC3841</a>] 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 [<a href="./rfc3840" title="and P. Kyzivat">RFC3840</a>].

   <a href="./rfc4028">RFC 4028</a>, Session Timers in SIP (S):  [<a href="./rfc4028" title="&quot;Session Timers in the Session Initiation Protocol (SIP)&quot;">RFC4028</a>] 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.

   <a href="./rfc4168">RFC 4168</a>, SCTP as a Transport for SIP (S):  [<a href="./rfc4168" title="&quot;The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)&quot;">RFC4168</a>] defines how to
      carry SIP messages over the Stream Control Transmission Protocol
      (SCTP) [<a href="./rfc4960" title="&quot;Stream Control Transmission Protocol&quot;">RFC4960</a>].  SCTP has seen very limited usage for SIP
      transport.







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   <a href="./rfc4244">RFC 4244</a>, An Extension to SIP for Request History Information (S):
      [<a href="./rfc4244" title="&quot;An Extension to the Session Initiation Protocol (SIP) for Request History Information&quot;">RFC4244</a>] defines the History-Info header field, which indicates
      information on how and why a call came to be routed to a
      particular destination.

   <a href="./rfc4145">RFC 4145</a>, TCP-Based Media Transport in the Session Description
   Protocol (SDP) (S):  [<a href="./rfc4145" title="&quot;TCP-Based Media Transport in the Session Description Protocol (SDP)&quot;">RFC4145</a>] 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.

   <a href="./rfc4091">RFC 4091</a>, The Alternative Network Address Types (ANAT) Semantics for
   the Session Description Protocol (SDP) Grouping Framework (S):
      [<a href="./rfc4091" title="&quot;The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework&quot;">RFC4091</a>] 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 [<a href="#ref-ICE" title="&quot;Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;">ICE</a>].

   SDP-MEDIA, SDP Media Capabilities Negotiation (S):  [<a href="#ref-SDP-MEDIA" title="&quot;SDP media capabilities Negotiation&quot;">SDP-MEDIA</a>]
      defines an extension to the SDP capability negotiation framework
      [<a href="#ref-SDP-CAP" title="&quot;SDP Capability Negotiation&quot;">SDP-CAP</a>] for negotiating codecs, codec parameters, and media
      streams.

   BODY-HANDLING, Message Body Handling in the Session Initiation
   Protocol (SIP):  [<a href="#ref-BODY-HANDLING" title="&quot;Message Body Handling in the Session Initiation Protocol (SIP)&quot;">BODY-HANDLING</a>] clarifies handling of bodies in SIP,
      focusing primarily on multi-part behavior, which was under-
      specified in SIP.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  NAT Traversal</span>

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

   ICE, Interactive Connectivity Establishment (ICE) (S):  [<a href="#ref-ICE" title="&quot;Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;">ICE</a>] 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 [<a href="#ref-OPTION-TAG" title="&quot;Indicating Support for Interactive Connectivity Establishment (ICE) in the Session Initiation Protocol (SIP)&quot;">OPTION-TAG</a>] have been defined for use with ICE.

   ICE-TCP, TCP Candidates with Interactive Connectivity Establishment
   (ICE) (S):  [<a href="#ref-ICE-TCP" title="&quot;TCP Candidates with Interactive Connectivity Establishment (ICE)&quot;">ICE-TCP</a>] 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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   <a href="./rfc3605">RFC 3605</a>, Real Time Control Protocol (RTCP) Attribute in the Session
   Description Protocol (SDP) (S):  [<a href="./rfc3605" title="&quot;Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)&quot;">RFC3605</a>] 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) [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>].  It is needed for devices
      behind NAT, and the specification is required by ICE.

   OUTBOUND, Managing Client Initiated Connections through SIP (S):
      [<a href="#ref-OUTBOUND" title="&quot;Managing Client Initiated Connections in the Session Initiation Protocol (SIP)&quot;">OUTBOUND</a>], 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.

   <a href="./rfc3581">RFC 3581</a>, An Extension to SIP for Symmetric Response Routing (S):
      [<a href="./rfc3581" title="&quot;An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing&quot;">RFC3581</a>] 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):  [<a href="#ref-GRUU" title="&quot;Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)&quot;">GRUU</a>] 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.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Call Control Primitives</span>

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

   <a href="./rfc3515">RFC 3515</a>, The REFER Method (S):  REFER [<a href="./rfc3515" title="&quot;The Session Initiation Protocol (SIP) Refer Method&quot;">RFC3515</a>] 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.

   <a href="./rfc3725">RFC 3725</a>, Best Current Practices for Third Party Call Control (3pcc)
   (B):  [<a href="./rfc3725" title="&quot;Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)&quot;">RFC3725</a>] defines a number of different call flows that allow
      one SIP entity, called the controller, to create SIP sessions
      amongst other SIP user agents.

   <a href="./rfc3911">RFC 3911</a>, The SIP Join Header Field (S):  [<a href="./rfc3911" title="&quot;The Session Initiation Protocol (SIP) &quot;">RFC3911</a>] 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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   <a href="./rfc3891">RFC 3891</a>, The SIP Replaces Header (S):  [<a href="./rfc3891" title="&quot;The Session Initiation Protocol (SIP) &quot;">RFC3891</a>] defines a mechanism
      that allows a new dialog to replace an existing dialog.  It is
      useful for certain advanced transfer services.

   <a href="./rfc3892">RFC 3892</a>, The SIP Referred-By Mechanism (S):  [<a href="./rfc3892" title="&quot;The Session Initiation Protocol (SIP) Referred-By Mechanism&quot;">RFC3892</a>] 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.

   <a href="./rfc4117">RFC 4117</a>, Transcoding Services Invocation in SIP Using Third Party
   Call Control (I):  [<a href="./rfc4117" title="&quot;Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)&quot;">RFC4117</a>] defines how to use 3pcc for the purposes
      of invoking transcoding services for a call.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Event Framework</span>

   <a href="./rfc3265">RFC 3265</a>, SIP-Specific Event Notification (S):  [<a href="./rfc3265" title="&quot;Session Initiation Protocol (SIP)- Specific Event Notification&quot;">RFC3265</a>] 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 <a href="./rfc3265">RFC 3265</a> implementation is required when any event
      package is used.

   <a href="./rfc3903">RFC 3903</a>, SIP Extension for Event State Publication (S):  [<a href="./rfc3903" title="&quot;Session Initiation Protocol (SIP) Extension for Event State Publication&quot;">RFC3903</a>]
      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.

   <a href="./rfc4662">RFC 4662</a>, A Session Initiation Protocol (SIP) Event Notification
   Extension for Resource Lists (S):  [<a href="./rfc4662" title="&quot;A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists&quot;">RFC4662</a>] defines an extension to
      <a href="./rfc3265">RFC 3265</a> 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):  [<a href="#ref-SUBNOT-ETAGS" title="&quot;An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification&quot;">SUBNOT-ETAGS</a>]
      defines an extension to <a href="./rfc3265">RFC 3265</a> 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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<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  Event Packages</span>

   These are event packages defined to utilize the SIP events framework.
   Many of these are also listed elsewhere in their respective areas.

   <a href="./rfc3680">RFC 3680</a>, A SIP Event Package for Registrations (S):  [<a href="./rfc3680" title="&quot;A Session Initiation Protocol (SIP) Event Package for Registrations&quot;">RFC3680</a>]
      defines an event package for finding out about changes in
      registration state.

   GRUU-REG (S):  [<a href="#ref-GRUU-REG" title="&quot;Registration Event Package Extension for Session Initiation Protocol (SIP) Globally Routable User Agent URIs (GRUUs)&quot;">GRUU-REG</a>] is an extension to the registration event
      package [<a href="./rfc3680" title="&quot;A Session Initiation Protocol (SIP) Event Package for Registrations&quot;">RFC3680</a>] that allows user agents to learn about their
      GRUUs.  It is particularly useful in helping to synchronize a
      client and its registrar with their currently valid temporary
      GRUU.

   <a href="./rfc3842">RFC 3842</a>, A Message Summary and Message Waiting Indication Event
   Package for SIP (S):  [<a href="./rfc3842" title="&quot;A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)&quot;">RFC3842</a>] defines a way for a user agent to
      find out about voicemails and other messages that are waiting for
      it.  Its primary purpose is to enable the voicemail waiting lamp
      on most business telephones.

   <a href="./rfc3856">RFC 3856</a>, A Presence Event Package for SIP (S):  [<a href="./rfc3856" title="&quot;A Presence Event Package for the Session Initiation Protocol (SIP)&quot;">RFC3856</a>] defines an
      event package for indicating user presence through SIP.

   <a href="./rfc3857">RFC 3857</a>, A Watcher Information Event Template Package for SIP (S):
      [<a href="./rfc3857" title="&quot;A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP)&quot;">RFC3857</a>], also known as winfo, provides a mechanism for a user
      agent to find out what subscriptions are in place for a particular
      event package.  Its primary usage is with presence, but it can be
      used with any event package.

   <a href="./rfc4235">RFC 4235</a>, An INVITE-Initiated Dialog Event Package for SIP (S):
      [<a href="./rfc4235" title="&quot;An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP)&quot;">RFC4235</a>] defines an event package for learning the state of the
      dialogs in progress at a user agent, and is one of several RFCs
      starting with the important number 42 [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>].

   <a href="./rfc4575">RFC 4575</a>, A SIP Event Package for Conference State (S):  [<a href="./rfc4575" title="&quot;A Session Initiation Protocol (SIP) Event Package for Conference State&quot;">RFC4575</a>]
      defines a mechanism for learning about changes in conference
      state, including conference membership.

   <a href="./rfc4730">RFC 4730</a>, A SIP Event Package for Key Press Stimulus (KPML) (S):
      [<a href="./rfc4730" title="&quot;A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML)&quot;">RFC4730</a>] defines a way for an application in the network to
      subscribe to the set of key presses made on the keypad of a
      traditional telephone.  It, along with <a href="./rfc4733">RFC 4733</a> [<a href="./rfc4733" title="H. and T. Taylor">RFC4733</a>], are the
      two mechanisms defined for handling DTMF.  <a href="./rfc4730">RFC 4730</a> is a
      signaling-path solution, and <a href="./rfc4733">RFC 4733</a> is a media-path solution.






