[Docs] [txt|pdf|xml] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-rosenberg-sip-hitchhikers-guide) 00 01 02 03 04 05 06 RFC 5411

SIP                                                         J. Rosenberg
Internet-Draft                                                     Cisco
Intended status: Informational                          November 3, 2008
Expires: May 7, 2009


     A Hitchhiker's Guide to the Session Initiation Protocol (SIP)
                  draft-ietf-sip-hitchhikers-guide-06

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on May 7, 2009.

Copyright Notice

   Copyright (C) The IETF Trust (2008).

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.




Rosenberg                  Expires May 7, 2009                  [Page 1]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Scope of this Document . . . . . . . . . . . . . . . . . . . .  5
   3.  Core SIP Specifications  . . . . . . . . . . . . . . . . . . .  6
   4.  Public Switched Telephone Network (PSTN) Interworking  . . . .  9
   5.  General Purpose Infrastructure Extensions  . . . . . . . . . . 11
   6.  NAT Traversal  . . . . . . . . . . . . . . . . . . . . . . . . 13
   7.  Call Control Primitives  . . . . . . . . . . . . . . . . . . . 14
   8.  Event Framework  . . . . . . . . . . . . . . . . . . . . . . . 15
   9.  Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 16
   10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 17
   11. Operations and Management  . . . . . . . . . . . . . . . . . . 18
   12. SIP Compression  . . . . . . . . . . . . . . . . . . . . . . . 19
   13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 19
   14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 20
   15. Security Mechanisms  . . . . . . . . . . . . . . . . . . . . . 22
   16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 26
   17. Instant Messaging, Presence and Multimedia . . . . . . . . . . 27
   18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 27
   19. Security Considerations  . . . . . . . . . . . . . . . . . . . 28
   20. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 28
   21. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 28
   22. Informative References . . . . . . . . . . . . . . . . . . . . 28
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 41
   Intellectual Property and Copyright Statements . . . . . . . . . . 42

























Rosenberg                  Expires May 7, 2009                  [Page 2]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


1.  Introduction

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

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

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

   E: Experimental

   B: Best Current Practice

   I: Informational

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

   Core:  The essential SIP specifications that are expected to be
      utilized for every session or registration.

   PSTN Interop:  Specifications related to interworking with the
      telephone network.

   General Purpose Infrastructure:  General purpose extensions to SIP,
      SDP and MIME, but ones that are not expected to always be used.

   NAT Traversal:  Specifications to deal with firewall and NAT
      traversal.

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







Rosenberg                  Expires May 7, 2009                  [Page 3]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   Conferencing:  Specifications for multimedia conferencing.

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

   Event Framework:  Defines 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.

   Security Mechanisms:  Specifications providing security functionality
      for SIP.

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

   Emergency Services:  SIP extensions related to emergency services.
      See [I-D.ietf-ecrit-framework] for a more complete treatment of
      additional functionality related to emergency services.

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

   Do not print all the specs cited here at once, as they might share
   the fate of the rules of Brockian Ultracricket when bound together:
   collapse under their own gravity and form a black hole [HGTTG].

   This document itself is not an update to RFC 3261 or an extension to
   SIP.  It is an informational document, meant to guide newcomers,
   implementors and deployers to the many of the specifications
   associated with SIP.



Rosenberg                  Expires May 7, 2009                  [Page 4]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


2.  Scope of this Document

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

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

   o  The basic SDP specification, RFC 4566 [RFC4566], and any
      specification that defines an extension to SDP whose primary
      purpose is to support SIP.

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

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

   The SIP change process [RFC3427] defines two types of extensions to
   SIP.  These are 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 which 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 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.






Rosenberg                  Expires May 7, 2009                  [Page 5]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


3.  Core SIP Specifications

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

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

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

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

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

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

   RFC 3263, Locating SIP Servers (S):  [RFC3263] provides DNS
      procedures for taking a SIP URI, and determining a SIP server that
      is associated with that SIP URI.  RFC 3263 is essential for any
      implementation using SIP with DNS.  RFC 3263 makes use of both DNS
      SRV records [RFC2782] and NAPTR records [RFC2915].

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

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



Rosenberg                  Expires May 7, 2009                  [Page 6]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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

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

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

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

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

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

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



Rosenberg                  Expires May 7, 2009                  [Page 7]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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

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

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

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

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

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




Rosenberg                  Expires May 7, 2009                  [Page 8]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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

   draft-ietf-sip-sips, The use of the SIPS URI Scheme in the Session
   Initiation Protocol (SIP) (S):  [I-D.ietf-sip-sips] formally updated
      RFC 3261.  It revises the processing of the SIPS URI, originally
      defined in RFC 3261, to fix many errors and problems that have
      been encountered with that mechanism.

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

   Essential Corrections to SIP:  A collection of fixes to SIP that
      address important bugs and vulnerabilities.  These include a fix
      requiring loop detection in any proxy that forks
      [I-D.ietf-sip-fork-loop-fix], a clarification on how record-
      routing works [I-D.ietf-sip-record-route-fix], and a correction to
      the IPv6 BNF [I-D.ietf-sip-ipv6-abnf-fix].


4.  Public Switched Telephone Network (PSTN) Interworking

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

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

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



Rosenberg                  Expires May 7, 2009                  [Page 9]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


      Gargle Blaster [HGTTG].

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

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

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

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

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

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

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



Rosenberg                  Expires May 7, 2009                 [Page 10]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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


5.  General Purpose Infrastructure Extensions

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

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

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

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

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



Rosenberg                  Expires May 7, 2009                 [Page 11]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


      non-interoperable extensions due to its poorly defined scope.

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

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

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

   RFC 3608, SIP Extension Header Field for Service Route Discovery
   During Registration (S):  [RFC3608] allows a client to determine,
      from a REGISTER response, a path of proxies to use in requests it
      sends outside of a dialog.  It can also be used by proxies to
      verify the Route header in client initiated requests.  In many
      respects, it is the inverse of the Path header field, but has seen
      less usage since default outbound proxies have been sufficient in
      many deployments.

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

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

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





Rosenberg                  Expires May 7, 2009                 [Page 12]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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

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

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

   draft-ietf-mmusic-sdp-media-capabilities, SDP Media Capabilities
   Negotiation (S):  [I-D.ietf-mmusic-sdp-media-capabilities] defines an
      extension to the SDP capability negotiation framework
      [I-D.ietf-mmusic-sdp-capability-negotiation] for negotiating
      codecs, codec parameters, and media streams.

   draft-ietf-sip-body-handling, Message Body Handling in the Session
   Initiation Protocol (SIP):  [I-D.ietf-sip-body-handling] clarifies
      handling of bodies in SIP, focusing primarily on multi-part
      behavior, which was underspecified in SIP.