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   RTCP-SUM, SIP Event Package for Voice Quality Reporting  (S):
      [<a href="#ref-RTCP-SUM" title="&quot;Session Initiation Protocol Package for Voice Quality Reporting Event&quot;">RTCP-SUM</a>] defines a SIP event package that enables the collection
      and reporting of metrics that measure the quality for Voice over
      Internet Protocol (VoIP) sessions.

   SESSION-POLICY, A Framework for Session Initiation Protocol (SIP)
   Session Policies (S):  [<a href="#ref-SESSION-POLICY" title="&quot;A Framework for Session Initiation Protocol (SIP) Session Policies&quot;">SESSION-POLICY</a>] defines a framework for
      session policies.  In this framework, policy servers are used to
      tell user agents about the media characteristics required for a
      particular session.  The session policy framework has not been
      widely implemented.

   POLICY-PACK, A Session Initiation Protocol (SIP) Event Package for
   Session-Specific Session Policies (S):  [<a href="#ref-POLICY-PACK" title="&quot;A Session Initiation Protocol (SIP) Event Package for Session-Specific Session Policies.&quot;">POLICY-PACK</a>] defines a SIP
      event package used in conjunction with the session policy
      framework [<a href="#ref-SESSION-POLICY" title="&quot;A Framework for Session Initiation Protocol (SIP) Session Policies&quot;">SESSION-POLICY</a>].

   <a href="./rfc5362">RFC 5362</a>, The Session Initiation Protocol (SIP) Pending Additions
   Event Package (S):  [<a href="./rfc5362" title="&quot;The Session Initiation Protocol (SIP) Pending Additions Event Package&quot;">RFC5362</a>] defines a SIP event package that allows
      a UA to learn whether consent has been given for the addition of
      an address to a SIP "mailing list".  It is used in conjunction
      with the SIP framework for consent [<a href="./rfc5360" title="&quot;A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)&quot;">RFC5360</a>].

<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a>.  Quality of Service</span>

   Several specifications concern themselves with the interactions of
   SIP with network Quality of Service (QoS) mechanisms.

   <a href="./rfc3312">RFC 3312</a>, Integration of Resource Management and SIP (S):  [<a href="./rfc3312" title="&quot;Integration of Resource Management and Session Initiation Protocol (SIP)&quot;">RFC3312</a>],
      updated by [<a href="./rfc4032" title="&quot;Update to the Session Initiation Protocol (SIP) Preconditions Framework&quot;">RFC4032</a>], defines a way to make sure that the phone of
      the called party doesn't ring until a QoS reservation has been
      installed in the network.  It does so by defining a general
      preconditions framework, which defines conditions that must be
      true in order for a SIP session to proceed.

   QoS-ID, Quality of Service (QoS) Mechanism Selection in the Session
   Description Protocol (SDP) (S):  [<a href="#ref-QoS-ID" title="and G. Camarillo">QoS-ID</a>] defines a way for user
      agents to negotiate what type of end-to-end QoS mechanism to use
      for a session.  At this time, there are two that can be used: the
      Resource Reservation Protocol (RSVP) and Next Steps in Signaling
      (NSIS).  This negotiation is done through an SDP extension.  Due
      to limited deployment of RSVP and even more limited deployment of
      NSIS, this extension has not been widely used.

   <a href="./rfc3313">RFC 3313</a>, Private SIP Extensions for Media Authorization (I):
      [<a href="./rfc3313" title="&quot;Private Session Initiation Protocol (SIP) Extensions for Media Authorization&quot;">RFC3313</a>] defines a P-header that provides a mechanism for passing
      an authorization token between SIP and a network QoS reservation
      protocol like RSVP.  Its purpose is to make sure network QoS is



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      only granted if a client has made a SIP call through the same
      provider's network.  This specification is sometimes referred to
      as the SIP walled-garden specification by the truly paranoid
      androids in the SIP community.  This is because it requires
      coupling of signaling and the underlying IP network.

   <a href="./rfc3524">RFC 3524</a>, Mapping of Media Streams to Resource Reservation Flows
   (S):  [<a href="./rfc3524" title="&quot;Mapping of Media Streams to Resource Reservation Flows&quot;">RFC3524</a>] defines a usage of the SDP grouping framework for
      indicating that a set of media streams should be handled by a
      single resource reservation.

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  Operations and Management</span>

   Several specifications have been defined to support operations and
   management of SIP systems.  These include mechanisms for
   configuration and network diagnostics.

   CONFIG-FRAME, A Framework for SIP User Agent Profile Delivery (S):
      [<a href="#ref-CONFIG-FRAME" title="&quot;A Framework for Session Initiation Protocol User Agent Profile Delivery&quot;">CONFIG-FRAME</a>] defines a mechanism that allows a SIP user agent to
      bootstrap its configuration from the network and receive updates
      to its configuration, should it change.  This is considered an
      essential piece of deploying a usable SIP network.

   RTCP-SUM, SIP Event Package for Voice Quality Reporting  (S):
      [<a href="#ref-RTCP-SUM" title="&quot;Session Initiation Protocol Package for Voice Quality Reporting Event&quot;">RTCP-SUM</a>] defines a SIP event package that enables the collection
      and reporting of metrics that measure the quality for Voice over
      Internet Protocol (VoIP) sessions.

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a>.  SIP Compression</span>

   Sigcomp [<a href="./rfc3320" title="&quot;Signaling Compression (SigComp)&quot;">RFC3320</a>] [<a href="./rfc4896" title="&quot;Signaling Compression (SigComp) Corrections and Clarifications&quot;">RFC4896</a>] was defined to allow compression of SIP
   messages over low bandwidth links.  Sigcomp is not formally part of
   SIP.  However, usage of Sigcomp with SIP has required extensions to
   SIP.

   <a href="./rfc3486">RFC 3486</a>, Compressing SIP (S):  [<a href="./rfc3486" title="&quot;Compressing the Session Initiation Protocol (SIP)&quot;">RFC3486</a>] defines a SIP URI parameter
      that can be used to indicate that a SIP server supports Sigcomp.

   <a href="./rfc5049">RFC 5049</a>, Applying Signaling Compression (SigComp) to the Session
   Initiation Protocol (SIP) (S):  [<a href="./rfc5049" title="&quot;Applying Signaling Compression (SigComp) to the Session Initiation Protocol (SIP)&quot;">RFC5049</a>] defines how to apply
      Sigcomp to SIP.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  SIP Service URIs</span>

   Several extensions define well-known services that can be invoked by
   constructing requests with specific structures for the Request URI,
   resulting in specific behaviors at the User Agent Server (UAS).




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   <a href="./rfc3087">RFC 3087</a>, Control of Service Context using Request URI (I):
      [<a href="./rfc3087" title="&quot;Control of Service Context using SIP Request-URI&quot;">RFC3087</a>] introduced the context of using Request URIs, encoded
      appropriately, to invoke services.

   <a href="./rfc4662">RFC 4662</a>, A SIP Event Notification Extension for Resource Lists (S):
      [<a href="./rfc4662" title="&quot;A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists&quot;">RFC4662</a>] defines a resource called a Resource List Server (RLS).
      A client can send a subscribe to this server.  The server will
      generate a series of subscriptions, compile the resulting
      information, and send it back to the subscriber.  The set of
      resources that the RLS will subscribe to is a property of the
      request URI in the SUBSCRIBE request.

   <a href="./rfc5363">RFC 5363</a>, Framework and Security Considerations for Session
   Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List
   Services (S):  [<a href="./rfc5363" title="&quot;Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services&quot;">RFC5363</a>] defines the framework for list services in
      SIP.  In this framework, a UA can include an XML list object in
      the body of various requests and the server will provide list-
      oriented services as a consequence.  For example, a SUBSCRIBE with
      a list subscribes to the URI in the list.

   <a href="./rfc5367">RFC 5367</a>, Subscriptions To Request-Contained Resource Lists in SIP
   (S):  [<a href="./rfc5367" title="&quot;Subscriptions to Request-Contained Resource Lists in the Session Initiation Protocol (SIP)&quot;">RFC5367</a>] uses the URI-list framework [<a href="./rfc5363" title="&quot;Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services&quot;">RFC5363</a>] and allows a
      client to subscribe to a resource called a Resource List Server.
      This server will generate subscriptions to the URI in the list,
      compile the resulting information, and send it back to the
      subscriber.

   <a href="./rfc5365">RFC 5365</a>, Multiple-Recipient MESSAGE Requests in SIP (S):  [<a href="./rfc5365" title="&quot;Multiple- Recipient MESSAGE Requests in the Session Initiation Protocol (SIP)&quot;">RFC5365</a>]
      uses the URI-list framework [<a href="./rfc5363" title="&quot;Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services&quot;">RFC5363</a>] and allows a client to send
      a MESSAGE to a number of recipients.

   <a href="./rfc5366">RFC 5366</a>, Conference Establishment Using Request-Contained Lists in
   SIP (S):  [<a href="./rfc5366" title="&quot;Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)&quot;">RFC5366</a>] uses the URI-list framework [<a href="./rfc5363" title="&quot;Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services&quot;">RFC5363</a>].  It allows
      a client to ask the server to act as a conference focus and send
      an invitation to each recipient in the list.

   <a href="./rfc4240">RFC 4240</a>, Basic Network Media Services with SIP (I):  [<a href="./rfc4240" title="and A. Spitzer">RFC4240</a>]
      defines a way for SIP application servers to invoke announcement
      and conferencing services from a media server.  This is
      accomplished through a set of defined URI parameters that tell the
      media server what to do, such as what file to play and what
      language to render it in.

   <a href="./rfc4458">RFC 4458</a>, Session Initiation Protocol (SIP) URIs for Applications
   such as Voicemail and Interactive Voice Response (IVR) (I):
      [<a href="./rfc4458" title="&quot;Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and Interactive Voice Response (IVR)&quot;">RFC4458</a>] defines a way to invoke voicemail and IVR services by
      using a SIP URI constructed in a particular way.




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<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a>.  Minor Extensions</span>

   These SIP extensions don't fit easily into a single specific use
   case.  They have somewhat general applicability, but they solve a
   relatively small problem or provide an optimization.

   <a href="./rfc4488">RFC 4488</a>, Suppression of the SIP REFER Implicit Subscription (S):
      [<a href="./rfc4488" title="&quot;Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription&quot;">RFC4488</a>] defines an enhancement to REFER.  REFER normally creates
      an implicit subscription to the target of the REFER.  This
      subscription is used to pass back updates on the progress of the
      referral.  This extension allows that implicit subscription to be
      bypassed as an optimization.

   <a href="./rfc4538">RFC 4538</a>, Request Authorization through Dialog Identification in SIP
   (S):  [<a href="./rfc4538" title="&quot;Request Authorization through Dialog Identification in the Session Initiation Protocol (SIP)&quot;">RFC4538</a>] provides a mechanism that allows a UAS to authorize a
      request because the requestor proves it knows a dialog that is in
      progress with the UAS.  The specification is useful in conjunction
      with the SIP application interaction framework [<a href="#ref-INTERACT-FRAME" title="&quot;A Framework for Application Interaction in the Session Initiation Protocol (SIP)&quot;">INTERACT-FRAME</a>].

   <a href="./rfc4508">RFC 4508</a>, Conveying Feature Tags with the REFER Method in SIP (S):
      [<a href="./rfc4508" title="&quot;Conveying Feature Tags with the Session Initiation Protocol (SIP) REFER Method&quot;">RFC4508</a>] defines a mechanism for carrying <a href="./rfc3840">RFC 3840</a> feature tags
      in REFER.  It is useful for informing the target of the REFER
      about the characteristics of the intended target of the referred
      request.

   <a href="./rfc5373">RFC 5373</a>, Requesting Answer Modes for SIP (S):  [<a href="./rfc5373" title="&quot;Requesting Answering Modes for the Session Initiation Protocol (SIP)&quot;">RFC5373</a>] defines an
      extension for indicating to the called party whether or not the
      phone should ring and/or be answered immediately.  This is useful
      for push-to-talk and for diagnostic applications.

   <a href="./rfc5079">RFC 5079</a>, Rejecting Anonymous Requests in SIP (S):  [<a href="./rfc5079" title="&quot;Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)&quot;">RFC5079</a>] defines
      a mechanism for a called party to indicate to the calling party
      that a call was rejected since the caller was anonymous.  This is
      needed for implementation of the Anonymous Call Rejection (ACR)
      feature in SIP.

   <a href="./rfc5368">RFC 5368</a>, Referring to Multiple Resources in SIP (S):  [<a href="./rfc5368" title="&quot;Referring to Multiple Resources in the Session Initiation Protocol (SIP)&quot;">RFC5368</a>]
      allows a UA sending a REFER to ask the recipient of the REFER to
      generate multiple SIP requests, not just one.  This is useful for
      conferencing, where a client would like to ask a conference server
      to eject multiple users.

   <a href="./rfc4483">RFC 4483</a>, A Mechanism for Content Indirection in Session Initiation
   Protocol (SIP) Messages (S):  [<a href="./rfc4483" title="&quot;A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages&quot;">RFC4483</a>] defines a mechanism for
      content indirection.  Instead of carrying an object within a SIP
      body, a URL reference is carried instead, and the recipient
      dereferences the URL to obtain the object.  The specification has
      potential applicability for sending large instant messages, but



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      has yet to find much actual use.

   <a href="./rfc3890">RFC 3890</a>, A Transport Independent Bandwidth Modifier for the Session
   Description Protocol (SDP) (S):  [<a href="./rfc3890" title="&quot;A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)&quot;">RFC3890</a>] specifies an SDP extension
      that allows for the description of the bandwidth for a media
      session that is independent of the underlying transport mechanism.

   <a href="./rfc4583">RFC 4583</a>, Session Description Protocol (SDP) Format for Binary Floor
   Control Protocol (BFCP) Streams (S):  [<a href="./rfc4583" title="&quot;Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams&quot;">RFC4583</a>] defines a mechanism
      in SDP to signal floor control streams that use BFCP.  It is used
      for push-to-talk and conference floor control.

   CONNECT-PRECON, Connectivity Preconditions for Session Description
   Protocol Media Streams (S):  [<a href="#ref-CONNECT-PRECON" title="&quot;Connectivity Preconditions for Session Description Protocol Media Streams&quot;">CONNECT-PRECON</a>] defines a usage of the
      precondition framework [<a href="./rfc3312" title="&quot;Integration of Resource Management and Session Initiation Protocol (SIP)&quot;">RFC3312</a>].  The connectivity precondition
      makes sure that the session doesn't get established until actual
      packet connectivity is checked.

   <a href="./rfc4796">RFC 4796</a>, The SDP (Session Description Protocol) Content Attribute
   (S):  [<a href="./rfc4796" title="&quot;The Session Description Protocol (SDP) Content Attribute&quot;">RFC4796</a>] defines an SDP attribute for describing the purpose
      of a media stream.  Examples include a slide view, the speaker, a
      sign language feed, and so on.

   IPv6-TRANS, IPv6 Transition in the Session Initiation Protocol (SIP)
   (S):  [<a href="#ref-IPv6-TRANS" title="&quot;IPv6 Transition in the Session Initiation Protocol (SIP)&quot;">IPv6-TRANS</a>] defines practices for interworking between IPv6
      and IPv6 user agents.  This is done through multi-homed proxies
      that interwork IPv4 and IPv6, along with ICE [<a href="#ref-ICE" title="&quot;Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;">ICE</a>] for media
      traversal.  The specification includes some minor extensions and
      clarifications to SDP in order to cover some additional cases.

   CONNECT-REUSE, Connection Reuse in the Session Initiation Protocol
   (SIP) (S):  [<a href="#ref-CONNECT-REUSE" title="&quot;Connection Reuse in the Session Initiation Protocol (SIP)&quot;">CONNECT-REUSE</a>] defines an extension to SIP that allows a
      Transport Layer Security (TLS) connection between servers to be
      reused for requests in both directions.  Normally, two connections
      are set up between a pair of servers, one for requests in each
      direction.

<span class="h2"><a class="selflink" id="section-15" href="#section-15">15</a>.  Security Mechanisms</span>

   Several extensions provide additional security features to SIP.

   <a href="./rfc4474">RFC 4474</a>, Enhancements for Authenticated Identity Management in SIP
   (S):  [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] 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 <a href="./rfc3325">RFC</a>
      <a href="./rfc3325">3325</a>.  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.



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   <a href="./rfc4916">RFC 4916</a>, Connected Identity in the Session Initiation Protocol (SIP)
   (S):  [<a href="./rfc4916" title="&quot;Connected Identity in the Session Initiation Protocol (SIP)&quot;">RFC4916</a>] formally updates <a href="./rfc3261">RFC 3261</a>.  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 [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] 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.

   SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation
   Protocol (SIP) (S):  [<a href="#ref-SIPS-URI" title="&quot;The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)&quot;">SIPS-URI</a>] is intended to update <a href="./rfc3261">RFC 3261</a>.  It
      revises the processing of the SIPS URI, originally defined in <a href="./rfc3261">RFC</a>
      <a href="./rfc3261">3261</a>, to fix many errors and problems that have been encountered
      with that mechanism.

   DOMAIN-CERTS, Domain Certificates in the Session Initiation Protocol
   (SIP) (B):  [<a href="#ref-DOMAIN-CERTS" title="&quot;Domain Certificates in the Session Initiation Protocol (SIP)&quot;">DOMAIN-CERTS</a>] clarifies the usage of SIP over TLS with
      regards to certificate handling, and defines additional procedures
      needed for interoperability.

   <a href="./rfc3323">RFC 3323</a>, A Privacy Mechanism for the Session Initiation Protocol
   (SIP) (S):  [<a href="./rfc3323" title="&quot;A Privacy Mechanism for the Session Initiation Protocol (SIP)&quot;">RFC3323</a>] 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
      [<a href="./rfc3325" title="&quot;Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks&quot;">RFC3325</a>].

   <a href="./rfc4567">RFC 4567</a>, Key Management Extensions for Session Description Protocol
   (SDP) and Real Time Streaming Protocol (RTSP) (S):  [<a href="./rfc4567" title="&quot;Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)&quot;">RFC4567</a>] defines
      extensions to SDP that allow tunneling of a key management
      protocol, namely MIKEY [<a href="./rfc3830" title="&quot;MIKEY: Multimedia Internet KEYing&quot;">RFC3830</a>], through offer/answer exchanges.
      This mechanism is one of three Secure Realtime Transport Protocol
      (SRTP) keying techniques specified for SIP, with Datagram
      Transport Layer Security (DTLS)-SRTP [<a href="#ref-SRTP-FRAME" title="&quot;Framework for Establishing an SRTP Security Context using DTLS&quot;">SRTP-FRAME</a>] having been
      selected as the final solution.

   <a href="./rfc4568">RFC 4568</a>, Session Description Protocol (SDP) Security Descriptions
   for Media Streams (S):  [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>] defines extensions to SDP that
      allow for the negotiation of keying material directly through
      offer/answer, without a separate key management protocol.  This
      mechanism, sometimes called sdescriptions, has the drawback that
      the media keys are available to any entity that has visibility to
      the SDP.  It is one of three SRTP keying techniques specified for
      SIP, with DTLS-SRTP [<a href="#ref-SRTP-FRAME" title="&quot;Framework for Establishing an SRTP Security Context using DTLS&quot;">SRTP-FRAME</a>] having been selected as the final
      solution.




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   SRTP-FRAME, Framework for Establishing an SRTP Security Context using
   DTLS (S):  [<a href="#ref-SRTP-FRAME" title="&quot;Framework for Establishing an SRTP Security Context using DTLS&quot;">SRTP-FRAME</a>] defines the overall framework and SDP and SIP
      processing required to perform key management for RTP using
      Datagram TLS (DTLS) [<a href="./rfc4347" title="&quot;Datagram Transport Layer Security&quot;">RFC4347</a>] directly between endpoints, over the
      media path.  It is one of three SRTP keying techniques specified
      for SIP, with DTLS-SRTP [<a href="#ref-SRTP-FRAME" title="&quot;Framework for Establishing an SRTP Security Context using DTLS&quot;">SRTP-FRAME</a>] having been selected as the
      final solution.