6.  NAT Traversal

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

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

   draft-ietf-mmusic-ice-tcp, TCP Candidates with Interactive
   Connectivity Establishment (ICE) (S):  [I-D.ietf-mmusic-ice-tcp]
      specifies the usage of ICE for TCP streams.  This allows for
      selection of RTP-based voice ontop of TCP only when NAT or



Rosenberg                  Expires May 7, 2009                 [Page 13]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


      firewalls would prevent UDP-based voice from working.

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

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

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

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


7.  Call Control Primitives

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

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

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





Rosenberg                  Expires May 7, 2009                 [Page 14]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


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

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

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

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


8.  Event Framework

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

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

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

   draft-ietf-sip-subnot-etags, An Extension to Session Initiation
   Protocol  (SIP) Events for Conditional Event Notification (S):
      [I-D.ietf-sip-subnot-etags] defines an extension to RFC 3265 to
      optimize the performance of notifications.  When a client



Rosenberg                  Expires May 7, 2009                 [Page 15]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


      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.


9.  Event Packages

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

   RFC 3680, A SIP Event Package for Registrations (S):  [RFC3680]
      defines an event package for finding out about changes in
      registration state.

   draft-ietf-sipping-gruu-reg-event (S):
      [I-D.ietf-sipping-gruu-reg-event] is an extension to the
      registration event package [RFC3680] that allows user agents to
      learn about their GRUUs.  It is particularly useful in helping to
      synchronize a client and its registrar with its currently valid
      temporary GRUU.

   RFC 3842, A Message Summary and Message Waiting Indication Event
   Package for SIP (S):  [RFC3482] defines a way for a user agent to
      find out about voicemails and other messages that are waiting for
      it.  Its primary purpose is to enable the voicemail waiting lamp
      on most business telephones.

   RFC 3856, A Presence Event Package for SIP (S):  [RFC3856] defines an
      event package for indicating user presence through SIP.

   RFC 3857, A Watcher Information Event Template Package for SIP (S):
      [RFC3857], also known as winfo, provides a mechanism for a user
      agent to find out what subscriptions are in place for a particular
      event package.  Its primary usage is with presence, but it can be
      used with any event package.

   RFC 4235, An INVITE Initiated Dialog Event Package for SIP (S):
      [RFC4235] 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 [HGTTG].

   RFC 4575, A SIP Event Package for Conference State (S):  [RFC4575]
      defines a mechanism for learning about changes in conference
      state, including conference membership.







Rosenberg                  Expires May 7, 2009                 [Page 16]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   RFC 4730, A SIP Event Package for Keypress Stimulus (KPML) (S):
      [RFC4730] defines a way for an application in the network to
      subscribe to the set of keypresses made on the keypad of a
      traditional telephone.  It, along with RFC 2833 [RFC2833], are the
      two mechanisms defined for handling DTMF.  RFC 4730 is a
      signaling-path solution, and RFC 2833 is a media-path solution.

   draft-ietf-sipping-rtcp-summary, SIP Event Package for Voice Quality
   Reporting (S):  [I-D.ietf-sipping-rtcp-summary] defines a SIP event
      package that enables the collection and reporting of metrics that
      measure the quality for Voice over Internet Protocol (VoIP)
      sessions.

   draft-ietf-sip-session-policy-framework, A Framework for Session
   Initiation Protocol (SIP) Session Policies (S):
      [I-D.ietf-sip-session-policy-framework] 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.

   draft-ietf-sipping-policy-package, A Session Initiation Protocol
   (SIP) Event  Package for Session-Specific Session Policies (S):
      [I-D.ietf-sipping-policy-package] defines a SIP event package used
      in conjunction with the session policy framework
      [I-D.ietf-sip-session-policy-framework].

   RFC 5362, The Session Initiation Protocol (SIP) Pending  Additions
   Event Package (S):  [RFC5362] 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 [RFC5360].


10.  Quality of Service

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

   RFC 3312, Integration of Resource Management and SIP (S):  [RFC3312],
      updated by [RFC4032] defines a way to make sure that the phone of
      the called party doesn't ring until a QoS reservation has been
      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.






Rosenberg                  Expires May 7, 2009                 [Page 17]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   draft-ietf-mmusic-qos-identification, Quality of Service (QoS)
   Mechanism Selection in the Session Description Protocol (SDP) (S):
      [I-D.ietf-mmusic-qos-identification] 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 - RSVP and
      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.

   RFC 3313, Private SIP Extensions for Media Authorization (I):
      [RFC3313] defines a P-header that provides a mechanism for passing
      an authorization token between SIP and a network QoS reservation
      protocol like RSVP.  Its purpose is to make sure network QoS is
      only granted if a client has made a SIP call through the same
      providers network.  This specification is sometimes referred to as
      the SIP walled garden specification by the truly paranoid androids
      in the SIP community.  This is because it requires coupling of
      signaling and the underlying IP network.

   RFC 3524, Mapping of Media Streams to Resource Reservation Flows
   (S):  [RFC3524] defines a usage of the SDP grouping framework for
      indicating that a set of media streams should be handled by a
      single resource reservation.


11.  Operations and Management

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

   draft-ietf-sipping-config-framework, A Framework for SIP User Agent
   Profile Delivery (S):  [I-D.ietf-sipping-config-framework] 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.

   draft-ietf-sipping-rtcp-summary, SIP Event Package for Voice Quality
   Reporting (S):  [I-D.ietf-sipping-rtcp-summary] defines a SIP event
      package that enables the collection and reporting of metrics that
      measure the quality for Voice over Internet Protocol (VoIP)
      sessions.








Rosenberg                  Expires May 7, 2009                 [Page 18]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


12.  SIP Compression

   Sigcomp [RFC3320] [RFC4896] was defined to allow compression of SIP
   messages over low bandwidth links.  Sigcomp is not formally part of
   SIP.  However, usage of Sigcomp with SIP has required extensions to
   SIP.

   RFC 3486, Compressing SIP (S):  [RFC3486] defines a SIP URI parameter
      that can be used to indicate that a SIP server supports Sigcomp.

   RFC 5049, Applying Signaling Compression (SigComp) to the  Session
   Initiation Protocol (SIP) (S):  [RFC5049] defines how to apply
      Sigcomp to SIP.


13.  SIP Service URIs

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

   RFC 3087, Control of Service Context using Request URI (I):
      [RFC3087] introduced the context of using Request URIs, encoded
      appropriately, to invoke services.

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

   RFC 5363, Framework and Security Considerations for Session
   Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List
   Services (S):  [RFC5363] 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 provides list-oriented
      services as a consequence.  For example, a SUBSCRIBE with a list
      subscribes to the URI in the list.

   RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP
   (S):  [RFC5367] uses the URI list framework [RFC5363] 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,
      and compile the resulting information and send it back to the
      subscriber.




Rosenberg                  Expires May 7, 2009                 [Page 19]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S):  [RFC5365]
      uses the URI list framework [RFC5363] and allows a client to send
      a MESSAGE to a number of recipients.

   RFC 5366, Conference Establishment Using  Request-Contained Lists in
   SIP (S):  [RFC5366] uses the URI list framework [RFC5363] and allows
      a client to ask the server to act as a conference focus and send
      an invitation to each recipient in the list.

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

   RFC 4458, Session Initiation Protocol (SIP) URIs for Applications
   such as Voicemail and Interactive Voice Response (IVR) (I):
      [RFC4458] defines a way to invoke voicemail and IVR services by
      using a SIP URI constructed in a particular way.


14.  Minor Extensions

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

   RFC 4488, Suppression of the SIP REFER Implicit Subscription (S):
      [RFC4488] defines an enhancement to REFER.  REFER normally creates
      an implicit subscription to the target of the REFER.  This
      subscription is used to pass back updates on the progress of the
      referral.  This extension allows that implicit subscription to be
      bypassed as an optimization.

   RFC 4538, Request Authorization through Dialog Identification in SIP
   (S):  [RFC4538] 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
      [I-D.ietf-sipping-app-interaction-framework].

   RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S):
      [RFC4508] defines a mechanism for carrying RFC 3840 feature tags
      in REFER.  It is useful for informing the target of the REFER
      about the characteristics of the intentended target of the
      referred request.




Rosenberg                  Expires May 7, 2009                 [Page 20]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   draft-ietf-sip-answermode, Requesting Answer Modes for SIP (S):
      [I-D.ietf-sip-answermode] 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.

   RFC 5079, Rejecting Anonymous Requests in SIP (S):  [RFC5079] defines
      a mechanism for a called party to indicate to the calling party
      that a call was rejected since the caller was anonymous.  This is
      needed for implementation of the Anonymous Call Rejection (ACR)
      feature in SIP.

   RFC 5368, Referring to Multiple Resources in SIP (S):  [RFC5368]
      allows a UA sending a REFER to ask the recipient of the REFER to
      generate multiple SIP requests, not just one.  This is useful for
      conferencing, where a client would like to ask a conference server
      to eject multiple users.

   RFC 4483, A Mechanism for Content Indirection in Session Initiation
   Protocol (SIP) Messages (S):  [RFC4483] defines a mechanism for
      content indirection.  Instead of carrying an object within a SIP
      body, a URL reference is carried instead, and the recipient
      dereferences the URL to obtain the object.  The specification has
      potential applicability for sending large instant messages, but
      has yet to find much actual use.

   RFC 3890, A Transport Independent Bandwidth Modifier for the Session
   Description Protocol (SDP) (S):  [RFC3890] specifies an SDP extension
      that allows for the description of the bandwidth for a media
      session that is independent of the underlying transport mechanism.

   RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
   Control Protocol (BFCP) Streams (S):  [RFC4583] defines a mechanism
      in SDP to signal floor control streams that use BFCP.  It is used
      for Push-To-Talk and conference floor control.

   draft-ietf-mmusic-connectivity-precon, Connectivity Preconditions for
   Session  Description Protocol Media Streams (S):
      [I-D.ietf-mmusic-connectivity-precon] defines a usage of the
      precondition framework [RFC3312].  The connectivity precondition
      makes sure that the session doesn't get established until actual
      packet connectivity is checked.

   RFC 4796, The SDP (Session Description Protocol) Content Attribute
   (S):  [RFC4796] 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.




Rosenberg                  Expires May 7, 2009                 [Page 21]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   draft-ietf-sipping-v6-transition, IPv6 Transition in the Session
   Initiation Protocol (SIP) (S):  [I-D.ietf-sipping-v6-transition]
      defines practices for interworking between IPv6 and IPv6 user
      agents.  This is done through multi-homed proxies which interwork
      IPv4 and IPv6, along with ICE [I-D.ietf-mmusic-ice] for media
      traversal.  The specification includes some minor extensions and
      clarifications to SDP in order to cover some additional cases.

   draft-ietf-sip-connect-reuse, Connection Reuse in the Session
   Initiation Protocol (SIP) (S):  [I-D.ietf-sip-connect-reuse] defines
      an extension to SIP that allows a 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.


15.  Security Mechanisms

   Several extensions provide additional security features to SIP.

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

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

   draft-ietf-sip-sips, The use of the SIPS URI Scheme in the Session
   Initiation Protocol (SIP) (S):  [I-D.ietf-sip-sips] formally updated
      RFC 3261.  It revises the processing of the SIPS URI, originally
      defined in RFC 3261, to fix many errors and problems that have
      been encountered with that mechanism.







Rosenberg                  Expires May 7, 2009                 [Page 22]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   draft-ietf-sip-domain-certs, Domain Certificates in the Session
   Initiation Protocol (SIP) (B):  [I-D.ietf-sip-domain-certs] clarifies
      the usage of SIP over TLS with regards to certificate handling,
      and defines additional procedures needed for interoperability.

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

   RFC 4567, Key Management Extensions for Session Description Protocol
   (SDP) and Real Time Streaming Protocol (RTSP) (S):  [RFC4567] defines
      extensions to SDP that allow tunneling of an key management
      protocol, namely MIKEY [RFC3830], through offer/answer exchanges.
      This mechanism is one of three SRTP keying techniques specified
      for SIP, with DTLS-SRTP [I-D.ietf-sip-dtls-srtp-framework] having
      been selected as the final solution.

   RFC 4568, Session Description Protocol (SDP) Security Descriptions
   for Media Streams (S):  [RFC4568] defines extensions to SDP that
      allow for the negotiation of keying material directly through
      offer/answer, without a separate key management protocol.  This
      mechanism, sometimes called sdescriptions, has the drawback that
      the media keys are available to any entity that has visibility to
      the SDP.  It is one of three SRTP keying techniques specified for
      SIP, with DTLS-SRTP [I-D.ietf-sip-dtls-srtp-framework] having been
      selected as the final solution.

   draft-ietf-sip-dtls-srtp-framework, Framework for Establishing an
   SRTP Security Context using DTLS (S):
      [I-D.ietf-sip-dtls-srtp-framework] defines the overall framework
      and SDP and SIP processing required to perform key management for
      RTP using Datagram TLS (DTLS) [RFC4347] directly between
      endpoints, over the media path.  It is one of three SRTP keying
      techniques specified for SIP, with DTLS-SRTP
      [I-D.ietf-sip-dtls-srtp-framework] having been selected as the
      final solution.

   draft-ietf-mmusic-sdp-dtls, Session Description Protocol (SDP)
   Indicators for Datagram Transport Layer Security (DTLS) (S):
      [I-D.ietf-mmusic-sdp-dtls] defines the usage of SDP with DTLS-
      SRTP.