   <a href="./rfc3853">RFC 3853</a>, S/MIME Advanced Encryption Standard (AES) Requirement for
   SIP (S):  [<a href="./rfc3853" title="&quot;S/MIME Advanced Encryption Standard (AES) Requirement for the Session Initiation Protocol (SIP)&quot;">RFC3853</a>] formally updates <a href="./rfc3261">RFC 3261</a>.  It is a brief
      specification that updates the cryptography mechanisms used in SIP
      S/MIME.  However, SIP S/MIME has seen very little deployment.

   CERTS, Certificate Management Service for the Session Initiation
   Protocol (SIP) (S):  [<a href="#ref-CERTS" title="&quot;Certificate Management Service for The Session Initiation Protocol (SIP)&quot;">CERTS</a>] defines a certificate service for SIP
      whose purpose is to facilitate the deployment of S/MIME.  The
      certificate service allows clients to store and retrieve their own
      certificates, in addition to obtaining the certificates for other
      users.

   <a href="./rfc3893">RFC 3893</a>, Session Initiation Protocol (SIP) Authenticated Identity
   Body (AIB) Format (S):  [<a href="./rfc3893" title="&quot;Session Initiation Protocol (SIP) Authenticated Identity Body (AIB) Format&quot;">RFC3893</a>] defines a SIP message fragment that
      can be signed in order to provide an authenticated identity over a
      request.  It was an early predecessor to [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>], and
      consequently AIB has seen no deployment.

   SAML, SIP SAML Profile and Binding (S):  [<a href="#ref-SAML" title="&quot;SIP SAML Profile and Binding&quot;">SAML</a>] defines the usage of
      the Security Assertion Markup Language (SAML) within SIP, and
      describes how to use it in conjunction with SIP identity [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>]
      to provide authenticated assertions about a user's role or
      attributes.

   <a href="./rfc5360">RFC 5360</a>, A Framework for Consent-Based Communications in the Session
   Initiation Protocol (SIP) (S):  [<a href="./rfc5360" title="&quot;A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)&quot;">RFC5360</a>] defines several extensions
      to SIP, including the Trigger-Consent and Permission-Missing
      header fields.  These header fields, in addition to the other
      procedures defined in the document, define a way to manage
      membership on "SIP mailing lists" used for instant messaging or
      conferencing.  In particular, it helps avoid the problem of using
      such amplification services for the purposes of an attack on the
      network by making sure a user authorizes the addition of their
      address onto such a service.

   <a href="./rfc5361">RFC 5361</a>, A Document Format for Requesting Consent (S):  [<a href="./rfc5361" title="&quot;A Document Format for Requesting Consent&quot;">RFC5361</a>]
      defines an XML object used by the consent framework.  Consent
      documents are sent from SIP "mailing list servers" to users to
      allow them to manage their membership on lists.




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   <a href="./rfc5362">RFC 5362</a>, The Session Initiation Protocol (SIP) Pending Additions
   Event Package (S):  [<a href="./rfc5362" title="&quot;The Session Initiation Protocol (SIP) Pending Additions Event Package&quot;">RFC5362</a>] defines a SIP event package that allows
      a UA to learn whether consent has been given for the addition of
      an address to a SIP "mailing list".  It is used in conjunction
      with the SIP framework for consent [<a href="./rfc5360" title="&quot;A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)&quot;">RFC5360</a>].

   <a href="./rfc3329">RFC 3329</a>, Security Mechanism Agreement for SIP (S):  [<a href="./rfc3329" title="&quot;Security Mechanism Agreement for the Session Initiation Protocol (SIP)&quot;">RFC3329</a>]
      defines a mechanism to prevent bid-down attacks in conjunction
      with SIP authentication.  The mechanism has seen very limited
      deployment.  It was defined as part of the 3GPP IP Multimedia
      Subsystem (IMS) specification suite [<a href="#ref-3GPP.24.229" title="&quot;Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3&quot;">3GPP.24.229</a>], and is needed
      only when there is a multiplicity of security mechanisms deployed
      at a particular server.  In practice, this has not been the case.

   <a href="./rfc4572">RFC 4572</a>, Connection-Oriented Media Transport over the Transport
   Layer Security (TLS) Protocol in the Session Description Protocol
   (SDP) (S):  [<a href="./rfc4572" title="&quot;Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)&quot;">RFC4572</a>] specifies a mechanism for signaling TLS-based
      media streams between endpoints.  It expands the TCP-based media
      signaling parameters defined in [<a href="./rfc4145" title="&quot;TCP-Based Media Transport in the Session Description Protocol (SDP)&quot;">RFC4145</a>] to include fingerprint
      information for TLS streams so that TLS can operate between end
      hosts using self-signed certificates.

   <a href="./rfc5027">RFC 5027</a>, Security Preconditions for Session Description Protocol
   Media Streams (S):  [<a href="./rfc5027" title="&quot;Security Preconditions for Session Description Protocol (SDP) Media Streams&quot;">RFC5027</a>] defines a precondition for use with the
      preconditions framework [<a href="./rfc3312" title="&quot;Integration of Resource Management and Session Initiation Protocol (SIP)&quot;">RFC3312</a>].  The security precondition
      prevents a session from being established until a security media
      stream is set up.

   <a href="./rfc3310">RFC 3310</a>, Hypertext Transfer Protocol (HTTP) Digest Authentication
   Using Authentication and Key Agreement (S):  [<a href="./rfc3310" title="&quot;Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA)&quot;">RFC3310</a>] defines an
      extension to digest authentication to allow it to work with the
      credentials stored in cell phones.  Though technically it is an
      extension to HTTP digest, its primary application is SIP.  This
      extension is useful primarily to implementors of IMS.

   <a href="./rfc4169">RFC 4169</a>, Hypertext Transfer Protocol (HTTP) Digest Authentication
   Using Authentication and Key Agreement (AKA) Version-2 (S):
      [<a href="./rfc4169" title="&quot;Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA) Version-2&quot;">RFC4169</a>] is an enhancement to [<a href="./rfc3310" title="&quot;Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA)&quot;">RFC3310</a>] that further improves
      security of the authentication.

<span class="h2"><a class="selflink" id="section-16" href="#section-16">16</a>.  Conferencing</span>

   Numerous SIP and SDP extensions are aimed at conferencing as their
   primary application.







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   <a href="./rfc4574">RFC 4574</a>, The SDP (Session Description Protocol) Label Attribute
   (S):  [<a href="./rfc4574" title="&quot;The Session Description Protocol (SDP) Label Attribute&quot;">RFC4574</a>] defines an SDP attribute for providing an opaque
      label for media streams.  These labels can be referred to by
      external documents, and in particular, by conference policy
      documents.  This allows a UA to tie together documents it may
      obtain through conferencing mechanisms to media streams to which
      they refer.

   <a href="./rfc3911">RFC 3911</a>, The SIP Join Header Field (S):  [<a href="./rfc3911" title="&quot;The Session Initiation Protocol (SIP) &quot;">RFC3911</a>] 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.

   <a href="./rfc4575">RFC 4575</a>, A SIP Event Package for Conference State (S):  [<a href="./rfc4575" title="&quot;A Session Initiation Protocol (SIP) Event Package for Conference State&quot;">RFC4575</a>]
      defines a mechanism for learning about changes in conference
      state, including conference membership.

   <a href="./rfc5368">RFC 5368</a>, Referring to Multiple Resources in SIP (S):  [<a href="./rfc5368" title="&quot;Referring to Multiple Resources in the Session Initiation Protocol (SIP)&quot;">RFC5368</a>]
      allows a UA sending a REFER to ask the recipient of the REFER to
      generate multiple SIP requests, not just one.  This is useful for
      conferencing, where a client would like to ask a conference server
      to eject multiple users.

   <a href="./rfc5366">RFC 5366</a>, Conference Establishment Using Request-Contained Lists in
   SIP (S):  [<a href="./rfc5366" title="&quot;Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)&quot;">RFC5366</a>] is similar to [<a href="./rfc5367" title="&quot;Subscriptions to Request-Contained Resource Lists in the Session Initiation Protocol (SIP)&quot;">RFC5367</a>].  However, instead of
      subscribing to the resource, an INVITE request is sent to the
      resource, and it will act as a conference focus and generate an
      invitation to each recipient in the list.

   <a href="./rfc4579">RFC4579</a>, Session Initiation Protocol (SIP) Call Control -
   Conferencing for User Agents (B):  [<a href="./rfc4579" title="&quot;Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents&quot;">RFC4579</a>] defines best practice
      procedures and call flows for conferencing.  This includes
      conference creation, joining, and dial out, amongst other
      capabilities.

   <a href="./rfc4583">RFC 4583</a>, Session Description Protocol (SDP) Format for Binary Floor
   Control Protocol (BFCP) Streams (S):  [<a href="./rfc4583" title="&quot;Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams&quot;">RFC4583</a>] defines a mechanism
      in SDP to signal floor control streams that use BFCP.  It is used
      for push-to-talk and conference floor control.

<span class="h2"><a class="selflink" id="section-17" href="#section-17">17</a>.  Instant Messaging, Presence, and Multimedia</span>

   SIP provides extensions for instant messaging, presence, and
   multimedia.







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   <a href="./rfc3428">RFC 3428</a>, SIP Extension for Instant Messaging (S):  [<a href="./rfc3428" title="&quot;Session Initiation Protocol (SIP) Extension for Instant Messaging&quot;">RFC3428</a>] defines
      the MESSAGE method, used for sending an instant message without
      setting up a session (sometimes called "page mode").

   <a href="./rfc3856">RFC 3856</a>, A Presence Event Package for SIP (S):  [<a href="./rfc3856" title="&quot;A Presence Event Package for the Session Initiation Protocol (SIP)&quot;">RFC3856</a>] defines an
      event package for indicating user presence through SIP.

   <a href="./rfc3857">RFC 3857</a>, A Watcher Information Event Template Package for SIP (S):
      [<a href="./rfc3857" title="&quot;A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP)&quot;">RFC3857</a>], also known as winfo, provides a mechanism for a user
      agent to find out what subscriptions are in place for a particular
      event package.  Its primary usage is with presence, but it can be
      used with any event package.

   TRANSFER-MECH, A Session Description Protocol (SDP)  Offer/Answer
   Mechanism to Enable File Transfer (S):  [<a href="#ref-TRANSFER-MECH" title="&quot;A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer&quot;">TRANSFER-MECH</a>] defines a
      mechanism for signaling a file transfer session with SIP.

<span class="h2"><a class="selflink" id="section-18" href="#section-18">18</a>.  Emergency Services</span>

   Emergency services include preemption features, which allow
   authorized individuals to gain access to network resources in time of
   emergency, along with traditional emergency calling.

   <a href="./rfc4411">RFC 4411</a>, Extending the SIP Reason Header for Preemption Events (S):
      [<a href="./rfc4411" title="&quot;Extending the Session Initiation Protocol (SIP) Reason Header for Preemption Events&quot;">RFC4411</a>] defines an extension to the Reason header, allowing a UA
      to know that its dialog was torn down because a higher priority
      session came through.

   <a href="./rfc4412">RFC 4412</a>, Communications Resource Priority for SIP (S):  [<a href="./rfc4412" title="&quot;Communications Resource Priority for the Session Initiation Protocol (SIP)&quot;">RFC4412</a>]
      defines a new header field, Resource-Priority, that allows a
      session to get priority treatment from the network.

   LOCATION, Location Conveyance for the Session Initiation Protocol
   (S):  [<a href="#ref-LOCATION" title="&quot;Location Conveyance for the Session Initiation Protocol&quot;">LOCATION</a>] defines a mechanism for carrying location objects in
      SIP messages.  This is used to convey location from a UA to an
      emergency call taker.

<span class="h2"><a class="selflink" id="section-19" href="#section-19">19</a>.  Security Considerations</span>

   This specification is an overview of existing specifications and does
   not introduce any security considerations on its own.  Of course, the
   world would be far more secure if everyone would follow one simple
   rule: "Don't Panic!"  [<a href="#ref-HGTTG" title="&quot;The Hitchhiker's Guide to the Galaxy&quot;">HGTTG</a>].

<span class="h2"><a class="selflink" id="section-20" href="#section-20">20</a>.  Acknowledgements</span>

   The author would like to thank Spencer Dawkins, Brian Stucker, Keith
   Drage, John Elwell, and Avshalom Houri for their comments on this



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   document.

<span class="h2"><a class="selflink" id="section-21" href="#section-21">21</a>.  Informative References</span>

   [<a id="ref-3GPP.24.229">3GPP.24.229</a>]     3GPP, "Internet Protocol (IP) multimedia call
                     control protocol based on Session Initiation
                     Protocol (SIP) and Session Description Protocol
                     (SDP); Stage 3", 3GPP TS 24.229 5.22.0,
                     September 2008.

   [<a id="ref-ABNF-FIX">ABNF-FIX</a>]        Gurbani, V. and B. Carpenter, "Essential correction
                     for IPv6 ABNF in <a href="./rfc3261">RFC3261</a>", Work in Progress,
                     November 2007.

   [<a id="ref-BODY-HANDLING">BODY-HANDLING</a>]   Camarillo, G., "Message Body Handling in the
                     Session Initiation Protocol (SIP)", Work
                     in Progress, November 2008.

   [<a id="ref-CERTS">CERTS</a>]           Jennings, C. and J. Fischl, "Certificate Management
                     Service for The Session Initiation Protocol (SIP)",
                     Work in Progress, November 2008.

   [<a id="ref-CONFIG-FRAME">CONFIG-FRAME</a>]    Channabasappa, S., "A Framework for Session
                     Initiation Protocol User Agent Profile Delivery",
                     Work in Progress, February 2008.

   [<a id="ref-CONNECT-PRECON">CONNECT-PRECON</a>]  Andreasen, F., Camarillo, G., Oran, D., and D.
                     Wing, "Connectivity Preconditions for Session
                     Description Protocol Media Streams", Work
                     in Progress, October 2008.

   [<a id="ref-CONNECT-REUSE">CONNECT-REUSE</a>]   Gurbani, V., Mahy, R., and B. Tate, "Connection
                     Reuse in the Session Initiation Protocol (SIP)",
                     Work in Progress, October 2008.

   [<a id="ref-DOMAIN-CERTS">DOMAIN-CERTS</a>]    Gurbani, V., Lawrence, S., and B. Laboratories,
                     "Domain Certificates in the Session Initiation
                     Protocol (SIP)", Work in Progress, October 2008.

   [<a id="ref-ECRIT-FRAME">ECRIT-FRAME</a>]     Rosen, B., Schulzrinne, H., Polk, J., and A.
                     Newton, "Framework for Emergency Calling using
                     Internet Multimedia", Work in Progress, July 2008.

   [<a id="ref-GRUU">GRUU</a>]            Rosenberg, J., "Obtaining and Using Globally
                     Routable User Agent (UA) URIs (GRUU) in the Session
                     Initiation Protocol (SIP)", Work in Progress,
                     October 2007.




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   [<a id="ref-GRUU-REG">GRUU-REG</a>]        Kyzivat, P., "Registration Event Package Extension
                     for Session Initiation Protocol (SIP)  Globally
                     Routable User Agent URIs (GRUUs)", Work
                     in Progress, July 2007.

   [<a id="ref-HGTTG">HGTTG</a>]           Adams, D., "The Hitchhiker's Guide to the Galaxy",
                     September 1979.

   [<a id="ref-ICE">ICE</a>]             Rosenberg, J., "Interactive Connectivity
                     Establishment (ICE): A Protocol for Network Address
                     Translator (NAT) Traversal for Offer/Answer
                     Protocols", Work in Progress, October 2007.

   [<a id="ref-ICE-TCP">ICE-TCP</a>]         Rosenberg, J., "TCP Candidates with Interactive
                     Connectivity Establishment (ICE)", Work
                     in Progress, July 2008.

   [<a id="ref-INTERACT-FRAME">INTERACT-FRAME</a>]  Rosenberg, J., "A Framework for Application
                     Interaction in the Session Initiation Protocol
                     (SIP)", Work in Progress, July 2005.

   [<a id="ref-IPv6-TRANS">IPv6-TRANS</a>]      Camarillo, G., "IPv6 Transition in the Session
                     Initiation Protocol (SIP)", Work in Progress,
                     August 2007.

   [<a id="ref-LOCATION">LOCATION</a>]        Polk, J. and B. Rosen, "Location Conveyance for the
                     Session Initiation Protocol", Work in Progress,
                     November 2008.

   [<a id="ref-LOOP-FIX">LOOP-FIX</a>]        Sparks, R., Lawrence, S., Hawrylyshen, A., and B.
                     Campen, "Addressing an Amplification Vulnerability
                     in Session Initiation Protocol  (SIP) Forking
                     Proxies", Work in Progress, October 2008.

   [<a id="ref-OPTION-TAG">OPTION-TAG</a>]      Rosenberg, J., "Indicating Support for Interactive
                     Connectivity Establishment (ICE) in the Session
                     Initiation Protocol (SIP)", Work in Progress,
                     June 2007.

   [<a id="ref-OUTBOUND">OUTBOUND</a>]        Jennings, C. and R. Mahy, "Managing Client
                     Initiated Connections in the Session Initiation
                     Protocol  (SIP)", Work in Progress, October 2008.

   [<a id="ref-POLICY-PACK">POLICY-PACK</a>]     Hilt, V. and G. Camarillo, "A Session Initiation
                     Protocol (SIP) Event Package for Session-Specific
                     Session Policies.", Work in Progress, July 2008.

   [<a id="ref-QoS-ID">QoS-ID</a>]          Polk, J., Dhesikan, S., and G. Camarillo, "Quality



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                     of Service (QoS) Mechanism Selection in the Session
                     Description Protocol (SDP)", Work in Progress,
                     November 2008.

   [<a id="ref-RECORD-ROUTE">RECORD-ROUTE</a>]    Froment, T., Lebel, C., and B. Bonnaerens,
                     "Addressing Record-Route issues in the Session
                     Initiation Protocol (SIP)", Work in Progress,
                     October 2008.

   [<a id="ref-RFC2026">RFC2026</a>]         Bradner, S., "The Internet Standards Process --
                     Revision 3", <a href="https://www.rfc-editor.org/bcp/bcp9">BCP 9</a>, <a href="./rfc2026">RFC 2026</a>, October 1996.

   [<a id="ref-RFC2543">RFC2543</a>]         Handley, M., Schulzrinne, H., Schooler, E., and J.
                     Rosenberg, "SIP: Session Initiation Protocol",
                     <a href="./rfc2543">RFC 2543</a>, March 1999.

   [<a id="ref-RFC2782">RFC2782</a>]         Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS
                     RR for specifying the location of services (DNS
                     SRV)", <a href="./rfc2782">RFC 2782</a>, February 2000.

   [<a id="ref-RFC2848">RFC2848</a>]         Petrack, S. and L. Conroy, "The PINT Service
                     Protocol: Extensions to SIP and SDP for IP Access
                     to Telephone Call Services", <a href="./rfc2848">RFC 2848</a>, June 2000.

   [<a id="ref-RFC2976">RFC2976</a>]         Donovan, S., "The SIP INFO Method", <a href="./rfc2976">RFC 2976</a>,
                     October 2000.

   [<a id="ref-RFC3087">RFC3087</a>]         Campbell, B. and R. Sparks, "Control of Service
                     Context using SIP Request-URI", <a href="./rfc3087">RFC 3087</a>,
                     April 2001.

   [<a id="ref-RFC3204">RFC3204</a>]         Zimmerer, E., Peterson, J., Vemuri, A., Ong, L.,
                     Audet, F., Watson, M., and M. Zonoun, "MIME media
                     types for ISUP and QSIG Objects", <a href="./rfc3204">RFC 3204</a>,
                     December 2001.