Rosenberg                  Expires May 7, 2009                 [Page 23]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   RFC 3853, S/MIME AES Requirement for SIP (S):  [RFC3853] formally
      updates RFC 3261.  It is a brief specification that updates the
      cryptography mechanisms used in SIP S/MIME.  However, SIP S/MIME
      has seen very little deployment.

   draft-ietf-sip-certs, Certificate Management Service for The Session
   Initiation Protocol (SIP) (S):  [I-D.ietf-sip-certs] defines a
      certificate service for SIP whose purpose is to facilitate the
      deployment of S/MIME.  The certificate service allows clients to
      store and retrieve their own certificates, in addition to
      obtaining the certificates for other users.

   RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity
   Body (AIB) Format (S):  [RFC3893] defines a SIP message fragment
      which can be signed in order to provide an authenticated identity
      over a request.  It was an early predecessor to [RFC4474], and
      consequently AIB has seen no deployment.

   draft-ietf-sip-saml, SIP SAML Profile and Binding (S):
      [I-D.ietf-sip-saml] defines the usage of the Security Assertion
      Markup Language (SAML) within SIP, and describes how to use it in
      conjunction with SIP identity [RFC4474] to provide authenticated
      assertions about a users role or attributes.

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

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

   RFC 5362, The Session Initiation Protocol (SIP) Pending  Additions
   Event Package (S):  [RFC5362] 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 [RFC5360].






Rosenberg                  Expires May 7, 2009                 [Page 24]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   RFC 3329, Security Mechanism Agreement for SIP (S):  [RFC3329]
      defines a mechanism to prevent bid-down attacks in conjunction
      with SIP authentication.  The mechanism has seen very limited
      deployment.  It was defined as part of the 3gpp IMS specification
      suite [3GPP.24.229], 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.

   draft-ietf-sip-e2m-sec, End-to-Middle Security in SIP (S):
      [I-D.ietf-sip-e2m-sec] defines mechanisms for providing
      confidentiality and integrity for SIP message bodies sent from
      user agents to specific network intermediaries.

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

   RFC 5027, Security Preconditions for Session Description  Protocol
   Media Streams (S):  [RFC5027] defines a precondition for use with the
      preconditions framework [RFC3312].  The security precondition
      prevents a session from being established until a security media
      stream is set up.

   RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication
   Using Authentication and Key Agreement (S):  [RFC3310] 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, it's primary application is SIP.  This
      extension is usefully primarily to implementors of IMS.

   RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication
   Using Authentication and Key Agreement (AKA) Version-2 (S):
      [RFC4169] is an enhancement to [RFC3310] which further improves
      security of the authentication.

   RFC 3329, Security Mechanism Agreement for SIP (S):  [RFC3329] is a
      SIP security extension meant to be used in conjunction with IMS,
      to ensure that there are no bid-down attacks in the negotiation of
      an authentication mechanism.








Rosenberg                  Expires May 7, 2009                 [Page 25]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


16.  Conferencing

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

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

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

   RFC 4575, A SIP Event Package for Conference State (S):  [RFC4575]
      defines a mechanism for learning about changes in conference
      state, including conference membership.

   RFC 5368, Referring to Multiple Resources in SIP (S):  [RFC5368]
      allows a UA sending a REFER to ask the recipient of the REFER to
      generate multiple SIP requests, not just one.  This is useful for
      conferencing, where a client would like to ask a conference server
      to eject multiple users.

   RFC 5366, Conference Establishment Using  Request-Contained Lists in
   SIP (S):  [RFC5366] is similar to [RFC5367].  However, instead of
      subscribing to the resource, an INVITE request is sent to the
      resource, and it will act as a conference focus and generate an
      invitation to each recipient in the list.

   RFC4579, Session Initiation Protocol (SIP) Call Control -
   Conferencing for User Agents (B):  [RFC4579] defines best practice
      procedures and call flows for conferencing.  This includes
      conference creation, joining, and dial out, amongst other
      capabilities.

   RFC 4583, Session Description Protocol (SDP) Format for Binary Floor
   Control Protocol (BFCP) Streams (S):  [RFC4583] defines a mechanism
      in SDP to signal floor control streams that use BFCP.  It is used
      for Push-To-Talk and conference floor control.







Rosenberg                  Expires May 7, 2009                 [Page 26]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


17.  Instant Messaging, Presence and Multimedia

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

   RFC 3428, SIP Extension for Instant Messaging (S):  [RFC3428] defines
      the MESSAGE method, used for sending an instant message without
      setting up a session (sometimes called "page mode").

   RFC 3856, A Presence Event Package for SIP (S):  [RFC3856] defines an
      event package for indicating user presence through SIP.

   RFC 3857, A Watcher Information Event Template Package for SIP (S):
      [RFC3857], 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.

   draft-ietf-mmusic-file-transfer-mech, A Session Description Protocol
   (SDP)  Offer/Answer Mechanism to Enable File Transfer (S):
      [I-D.ietf-mmusic-file-transfer-mech] defines a mechanism for
      signaling a file transfer session with SIP.


18.  Emergency Services

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

   RFC 4411, Extending the SIP Reason Header for Preemption Events (S):
      [RFC4411] defines an extension to the Reason header, allowing a UA
      to know that its dialog was torn down because a higher priority
      session came through.

   RFC 4412, Communications Resource Priority for SIP (S):  [RFC4412]
      defines a new header field, Resource-Priority, that allows a
      session to get priority treatment from the network.

   draft-ietf-sip-location-conveyance, Location Conveyance for the
   Session Initiation Protocol (S):  [I-D.ietf-sip-location-conveyance]
      defines a mechanism for carrying location objects in SIP messages.
      This is used to convey location from a UA to an emergency call
      taker.







Rosenberg                  Expires May 7, 2009                 [Page 27]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


19.  Security Considerations

   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!"  [HGTTG].


20.  IANA Considerations

   None.


21.  Acknowledgements

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


22.  Informative References

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC2026]  Bradner, S., "The Internet Standards Process -- Revision
              3", BCP 9, RFC 2026, October 1996.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", RFC 3550, July 2003.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [I-D.ietf-mmusic-ice]
              Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address  Translator (NAT)
              Traversal for Offer/Answer Protocols",
              draft-ietf-mmusic-ice-19 (work in progress), October 2007.