   [<a id="ref-RFC3261">RFC3261</a>]         Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                     Johnston, A., Peterson, J., Sparks, R., Handley,
                     M., and E. Schooler, "SIP: Session Initiation
                     Protocol", <a href="./rfc3261">RFC 3261</a>, June 2002.

   [<a id="ref-RFC3262">RFC3262</a>]         Rosenberg, J. and H. Schulzrinne, "Reliability of
                     Provisional Responses in Session Initiation
                     Protocol (SIP)", <a href="./rfc3262">RFC 3262</a>, June 2002.

   [<a id="ref-RFC3263">RFC3263</a>]         Rosenberg, J. and H. Schulzrinne, "Session
                     Initiation Protocol (SIP): Locating SIP Servers",
                     <a href="./rfc3263">RFC 3263</a>, June 2002.



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   [<a id="ref-RFC3264">RFC3264</a>]         Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
                     Model with Session Description Protocol (SDP)",
                     <a href="./rfc3264">RFC 3264</a>, June 2002.

   [<a id="ref-RFC3265">RFC3265</a>]         Roach, A., "Session Initiation Protocol (SIP)-
                     Specific Event Notification", <a href="./rfc3265">RFC 3265</a>, June 2002.

   [<a id="ref-RFC3310">RFC3310</a>]         Niemi, A., Arkko, J., and V. Torvinen, "Hypertext
                     Transfer Protocol (HTTP) Digest Authentication
                     Using Authentication and Key Agreement (AKA)",
                     <a href="./rfc3310">RFC 3310</a>, September 2002.

   [<a id="ref-RFC3311">RFC3311</a>]         Rosenberg, J., "The Session Initiation Protocol
                     (SIP) UPDATE Method", <a href="./rfc3311">RFC 3311</a>, October 2002.

   [<a id="ref-RFC3312">RFC3312</a>]         Camarillo, G., Marshall, W., and J. Rosenberg,
                     "Integration of Resource Management and Session
                     Initiation Protocol (SIP)", <a href="./rfc3312">RFC 3312</a>, October 2002.

   [<a id="ref-RFC3313">RFC3313</a>]         Marshall, W., "Private Session Initiation Protocol
                     (SIP) Extensions for Media Authorization",
                     <a href="./rfc3313">RFC 3313</a>, January 2003.

   [<a id="ref-RFC3320">RFC3320</a>]         Price, R., Bormann, C., Christoffersson, J., Hannu,
                     H., Liu, Z., and J. Rosenberg, "Signaling
                     Compression (SigComp)", <a href="./rfc3320">RFC 3320</a>, January 2003.

   [<a id="ref-RFC3323">RFC3323</a>]         Peterson, J., "A Privacy Mechanism for the Session
                     Initiation Protocol (SIP)", <a href="./rfc3323">RFC 3323</a>,
                     November 2002.

   [<a id="ref-RFC3325">RFC3325</a>]         Jennings, C., Peterson, J., and M. Watson, "Private
                     Extensions to the Session Initiation Protocol (SIP)
                     for Asserted Identity within Trusted Networks",
                     <a href="./rfc3325">RFC 3325</a>, November 2002.

   [<a id="ref-RFC3326">RFC3326</a>]         Schulzrinne, H., Oran, D., and G. Camarillo, "The
                     Reason Header Field for the Session Initiation
                     Protocol (SIP)", <a href="./rfc3326">RFC 3326</a>, December 2002.

   [<a id="ref-RFC3327">RFC3327</a>]         Willis, D. and B. Hoeneisen, "Session Initiation
                     Protocol (SIP) Extension Header Field for
                     Registering Non-Adjacent Contacts", <a href="./rfc3327">RFC 3327</a>,
                     December 2002.

   [<a id="ref-RFC3329">RFC3329</a>]         Arkko, J., Torvinen, V., Camarillo, G., Niemi, A.,
                     and T. Haukka, "Security Mechanism Agreement for
                     the Session Initiation Protocol (SIP)", <a href="./rfc3329">RFC 3329</a>,



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                     January 2003.

   [<a id="ref-RFC3372">RFC3372</a>]         Vemuri, A. and J. Peterson, "Session Initiation
                     Protocol for Telephones (SIP-T): Context and
                     Architectures", <a href="https://www.rfc-editor.org/bcp/bcp63">BCP 63</a>, <a href="./rfc3372">RFC 3372</a>, September 2002.

   [<a id="ref-RFC3388">RFC3388</a>]         Camarillo, G., Eriksson, G., Holler, J., and H.
                     Schulzrinne, "Grouping of Media Lines in the
                     Session Description Protocol (SDP)", <a href="./rfc3388">RFC 3388</a>,
                     December 2002.

   [<a id="ref-RFC3398">RFC3398</a>]         Camarillo, G., Roach, A., Peterson, J., and L. Ong,
                     "Integrated Services Digital Network (ISDN) User
                     Part (ISUP) to Session Initiation Protocol (SIP)
                     Mapping", <a href="./rfc3398">RFC 3398</a>, December 2002.

   [<a id="ref-RFC3401">RFC3401</a>]         Mealling, M., "Dynamic Delegation Discovery System
                     (DDDS) Part One: The Comprehensive DDDS", <a href="./rfc3401">RFC 3401</a>,
                     October 2002.

   [<a id="ref-RFC3420">RFC3420</a>]         Sparks, R., "Internet Media Type message/sipfrag",
                     <a href="./rfc3420">RFC 3420</a>, November 2002.

   [<a id="ref-RFC3427">RFC3427</a>]         Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott,
                     J., and B. Rosen, "Change Process for the Session
                     Initiation Protocol (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp67">BCP 67</a>, <a href="./rfc3427">RFC 3427</a>,
                     December 2002.

   [<a id="ref-RFC3428">RFC3428</a>]         Campbell, B., Rosenberg, J., Schulzrinne, H.,
                     Huitema, C., and D. Gurle, "Session Initiation
                     Protocol (SIP) Extension for Instant Messaging",
                     <a href="./rfc3428">RFC 3428</a>, December 2002.

   [<a id="ref-RFC3482">RFC3482</a>]         Foster, M., McGarry, T., and J. Yu, "Number
                     Portability in the Global Switched Telephone
                     Network (GSTN): An Overview", <a href="./rfc3482">RFC 3482</a>,
                     February 2003.

   [<a id="ref-RFC3486">RFC3486</a>]         Camarillo, G., "Compressing the Session Initiation
                     Protocol (SIP)", <a href="./rfc3486">RFC 3486</a>, February 2003.

   [<a id="ref-RFC3515">RFC3515</a>]         Sparks, R., "The Session Initiation Protocol (SIP)
                     Refer Method", <a href="./rfc3515">RFC 3515</a>, April 2003.

   [<a id="ref-RFC3524">RFC3524</a>]         Camarillo, G. and A. Monrad, "Mapping of Media
                     Streams to Resource Reservation Flows", <a href="./rfc3524">RFC 3524</a>,
                     April 2003.




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   [<a id="ref-RFC3550">RFC3550</a>]         Schulzrinne, H., Casner, S., Frederick, R., and V.
                     Jacobson, "RTP: A Transport Protocol for Real-Time
                     Applications", STD 64, <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-RFC3578">RFC3578</a>]         Camarillo, G., Roach, A., Peterson, J., and L. Ong,
                     "Mapping of Integrated Services Digital Network
                     (ISDN) User Part (ISUP) Overlap Signalling to the
                     Session Initiation Protocol (SIP)", <a href="./rfc3578">RFC 3578</a>,
                     August 2003.

   [<a id="ref-RFC3581">RFC3581</a>]         Rosenberg, J. and H. Schulzrinne, "An Extension to
                     the Session Initiation Protocol (SIP) for Symmetric
                     Response Routing", <a href="./rfc3581">RFC 3581</a>, August 2003.

   [<a id="ref-RFC3605">RFC3605</a>]         Huitema, C., "Real Time Control Protocol (RTCP)
                     attribute in Session Description Protocol (SDP)",
                     <a href="./rfc3605">RFC 3605</a>, October 2003.

   [<a id="ref-RFC3608">RFC3608</a>]         Willis, D. and B. Hoeneisen, "Session Initiation
                     Protocol (SIP) Extension Header Field for Service
                     Route Discovery During Registration", <a href="./rfc3608">RFC 3608</a>,
                     October 2003.

   [<a id="ref-RFC3665">RFC3665</a>]         Johnston, A., Donovan, S., Sparks, R., Cunningham,
                     C., and K. Summers, "Session Initiation Protocol
                     (SIP) Basic Call Flow Examples", <a href="https://www.rfc-editor.org/bcp/bcp75">BCP 75</a>, <a href="./rfc3665">RFC 3665</a>,
                     December 2003.

   [<a id="ref-RFC3666">RFC3666</a>]         Johnston, A., Donovan, S., Sparks, R., Cunningham,
                     C., and K. Summers, "Session Initiation Protocol
                     (SIP) Public Switched Telephone Network (PSTN) Call
                     Flows", <a href="https://www.rfc-editor.org/bcp/bcp76">BCP 76</a>, <a href="./rfc3666">RFC 3666</a>, December 2003.

   [<a id="ref-RFC3680">RFC3680</a>]         Rosenberg, J., "A Session Initiation Protocol (SIP)
                     Event Package for Registrations", <a href="./rfc3680">RFC 3680</a>,
                     March 2004.

   [<a id="ref-RFC3725">RFC3725</a>]         Rosenberg, J., Peterson, J., Schulzrinne, H., and
                     G. Camarillo, "Best Current Practices for Third
                     Party Call Control (3pcc) in the Session Initiation
                     Protocol (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp85">BCP 85</a>, <a href="./rfc3725">RFC 3725</a>, April 2004.

   [<a id="ref-RFC3830">RFC3830</a>]         Arkko, J., Carrara, E., Lindholm, F., Naslund, M.,
                     and K. Norrman, "MIKEY: Multimedia Internet
                     KEYing", <a href="./rfc3830">RFC 3830</a>, August 2004.

   [<a id="ref-RFC3840">RFC3840</a>]         Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                     "Indicating User Agent Capabilities in the Session



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                     Initiation Protocol (SIP)", <a href="./rfc3840">RFC 3840</a>, August 2004.