   [RFC3320]  Price, R., Bormann, C., Christoffersson, J., Hannu, H.,
              Liu, Z., and J. Rosenberg, "Signaling Compression
              (SigComp)", RFC 3320, January 2003.




Rosenberg                  Expires May 7, 2009                 [Page 28]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [RFC3893]  Peterson, J., "Session Initiation Protocol (SIP)
              Authenticated Identity Body (AIB) Format", RFC 3893,
              September 2004.

   [RFC3427]  Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J.,
              and B. Rosen, "Change Process for the Session Initiation
              Protocol (SIP)", BCP 67, RFC 3427, December 2002.

   [RFC2543]  Handley, M., Schulzrinne, H., Schooler, E., and J.
              Rosenberg, "SIP: Session Initiation Protocol", RFC 2543,
              March 1999.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation
              Protocol (SIP): Locating SIP Servers", RFC 3263,
              June 2002.

   [RFC2782]  Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
              specifying the location of services (DNS SRV)", RFC 2782,
              February 2000.

   [RFC2915]  Mealling, M. and R. Daniel, "The Naming Authority Pointer
              (NAPTR) DNS Resource Record", RFC 2915, September 2000.

   [RFC3265]  Roach, A., "Session Initiation Protocol (SIP)-Specific
              Event Notification", RFC 3265, June 2002.

   [RFC3323]  Peterson, J., "A Privacy Mechanism for the Session
              Initiation Protocol (SIP)", RFC 3323, November 2002.

   [RFC3325]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", RFC 3325,
              November 2002.

   [RFC3327]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Registering Non-Adjacent
              Contacts", RFC 3327, December 2002.

   [RFC3581]  Rosenberg, J. and H. Schulzrinne, "An Extension to the
              Session Initiation Protocol (SIP) for Symmetric Response
              Routing", RFC 3581, August 2003.

   [RFC4320]  Sparks, R., "Actions Addressing Identified Issues with the
              Session Initiation Protocol's (SIP) Non-INVITE
              Transaction", RFC 4320, January 2006.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session



Rosenberg                  Expires May 7, 2009                 [Page 29]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [I-D.ietf-sip-gruu]
              Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent (UA) URIs (GRUU) in the  Session Initiation Protocol
              (SIP)", draft-ietf-sip-gruu-15 (work in progress),
              October 2007.

   [I-D.ietf-sip-outbound]
              Jennings, C. and R. Mahy, "Managing Client Initiated
              Connections in the Session Initiation Protocol  (SIP)",
              draft-ietf-sip-outbound-16 (work in progress),
              October 2008.

   [RFC2848]  Petrack, S. and L. Conroy, "The PINT Service Protocol:
              Extensions to SIP and SDP for IP Access to Telephone Call
              Services", RFC 2848, June 2000.

   [RFC3910]  Gurbani, V., Brusilovsky, A., Faynberg, I., Gato, J., Lu,
              H., and M. Unmehopa, "The SPIRITS (Services in PSTN
              requesting Internet Services) Protocol", RFC 3910,
              October 2004.

   [RFC3372]  Vemuri, A. and J. Peterson, "Session Initiation Protocol
              for Telephones (SIP-T): Context and Architectures",
              BCP 63, RFC 3372, September 2002.

   [RFC3398]  Camarillo, G., Roach, A., Peterson, J., and L. Ong,
              "Integrated Services Digital Network (ISDN) User Part
              (ISUP) to Session Initiation Protocol (SIP) Mapping",
              RFC 3398, December 2002.

   [RFC3578]  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)", RFC 3578, August 2003.

   [RFC3960]  Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
              Tone Generation in the Session Initiation Protocol (SIP)",
              RFC 3960, December 2004.

   [RFC3262]  Rosenberg, J. and H. Schulzrinne, "Reliability of
              Provisional Responses in Session Initiation Protocol
              (SIP)", RFC 3262, June 2002.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, October 2002.




Rosenberg                  Expires May 7, 2009                 [Page 30]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [RFC2976]  Donovan, S., "The SIP INFO Method", RFC 2976,
              October 2000.

   [RFC3326]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, December 2002.

   [RFC3608]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Service Route Discovery
              During Registration", RFC 3608, October 2003.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840, August 2004.

   [RFC3841]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
              Preferences for the Session Initiation Protocol (SIP)",
              RFC 3841, August 2004.

   [RFC4028]  Donovan, S. and J. Rosenberg, "Session Timers in the
              Session Initiation Protocol (SIP)", RFC 4028, April 2005.

   [RFC4168]  Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The
              Stream Control Transmission Protocol (SCTP) as a Transport
              for the Session Initiation Protocol (SIP)", RFC 4168,
              October 2005.

   [RFC4244]  Barnes, M., "An Extension to the Session Initiation
              Protocol (SIP) for Request History Information", RFC 4244,
              November 2005.

   [RFC4488]  Levin, O., "Suppression of Session Initiation Protocol
              (SIP) REFER Method Implicit Subscription", RFC 4488,
              May 2006.

   [RFC4538]  Rosenberg, J., "Request Authorization through Dialog
              Identification in the Session Initiation Protocol (SIP)",
              RFC 4538, June 2006.

   [RFC4508]  Levin, O. and A. Johnston, "Conveying Feature Tags with
              the Session Initiation Protocol (SIP) REFER Method",
              RFC 4508, May 2006.

   [I-D.ietf-sip-answermode]
              Willis, D. and A. Allen, "Requesting Answering Modes for
              the Session Initiation Protocol (SIP)",
              draft-ietf-sip-answermode-07 (work in progress), May 2008.




Rosenberg                  Expires May 7, 2009                 [Page 31]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [HGTTG]    Adams, D., "The Hitchhiker's Guide to the Galaxy",
              September 1979.

   [RFC5079]  Rosenberg, J., "Rejecting Anonymous Requests in the
              Session Initiation Protocol (SIP)", RFC 5079,
              December 2007.

   [RFC5368]  Camarillo, G., Niemi, A., Isomaki, M., Garcia-Martin, M.,
              and H. Khartabil, "Referring to Multiple Resources in the
              Session Initiation Protocol (SIP)", RFC 5368,
              October 2008.

   [RFC3515]  Sparks, R., "The Session Initiation Protocol (SIP) Refer
              Method", RFC 3515, April 2003.

   [RFC3725]  Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
              Camarillo, "Best Current Practices for Third Party Call
              Control (3pcc) in the Session Initiation Protocol (SIP)",
              BCP 85, RFC 3725, April 2004.

   [RFC3891]  Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
              Protocol (SIP) "Replaces" Header", RFC 3891,
              September 2004.

   [RFC3892]  Sparks, R., "The Session Initiation Protocol (SIP)
              Referred-By Mechanism", RFC 3892, September 2004.

   [RFC3911]  Mahy, R. and D. Petrie, "The Session Initiation Protocol
              (SIP) "Join" Header", RFC 3911, October 2004.

   [RFC4117]  Camarillo, G., Burger, E., Schulzrinne, H., and A. van
              Wijk, "Transcoding Services Invocation in the Session
              Initiation Protocol (SIP) Using Third Party Call Control
              (3pcc)", RFC 4117, June 2005.

   [RFC3903]  Niemi, A., "Session Initiation Protocol (SIP) Extension
              for Event State Publication", RFC 3903, October 2004.

   [RFC3680]  Rosenberg, J., "A Session Initiation Protocol (SIP) Event
              Package for Registrations", RFC 3680, March 2004.

   [RFC3856]  Rosenberg, J., "A Presence Event Package for the Session
              Initiation Protocol (SIP)", RFC 3856, August 2004.

   [RFC3857]  Rosenberg, J., "A Watcher Information Event Template-
              Package for the Session Initiation Protocol (SIP)",
              RFC 3857, August 2004.




Rosenberg                  Expires May 7, 2009                 [Page 32]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [RFC4235]  Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE-
              Initiated Dialog Event Package for the Session Initiation
              Protocol (SIP)", RFC 4235, November 2005.

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
              Initiation Protocol (SIP) Event Package for Conference
              State", RFC 4575, August 2006.

   [RFC4730]  Burger, E. and M. Dolly, "A Session Initiation Protocol
              (SIP) Event Package for Key Press Stimulus (KPML)",
              RFC 4730, November 2006.

   [I-D.ietf-sipping-rtcp-summary]
              Clark, A., Pendleton, A., Johnston, A., and H. Sinnreich,
              "Session Initiation Protocol Package for Voice Quality
              Reporting Event", draft-ietf-sipping-rtcp-summary-05 (work
              in progress), October 2008.

   [RFC3312]  Camarillo, G., Marshall, W., and J. Rosenberg,
              "Integration of Resource Management and Session Initiation
              Protocol (SIP)", RFC 3312, October 2002.

   [RFC4032]  Camarillo, G. and P. Kyzivat, "Update to the Session
              Initiation Protocol (SIP) Preconditions Framework",
              RFC 4032, March 2005.

   [RFC3313]  Marshall, W., "Private Session Initiation Protocol (SIP)
              Extensions for Media Authorization", RFC 3313,
              January 2003.

   [I-D.ietf-sipping-config-framework]
              Channabasappa, S., "A Framework for Session Initiation
              Protocol User Agent Profile Delivery",
              draft-ietf-sipping-config-framework-15 (work in progress),
              February 2008.

   [RFC3486]  Camarillo, G., "Compressing the Session Initiation
              Protocol (SIP)", RFC 3486, February 2003.

   [RFC3482]  Foster, M., McGarry, T., and J. Yu, "Number Portability in
              the Global Switched Telephone Network (GSTN): An
              Overview", RFC 3482, February 2003.

   [RFC3087]  Campbell, B. and R. Sparks, "Control of Service Context
              using SIP Request-URI", RFC 3087, April 2001.

   [RFC4662]  Roach, A., Campbell, B., and J. Rosenberg, "A Session
              Initiation Protocol (SIP) Event Notification Extension for



Rosenberg                  Expires May 7, 2009                 [Page 33]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Resource Lists", RFC 4662, August 2006.

   [RFC5367]  Camarillo, G., Roach, A., and O. Levin, "Subscriptions to
              Request-Contained Resource Lists in the Session Initiation
              Protocol (SIP)", RFC 5367, October 2008.

   [RFC5365]  Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient
              MESSAGE Requests in the Session Initiation Protocol
              (SIP)", RFC 5365, October 2008.

   [RFC5366]  Camarillo, G. and A. Johnston, "Conference Establishment
              Using Request-Contained Lists in the Session Initiation
              Protocol (SIP)", RFC 5366, October 2008.

   [RFC3853]  Peterson, J., "S/MIME Advanced Encryption Standard (AES)
              Requirement for the Session Initiation Protocol (SIP)",
              RFC 3853, July 2004.

   [RFC3329]  Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T.
              Haukka, "Security Mechanism Agreement for the Session
              Initiation Protocol (SIP)", RFC 3329, January 2003.

   [I-D.ietf-sip-e2m-sec]
              Ono, K. and S. Tachimoto, "End-to-middle Security in the
              Session Initiation Protocol (SIP)",
              draft-ietf-sip-e2m-sec-06 (work in progress), July 2007.

   [RFC3428]  Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.,
              and D. Gurle, "Session Initiation Protocol (SIP) Extension
              for Instant Messaging", RFC 3428, December 2002.

   [RFC4411]  Polk, J., "Extending the Session Initiation Protocol (SIP)
              Reason Header for Preemption Events", RFC 4411,
              February 2006.

   [RFC4412]  Schulzrinne, H. and J. Polk, "Communications Resource
              Priority for the Session Initiation Protocol (SIP)",
              RFC 4412, February 2006.

   [I-D.ietf-sipping-app-interaction-framework]
              Rosenberg, J., "A Framework for Application Interaction in
              the Session Initiation Protocol  (SIP)",
              draft-ietf-sipping-app-interaction-framework-05 (work in
              progress), July 2005.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.




Rosenberg                  Expires May 7, 2009                 [Page 34]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [RFC3388]  Camarillo, G., Eriksson, G., Holler, J., and H.
              Schulzrinne, "Grouping of Media Lines in the Session
              Description Protocol (SDP)", RFC 3388, December 2002.

   [RFC3605]  Huitema, C., "Real Time Control Protocol (RTCP) attribute
              in Session Description Protocol (SDP)", RFC 3605,
              October 2003.

   [RFC4916]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", RFC 4916, June 2007.

   [I-D.ietf-sip-fork-loop-fix]
              Sparks, R., Lawrence, S., Hawrylyshen, A., and B. Campen,
              "Addressing an Amplification Vulnerability in Session
              Initiation Protocol  (SIP) Forking Proxies",
              draft-ietf-sip-fork-loop-fix-08 (work in progress),
              October 2008.

   [RFC3959]  Camarillo, G., "The Early Session Disposition Type for the
              Session Initiation Protocol (SIP)", RFC 3959,
              December 2004.

   [RFC3204]  Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet,
              F., Watson, M., and M. Zonoun, "MIME media types for ISUP
              and QSIG Objects", RFC 3204, December 2001.