   [<a id="ref-RFC3841">RFC3841</a>]         Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                     "Caller Preferences for the Session Initiation
                     Protocol (SIP)", <a href="./rfc3841">RFC 3841</a>, August 2004.

   [<a id="ref-RFC3842">RFC3842</a>]         Mahy, R., "A Message Summary and Message Waiting
                     Indication Event Package for the Session Initiation
                     Protocol (SIP)", <a href="./rfc3842">RFC 3842</a>, August 2004.

   [<a id="ref-RFC3853">RFC3853</a>]         Peterson, J., "S/MIME Advanced Encryption Standard
                     (AES) Requirement for the Session Initiation
                     Protocol (SIP)", <a href="./rfc3853">RFC 3853</a>, July 2004.

   [<a id="ref-RFC3856">RFC3856</a>]         Rosenberg, J., "A Presence Event Package for the
                     Session Initiation Protocol (SIP)", <a href="./rfc3856">RFC 3856</a>,
                     August 2004.

   [<a id="ref-RFC3857">RFC3857</a>]         Rosenberg, J., "A Watcher Information Event
                     Template-Package for the Session Initiation
                     Protocol (SIP)", <a href="./rfc3857">RFC 3857</a>, August 2004.

   [<a id="ref-RFC3890">RFC3890</a>]         Westerlund, M., "A Transport Independent Bandwidth
                     Modifier for the Session Description Protocol
                     (SDP)", <a href="./rfc3890">RFC 3890</a>, September 2004.

   [<a id="ref-RFC3891">RFC3891</a>]         Mahy, R., Biggs, B., and R. Dean, "The Session
                     Initiation Protocol (SIP) "Replaces" Header",
                     <a href="./rfc3891">RFC 3891</a>, September 2004.

   [<a id="ref-RFC3892">RFC3892</a>]         Sparks, R., "The Session Initiation Protocol (SIP)
                     Referred-By Mechanism", <a href="./rfc3892">RFC 3892</a>, September 2004.

   [<a id="ref-RFC3893">RFC3893</a>]         Peterson, J., "Session Initiation Protocol (SIP)
                     Authenticated Identity Body (AIB) Format",
                     <a href="./rfc3893">RFC 3893</a>, September 2004.

   [<a id="ref-RFC3903">RFC3903</a>]         Niemi, A., "Session Initiation Protocol (SIP)
                     Extension for Event State Publication", <a href="./rfc3903">RFC 3903</a>,
                     October 2004.

   [<a id="ref-RFC3910">RFC3910</a>]         Gurbani, V., Brusilovsky, A., Faynberg, I., Gato,
                     J., Lu, H., and M. Unmehopa, "The SPIRITS (Services
                     in PSTN requesting Internet Services) Protocol",
                     <a href="./rfc3910">RFC 3910</a>, October 2004.

   [<a id="ref-RFC3911">RFC3911</a>]         Mahy, R. and D. Petrie, "The Session Initiation
                     Protocol (SIP) "Join" Header", <a href="./rfc3911">RFC 3911</a>,



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-33" ></span>
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                     October 2004.

   [<a id="ref-RFC3959">RFC3959</a>]         Camarillo, G., "The Early Session Disposition Type
                     for the Session Initiation Protocol (SIP)",
                     <a href="./rfc3959">RFC 3959</a>, December 2004.

   [<a id="ref-RFC3960">RFC3960</a>]         Camarillo, G. and H. Schulzrinne, "Early Media and
                     Ringing Tone Generation in the Session Initiation
                     Protocol (SIP)", <a href="./rfc3960">RFC 3960</a>, December 2004.

   [<a id="ref-RFC4028">RFC4028</a>]         Donovan, S. and J. Rosenberg, "Session Timers in
                     the Session Initiation Protocol (SIP)", <a href="./rfc4028">RFC 4028</a>,
                     April 2005.

   [<a id="ref-RFC4032">RFC4032</a>]         Camarillo, G. and P. Kyzivat, "Update to the
                     Session Initiation Protocol (SIP) Preconditions
                     Framework", <a href="./rfc4032">RFC 4032</a>, March 2005.

   [<a id="ref-RFC4091">RFC4091</a>]         Camarillo, G. and J. Rosenberg, "The Alternative
                     Network Address Types (ANAT) Semantics for the
                     Session Description Protocol (SDP) Grouping
                     Framework", <a href="./rfc4091">RFC 4091</a>, June 2005.

   [<a id="ref-RFC4117">RFC4117</a>]         Camarillo, G., Burger, E., Schulzrinne, H., and A.
                     van Wijk, "Transcoding Services Invocation in the
                     Session Initiation Protocol (SIP) Using Third Party
                     Call Control (3pcc)", <a href="./rfc4117">RFC 4117</a>, June 2005.

   [<a id="ref-RFC4145">RFC4145</a>]         Yon, D. and G. Camarillo, "TCP-Based Media
                     Transport in the Session Description Protocol
                     (SDP)", <a href="./rfc4145">RFC 4145</a>, September 2005.

   [<a id="ref-RFC4168">RFC4168</a>]         Rosenberg, J., Schulzrinne, H., and G. Camarillo,
                     "The Stream Control Transmission Protocol (SCTP) as
                     a Transport for the Session Initiation Protocol
                     (SIP)", <a href="./rfc4168">RFC 4168</a>, October 2005.

   [<a id="ref-RFC4169">RFC4169</a>]         Torvinen, V., Arkko, J., and M. Naslund, "Hypertext
                     Transfer Protocol (HTTP) Digest Authentication
                     Using Authentication and Key Agreement (AKA)
                     Version-2", <a href="./rfc4169">RFC 4169</a>, November 2005.

   [<a id="ref-RFC4235">RFC4235</a>]         Rosenberg, J., Schulzrinne, H., and R. Mahy, "An
                     INVITE-Initiated Dialog Event Package for the
                     Session Initiation Protocol (SIP)", <a href="./rfc4235">RFC 4235</a>,
                     November 2005.

   [<a id="ref-RFC4240">RFC4240</a>]         Burger, E., Van Dyke, J., and A. Spitzer, "Basic



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                     Network Media Services with SIP", <a href="./rfc4240">RFC 4240</a>,
                     December 2005.

   [<a id="ref-RFC4244">RFC4244</a>]         Barnes, M., "An Extension to the Session Initiation
                     Protocol (SIP) for Request History Information",
                     <a href="./rfc4244">RFC 4244</a>, November 2005.

   [<a id="ref-RFC4320">RFC4320</a>]         Sparks, R., "Actions Addressing Identified Issues
                     with the Session Initiation Protocol's (SIP) Non-
                     INVITE Transaction", <a href="./rfc4320">RFC 4320</a>, January 2006.

   [<a id="ref-RFC4347">RFC4347</a>]         Rescorla, E. and N. Modadugu, "Datagram Transport
                     Layer Security", <a href="./rfc4347">RFC 4347</a>, April 2006.

   [<a id="ref-RFC4411">RFC4411</a>]         Polk, J., "Extending the Session Initiation
                     Protocol (SIP) Reason Header for Preemption
                     Events", <a href="./rfc4411">RFC 4411</a>, February 2006.

   [<a id="ref-RFC4412">RFC4412</a>]         Schulzrinne, H. and J. Polk, "Communications
                     Resource Priority for the Session Initiation
                     Protocol (SIP)", <a href="./rfc4412">RFC 4412</a>, February 2006.

   [<a id="ref-RFC4458">RFC4458</a>]         Jennings, C., Audet, F., and J. Elwell, "Session
                     Initiation Protocol (SIP) URIs for Applications
                     such as Voicemail and Interactive Voice Response
                     (IVR)", <a href="./rfc4458">RFC 4458</a>, April 2006.

   [<a id="ref-RFC4474">RFC4474</a>]         Peterson, J. and C. Jennings, "Enhancements for
                     Authenticated Identity Management in the Session
                     Initiation Protocol (SIP)", <a href="./rfc4474">RFC 4474</a>, August 2006.

   [<a id="ref-RFC4483">RFC4483</a>]         Burger, E., "A Mechanism for Content Indirection in
                     Session Initiation Protocol (SIP) Messages",
                     <a href="./rfc4483">RFC 4483</a>, May 2006.

   [<a id="ref-RFC4488">RFC4488</a>]         Levin, O., "Suppression of Session Initiation
                     Protocol (SIP) REFER Method Implicit Subscription",
                     <a href="./rfc4488">RFC 4488</a>, May 2006.

   [<a id="ref-RFC4497">RFC4497</a>]         Elwell, J., Derks, F., Mourot, P., and O. Rousseau,
                     "Interworking between the Session Initiation
                     Protocol (SIP) and QSIG", <a href="https://www.rfc-editor.org/bcp/bcp117">BCP 117</a>, <a href="./rfc4497">RFC 4497</a>,
                     May 2006.

   [<a id="ref-RFC4508">RFC4508</a>]         Levin, O. and A. Johnston, "Conveying Feature Tags
                     with the Session Initiation Protocol (SIP) REFER
                     Method", <a href="./rfc4508">RFC 4508</a>, May 2006.




<span class="grey">Rosenberg                     Informational                    [Page 34]</span></pre>
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   [<a id="ref-RFC4538">RFC4538</a>]         Rosenberg, J., "Request Authorization through
                     Dialog Identification in the Session Initiation
                     Protocol (SIP)", <a href="./rfc4538">RFC 4538</a>, June 2006.

   [<a id="ref-RFC4566">RFC4566</a>]         Handley, M., Jacobson, V., and C. Perkins, "SDP:
                     Session Description Protocol", <a href="./rfc4566">RFC 4566</a>, July 2006.

   [<a id="ref-RFC4567">RFC4567</a>]         Arkko, J., Lindholm, F., Naslund, M., Norrman, K.,
                     and E. Carrara, "Key Management Extensions for
                     Session Description Protocol (SDP) and Real Time
                     Streaming Protocol (RTSP)", <a href="./rfc4567">RFC 4567</a>, July 2006.

   [<a id="ref-RFC4568">RFC4568</a>]         Andreasen, F., Baugher, M., and D. Wing, "Session
                     Description Protocol (SDP) Security Descriptions
                     for Media Streams", <a href="./rfc4568">RFC 4568</a>, July 2006.