   [RFC3420]  Sparks, R., "Internet Media Type message/sipfrag",
              RFC 3420, November 2002.

   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

   [RFC4091]  Camarillo, G. and J. Rosenberg, "The Alternative Network
              Address Types (ANAT) Semantics for the Session Description
              Protocol (SDP) Grouping Framework", RFC 4091, June 2005.

   [I-D.ietf-mmusic-ice-tcp]
              Rosenberg, J., "TCP Candidates with Interactive
              Connectivity Establishment (ICE)",
              draft-ietf-mmusic-ice-tcp-07 (work in progress),
              July 2008.

   [RFC4483]  Burger, E., "A Mechanism for Content Indirection in
              Session Initiation Protocol (SIP) Messages", RFC 4483,
              May 2006.

   [RFC3890]  Westerlund, M., "A Transport Independent Bandwidth



Rosenberg                  Expires May 7, 2009                 [Page 35]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Modifier for the Session Description Protocol (SDP)",
              RFC 3890, September 2004.

   [RFC4583]  Camarillo, G., "Session Description Protocol (SDP) Format
              for Binary Floor Control Protocol (BFCP) Streams",
              RFC 4583, November 2006.

   [RFC5027]  Andreasen, F. and D. Wing, "Security Preconditions for
              Session Description Protocol (SDP) Media Streams",
              RFC 5027, October 2007.

   [I-D.ietf-mmusic-connectivity-precon]
              Andreasen, F., Camarillo, G., Oran, D., and D. Wing,
              "Connectivity Preconditions for Session Description
              Protocol Media Streams",
              draft-ietf-mmusic-connectivity-precon-05 (work in
              progress), October 2008.

   [RFC4796]  Hautakorpi, J. and G. Camarillo, "The Session Description
              Protocol (SDP) Content Attribute", RFC 4796,
              February 2007.

   [RFC4574]  Levin, O. and G. Camarillo, "The Session Description
              Protocol (SDP) Label Attribute", RFC 4574, August 2006.

   [I-D.ietf-sipping-policy-package]
              Hilt, V. and G. Camarillo, "A Session Initiation Protocol
              (SIP) Event Package for Session-Specific  Session
              Policies.", draft-ietf-sipping-policy-package-05 (work in
              progress), July 2008.

   [RFC3524]  Camarillo, G. and A. Monrad, "Mapping of Media Streams to
              Resource Reservation Flows", RFC 3524, April 2003.

   [RFC4240]  Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network
              Media Services with SIP", RFC 4240, December 2005.

   [I-D.ietf-sip-certs]
              Jennings, C. and J. Fischl, "Certificate Management
              Service for The Session Initiation Protocol (SIP)",
              draft-ietf-sip-certs-06 (work in progress), April 2008.

   [RFC5360]  Rosenberg, J., Camarillo, G., and D. Willis, "A Framework
              for Consent-Based Communications in the Session Initiation
              Protocol (SIP)", RFC 5360, October 2008.

   [I-D.ietf-sip-saml]
              Tschofenig, H., Hodges, J., Peterson, J., Polk, J., and D.



Rosenberg                  Expires May 7, 2009                 [Page 36]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Sicker, "SIP SAML Profile and Binding",
              draft-ietf-sip-saml-04 (work in progress), July 2008.

   [RFC5362]  Camarillo, G., "The Session Initiation Protocol (SIP)
              Pending Additions Event Package", RFC 5362, October 2008.

   [RFC4572]  Lennox, J., "Connection-Oriented Media Transport over the
              Transport Layer Security (TLS) Protocol in the Session
              Description Protocol (SDP)", RFC 4572, July 2006.

   [I-D.ietf-mmusic-sdp-capability-negotiation]
              Andreasen, F., "SDP Capability Negotiation",
              draft-ietf-mmusic-sdp-capability-negotiation-09 (work in
              progress), July 2008.

   [I-D.ietf-mmusic-sdp-media-capabilities]
              Gilman, R., Even, R., and F. Andreasen, "SDP media
              capabilities Negotiation",
              draft-ietf-mmusic-sdp-media-capabilities-05 (work in
              progress), July 2008.

   [I-D.ietf-mmusic-file-transfer-mech]
              Garcia-Martin, M., Isomaki, M., Camarillo, G., Loreto, S.,
              and P. Kyzivat, "A Session Description Protocol (SDP)
              Offer/Answer Mechanism to Enable File  Transfer",
              draft-ietf-mmusic-file-transfer-mech-08 (work in
              progress), May 2008.

   [I-D.ietf-sip-ice-option-tag]
              Rosenberg, J., "Indicating Support for Interactive
              Connectivity Establishment (ICE) in the  Session
              Initiation Protocol (SIP)",
              draft-ietf-sip-ice-option-tag-02 (work in progress),
              June 2007.

   [3GPP.24.229]
              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.

   [I-D.ietf-sip-record-route-fix]
              Froment, T., Lebel, C., and B. Bonnaerens, "Addressing
              Record-Route issues in the Session Initiation Protocol
              (SIP)", draft-ietf-sip-record-route-fix-05 (work in
              progress), October 2008.

   [I-D.ietf-sip-subnot-etags]



Rosenberg                  Expires May 7, 2009                 [Page 37]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Niemi, A., "An Extension to Session Initiation Protocol
              (SIP) Events for Conditional  Event Notification",
              draft-ietf-sip-subnot-etags-03 (work in progress),
              July 2008.

   [I-D.ietf-sip-sips]
              Audet, F., "The use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", draft-ietf-sip-sips-08 (work
              in progress), February 2008.

   [RFC4896]  Surtees, A., West, M., and A. Roach, "Signaling
              Compression (SigComp) Corrections and Clarifications",
              RFC 4896, June 2007.

   [RFC5049]  Bormann, C., Liu, Z., Price, R., and G. Camarillo,
              "Applying Signaling Compression (SigComp) to the Session
              Initiation Protocol (SIP)", RFC 5049, December 2007.

   [I-D.ietf-simple-simple]
              Rosenberg, J., "SIMPLE made Simple: An Overview of the
              IETF Specifications for Instant  Messaging and Presence
              using the Session Initiation Protocol (SIP)",
              draft-ietf-simple-simple-03 (work in progress), July 2008.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol",
              RFC 4960, September 2007.

   [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
              Carrara, "Key Management Extensions for Session
              Description Protocol (SDP) and Real Time Streaming
              Protocol (RTSP)", RFC 4567, July 2006.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [I-D.ietf-sip-dtls-srtp-framework]
              Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing an SRTP Security Context using DTLS",
              draft-ietf-sip-dtls-srtp-framework-05 (work in progress),
              October 2008.