   [<a id="ref-RFC4572">RFC4572</a>]         Lennox, J., "Connection-Oriented Media Transport
                     over the Transport Layer Security (TLS) Protocol in
                     the Session Description Protocol (SDP)", <a href="./rfc4572">RFC 4572</a>,
                     July 2006.

   [<a id="ref-RFC4574">RFC4574</a>]         Levin, O. and G. Camarillo, "The Session
                     Description Protocol (SDP) Label Attribute",
                     <a href="./rfc4574">RFC 4574</a>, August 2006.

   [<a id="ref-RFC4575">RFC4575</a>]         Rosenberg, J., Schulzrinne, H., and O. Levin, "A
                     Session Initiation Protocol (SIP) Event Package for
                     Conference State", <a href="./rfc4575">RFC 4575</a>, August 2006.

   [<a id="ref-RFC4579">RFC4579</a>]         Johnston, A. and O. Levin, "Session Initiation
                     Protocol (SIP) Call Control - Conferencing for User
                     Agents", <a href="https://www.rfc-editor.org/bcp/bcp119">BCP 119</a>, <a href="./rfc4579">RFC 4579</a>, August 2006.

   [<a id="ref-RFC4583">RFC4583</a>]         Camarillo, G., "Session Description Protocol (SDP)
                     Format for Binary Floor Control Protocol (BFCP)
                     Streams", <a href="./rfc4583">RFC 4583</a>, November 2006.

   [<a id="ref-RFC4662">RFC4662</a>]         Roach, A., Campbell, B., and J. Rosenberg, "A
                     Session Initiation Protocol (SIP) Event
                     Notification Extension for Resource Lists",
                     <a href="./rfc4662">RFC 4662</a>, August 2006.

   [<a id="ref-RFC4730">RFC4730</a>]         Burger, E. and M. Dolly, "A Session Initiation
                     Protocol (SIP) Event Package for Key Press Stimulus
                     (KPML)", <a href="./rfc4730">RFC 4730</a>, November 2006.

   [<a id="ref-RFC4733">RFC4733</a>]         Schulzrinne, H. and T. Taylor, "RTP Payload for
                     DTMF Digits, Telephony Tones, and Telephony



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                     Signals", <a href="./rfc4733">RFC 4733</a>, December 2006.

   [<a id="ref-RFC4796">RFC4796</a>]         Hautakorpi, J. and G. Camarillo, "The Session
                     Description Protocol (SDP) Content Attribute",
                     <a href="./rfc4796">RFC 4796</a>, February 2007.

   [<a id="ref-RFC4896">RFC4896</a>]         Surtees, A., West, M., and A. Roach, "Signaling
                     Compression (SigComp) Corrections and
                     Clarifications", <a href="./rfc4896">RFC 4896</a>, June 2007.

   [<a id="ref-RFC4916">RFC4916</a>]         Elwell, J., "Connected Identity in the Session
                     Initiation Protocol (SIP)", <a href="./rfc4916">RFC 4916</a>, June 2007.

   [<a id="ref-RFC4960">RFC4960</a>]         Stewart, R., "Stream Control Transmission
                     Protocol", <a href="./rfc4960">RFC 4960</a>, September 2007.

   [<a id="ref-RFC5027">RFC5027</a>]         Andreasen, F. and D. Wing, "Security Preconditions
                     for Session Description Protocol (SDP) Media
                     Streams", <a href="./rfc5027">RFC 5027</a>, October 2007.

   [<a id="ref-RFC5049">RFC5049</a>]         Bormann, C., Liu, Z., Price, R., and G. Camarillo,
                     "Applying Signaling Compression (SigComp) to the
                     Session Initiation Protocol (SIP)", <a href="./rfc5049">RFC 5049</a>,
                     December 2007.

   [<a id="ref-RFC5079">RFC5079</a>]         Rosenberg, J., "Rejecting Anonymous Requests in the
                     Session Initiation Protocol (SIP)", <a href="./rfc5079">RFC 5079</a>,
                     December 2007.

   [<a id="ref-RFC5360">RFC5360</a>]         Rosenberg, J., Camarillo, G., and D. Willis, "A
                     Framework for Consent-Based Communications in the
                     Session Initiation Protocol (SIP)", <a href="./rfc5360">RFC 5360</a>,
                     October 2008.

   [<a id="ref-RFC5361">RFC5361</a>]         Camarillo, G., "A Document Format for Requesting
                     Consent", <a href="./rfc5361">RFC 5361</a>, October 2008.

   [<a id="ref-RFC5362">RFC5362</a>]         Camarillo, G., "The Session Initiation Protocol
                     (SIP) Pending Additions Event Package", <a href="./rfc5362">RFC 5362</a>,
                     October 2008.

   [<a id="ref-RFC5363">RFC5363</a>]         Camarillo, G. and A. Roach, "Framework and Security
                     Considerations for Session Initiation Protocol
                     (SIP) URI-List Services", <a href="./rfc5363">RFC 5363</a>, October 2008.

   [<a id="ref-RFC5365">RFC5365</a>]         Garcia-Martin, M. and G. Camarillo, "Multiple-
                     Recipient MESSAGE Requests in the Session
                     Initiation Protocol (SIP)", <a href="./rfc5365">RFC 5365</a>, October 2008.



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   [<a id="ref-RFC5366">RFC5366</a>]         Camarillo, G. and A. Johnston, "Conference
                     Establishment Using Request-Contained Lists in the
                     Session Initiation Protocol (SIP)", <a href="./rfc5366">RFC 5366</a>,
                     October 2008.

   [<a id="ref-RFC5367">RFC5367</a>]         Camarillo, G., Roach, A., and O. Levin,
                     "Subscriptions to Request-Contained Resource Lists
                     in the Session Initiation Protocol (SIP)",
                     <a href="./rfc5367">RFC 5367</a>, October 2008.

   [<a id="ref-RFC5368">RFC5368</a>]         Camarillo, G., Niemi, A., Isomaki, M., Garcia-
                     Martin, M., and H. Khartabil, "Referring to
                     Multiple Resources in the Session Initiation
                     Protocol (SIP)", <a href="./rfc5368">RFC 5368</a>, October 2008.

   [<a id="ref-RFC5373">RFC5373</a>]         Willis, D. and A. Allen, "Requesting Answering
                     Modes for the Session Initiation Protocol (SIP)",
                     <a href="./rfc5373">RFC 5373</a>, November 2008.

   [<a id="ref-RTCP-SUM">RTCP-SUM</a>]        Clark, A., Pendleton, A., Johnston, A., and H.
                     Sinnreich, "Session Initiation Protocol Package for
                     Voice Quality Reporting Event", Work in Progress,
                     October 2008.

   [<a id="ref-SAML">SAML</a>]            Tschofenig, H., Hodges, J., Peterson, J., Polk, J.,
                     and D. Sicker, "SIP SAML Profile and Binding", Work
                     in Progress, November 2008.

   [<a id="ref-SDP-CAP">SDP-CAP</a>]         Andreasen, F., <a style="text-decoration: none" href='https://www.google.com/search?sitesearch=datatracker.ietf.org%2Fdoc%2Fhtml%2F&amp;q=inurl:draft-+%22SDP+Capability+Negotiation%22'>"SDP Capability Negotiation"</a>, Work
                     in Progress, July 2008.

   [<a id="ref-SDP-MEDIA">SDP-MEDIA</a>]       Gilman, R., Even, R., and F. Andreasen, "SDP media
                     capabilities Negotiation", Work in Progress,
                     July 2008.

   [<a id="ref-SESSION-POLICY">SESSION-POLICY</a>]  Hilt, V., Camarillo, G., and J. Rosenberg, "A
                     Framework for Session Initiation Protocol (SIP)
                     Session Policies", Work in Progress, November 2008.

   [<a id="ref-SIMPLE">SIMPLE</a>]          Rosenberg, J., "SIMPLE made Simple: An Overview of
                     the IETF Specifications for Instant Messaging and
                     Presence using the Session Initiation Protocol
                     (SIP)", Work in Progress, October 2008.

   [<a id="ref-SIPS-URI">SIPS-URI</a>]        Audet, F., "The Use of the SIPS URI Scheme in the
                     Session Initiation Protocol (SIP)", Work
                     in Progress, November 2008.




<span class="grey">Rosenberg                     Informational                    [Page 37]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-38" ></span>
<span class="grey"><a href="./rfc5411">RFC 5411</a>                Hitchhiker's Guide to SIP           January 2009</span>


   [<a id="ref-SRTP-FRAME">SRTP-FRAME</a>]      Fischl, J., Tschofenig, H., and E. Rescorla,
                     "Framework for Establishing an SRTP Security
                     Context using DTLS", Work in Progress,
                     October 2008.

   [<a id="ref-SUBNOT-ETAGS">SUBNOT-ETAGS</a>]    Niemi, A., "An Extension to Session Initiation
                     Protocol (SIP) Events for Conditional Event
                     Notification", Work in Progress, July 2008.

   [<a id="ref-TRANSFER-MECH">TRANSFER-MECH</a>]   Garcia, M., Isomaki, M., Camarillo, G., Loreto, S.,
                     and P. Kyzivat, "A Session Description Protocol
                     (SDP) Offer/Answer Mechanism to Enable File
                     Transfer", Work in Progress, November 2008.

   [<a id="ref-UA-PRIVACY">UA-PRIVACY</a>]      Munakata, M., Schubert, S., and T. Ohba, "UA-Driven
                     Privacy Mechanism for SIP", Work in Progress,
                     October 2008.

   [<a id="ref-UPDATE-PAI">UPDATE-PAI</a>]      Elwell, J., "Updates to Asserted Identity in the
                     Session Initiation Protocol (SIP)", Work
                     in Progress, October 2008.

Author's Address

   Jonathan Rosenberg
   Cisco
   Iselin, NJ
   US

   EMail: jdrosen@cisco.com
   URI:   <a href="http://www.jdrosen.net">http://www.jdrosen.net</a>




















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<span class="grey"><a href="./rfc5411">RFC 5411</a>                Hitchhiker's Guide to SIP           January 2009</span>


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