   [I-D.ietf-ecrit-framework]
              Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
              "Framework for Emergency Calling using Internet
              Multimedia", draft-ietf-ecrit-framework-06 (work in
              progress), July 2008.




Rosenberg                  Expires May 7, 2009                 [Page 38]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


   [RFC2833]  Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF
              Digits, Telephony Tones and Telephony Signals", RFC 2833,
              May 2000.

   [RFC4458]  Jennings, C., Audet, F., and J. Elwell, "Session
              Initiation Protocol (SIP) URIs for Applications such as
              Voicemail and Interactive Voice Response (IVR)", RFC 4458,
              April 2006.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

   [I-D.ietf-sipping-v6-transition]
              Camarillo, G., "IPv6 Transition in the Session Initiation
              Protocol (SIP)", draft-ietf-sipping-v6-transition-07 (work
              in progress), August 2007.

   [I-D.ietf-sipping-update-pai]
              Elwell, J., "Updates to Asserted Identity in the Session
              Initiation Protocol (SIP)",
              draft-ietf-sipping-update-pai-07 (work in progress),
              October 2008.

   [RFC3665]  Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and
              K. Summers, "Session Initiation Protocol (SIP) Basic Call
              Flow Examples", BCP 75, RFC 3665, December 2003.

   [RFC3666]  Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and
              K. Summers, "Session Initiation Protocol (SIP) Public
              Switched Telephone Network (PSTN) Call Flows", BCP 76,
              RFC 3666, December 2003.

   [I-D.ietf-sip-ipv6-abnf-fix]
              Gurbani, V., Carpenter, B., and B. Tate, "Essential
              correction for IPv6 ABNF and URI comparison in RFC3261",
              draft-ietf-sip-ipv6-abnf-fix-02 (work in progress),
              May 2008.

   [RFC4497]  Elwell, J., Derks, F., Mourot, P., and O. Rousseau,
              "Interworking between the Session Initiation Protocol
              (SIP) and QSIG", BCP 117, RFC 4497, May 2006.

   [I-D.ietf-sip-ua-privacy]
              Munakata, M., Schubert, S., and T. Ohba, "UA-Driven



Rosenberg                  Expires May 7, 2009                 [Page 39]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              Privacy Mechanism for SIP", draft-ietf-sip-ua-privacy-03
              (work in progress), October 2008.

   [I-D.ietf-sip-body-handling]
              Camarillo, G., "Message Body Handling in the Session
              Initiation Protocol (SIP)",
              draft-ietf-sip-body-handling-04 (work in progress),
              October 2008.

   [I-D.ietf-sip-domain-certs]
              Gurbani, V., Lawrence, S., and B. Laboratories, "Domain
              Certificates in the Session Initiation Protocol (SIP)",
              draft-ietf-sip-domain-certs-02 (work in progress),
              October 2008.

   [I-D.ietf-sipping-gruu-reg-event]
              Kyzivat, P., "Registration Event Package Extension for
              Session Initiation Protocol (SIP)  Globally Routable User
              Agent URIs (GRUUs)", draft-ietf-sipping-gruu-reg-event-09
              (work in progress), July 2007.

   [I-D.ietf-sip-session-policy-framework]
              Hilt, V., Camarillo, G., and J. Rosenberg, "A Framework
              for Session Initiation Protocol (SIP) Session Policies",
              draft-ietf-sip-session-policy-framework-04 (work in
              progress), August 2008.

   [I-D.ietf-mmusic-qos-identification]
              Polk, J., Dhesikan, S., and G. Camarillo, "Quality of
              Service (QoS) Mechanism Selection in the Session
              Description  Protocol (SDP)",
              draft-ietf-mmusic-qos-identification-02 (work in
              progress), October 2008.

   [RFC5363]  Camarillo, G. and A. Roach, "Framework and Security
              Considerations for Session Initiation Protocol (SIP) URI-
              List Services", RFC 5363, October 2008.

   [I-D.ietf-sip-connect-reuse]
              Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse in
              the Session Initiation Protocol (SIP)",
              draft-ietf-sip-connect-reuse-12 (work in progress),
              October 2008.

   [I-D.ietf-mmusic-sdp-dtls]
              Fischl, J. and H. Tschofenig, "Session Description
              Protocol (SDP) Indicators for Datagram Transport Layer
              Security (DTLS)", draft-ietf-mmusic-sdp-dtls-00 (work in



Rosenberg                  Expires May 7, 2009                 [Page 40]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


              progress), January 2008.

   [RFC5361]  Camarillo, G., "A Document Format for Requesting Consent",
              RFC 5361, October 2008.

   [RFC4579]  Johnston, A. and O. Levin, "Session Initiation Protocol
              (SIP) Call Control - Conferencing for User Agents",
              BCP 119, RFC 4579, August 2006.

   [I-D.ietf-sip-location-conveyance]
              Polk, J. and B. Rosen, "Location Conveyance for the
              Session Initiation Protocol",
              draft-ietf-sip-location-conveyance-10 (work in progress),
              September 2008.

   [RFC3310]  Niemi, A., Arkko, J., and V. Torvinen, "Hypertext Transfer
              Protocol (HTTP) Digest Authentication Using Authentication
              and Key Agreement (AKA)", RFC 3310, September 2002.

   [RFC4169]  Torvinen, V., Arkko, J., and M. Naslund, "Hypertext
              Transfer Protocol (HTTP) Digest Authentication Using
              Authentication and Key Agreement (AKA) Version-2",
              RFC 4169, November 2005.


Author's Address

   Jonathan Rosenberg
   Cisco
   Iselin, NJ
   US

   Email: jdrosen@cisco.com
   URI:   http://www.jdrosen.net

















Rosenberg                  Expires May 7, 2009                 [Page 41]

Internet-Draft          Hitchhiker's Guide to SIP          November 2008


Full Copyright Statement

   Copyright (C) The IETF Trust (2008).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

   This document and the information contained herein are provided on an
   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
   THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS
   OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   Intellectual Property Rights or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; nor does it represent that it has
   made any independent effort to identify any such rights.  Information
   on the procedures with respect to rights in RFC documents can be
   found in BCP 78 and BCP 79.

   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
   attempt made to obtain a general license or permission for the use of
   such proprietary rights by implementers or users of this
   specification can be obtained from the IETF on-line IPR repository at
   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights that may cover technology that may be required to implement
   this standard.  Please address the information to the IETF at
   ietf-ipr@ietf.org.


Acknowledgment

   Funding for the RFC Editor function is provided by the IETF
   Administrative Support Activity (IASA).





Rosenberg                  Expires May 7, 2009                 [Page 42]


Html markup produced by rfcmarkup 1.108, available from http://tools.ietf.org/tools/rfcmarkup/