SIPPING WG                                                       R. Mahy
Internet-Draft                                               Plantronics
Intended status: Informational                                 R. Sparks
Expires: October 18, 2008                               Estacado Systems
                                                            J. Rosenberg
                                                           Cisco Systems
                                                               D. Petrie
                                                                  SIP EZ
                                                        A. Johnston, Ed.
                                                          April 16, 2008

     A Call Control and Multi-party usage framework for the Session
                       Initiation Protocol (SIP)

Status of this Memo

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   This document defines a framework and requirements for call control
   and multi-party usage of SIP.  To enable discussion of multi-party
   features and applications we define an abstract call model for
   describing the media relationships required by many of these.  The

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   model and actions described here are specifically chosen to be
   independent of the SIP signaling and/or mixing approach chosen to
   actually setup the media relationships.  In addition to its dialog
   manipulation aspect, this framework includes requirements for
   communicating related information and events such as conference and
   session state, and session history.  This framework also describes
   other goals that embody the spirit of SIP applications as used on the

Table of Contents

   1.  Motivation and Background  . . . . . . . . . . . . . . . . . .  4
   2.  Key Concepts . . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.1.  "Conversation Space" Model . . . . . . . . . . . . . . . .  6
     2.2.  Relationship Between Conversation Space, SIP Dialogs,
           and SIP Sessions . . . . . . . . . . . . . . . . . . . . .  7
     2.3.  Signaling Models . . . . . . . . . . . . . . . . . . . . .  8
     2.4.  Mixing Models  . . . . . . . . . . . . . . . . . . . . . .  9
       2.4.1.  Tightly Coupled  . . . . . . . . . . . . . . . . . . . 10
       2.4.2.  Loosely Coupled  . . . . . . . . . . . . . . . . . . . 11
     2.5.  Conveying Information and Events . . . . . . . . . . . . . 12
     2.6.  Componentization and Decomposition . . . . . . . . . . . . 14
       2.6.1.  Media Intermediaries . . . . . . . . . . . . . . . . . 14
       2.6.2.  Mixer  . . . . . . . . . . . . . . . . . . . . . . . . 14
       2.6.3.  Transcoder . . . . . . . . . . . . . . . . . . . . . . 15
       2.6.4.  Media Relay  . . . . . . . . . . . . . . . . . . . . . 15
       2.6.5.  Queue Server . . . . . . . . . . . . . . . . . . . . . 15
       2.6.6.  Parking Place  . . . . . . . . . . . . . . . . . . . . 15
       2.6.7.  Announcements and Voice Dialogs  . . . . . . . . . . . 15
     2.7.  Use of URIs  . . . . . . . . . . . . . . . . . . . . . . . 17
       2.7.1.  Naming Users in SIP  . . . . . . . . . . . . . . . . . 18
       2.7.2.  Naming Services with SIP URIs  . . . . . . . . . . . . 19
     2.8.  Invoker Independence . . . . . . . . . . . . . . . . . . . 21
     2.9.  Billing issues . . . . . . . . . . . . . . . . . . . . . . 21
   3.  Catalog of call control actions and sample features  . . . . . 22
     3.1.  Remote Call Control Actions on Early Dialogs . . . . . . . 22
       3.1.1.  Remote Answer  . . . . . . . . . . . . . . . . . . . . 23
       3.1.2.  Remote Forward or Put  . . . . . . . . . . . . . . . . 23
       3.1.3.  Remote Busy or Error Out . . . . . . . . . . . . . . . 23
     3.2.  Remote Call Control Actions on Single Dialogs  . . . . . . 23
       3.2.1.  Remote Dial  . . . . . . . . . . . . . . . . . . . . . 23
       3.2.2.  Remote On and Off Hold . . . . . . . . . . . . . . . . 23
       3.2.3.  Remote Hangup  . . . . . . . . . . . . . . . . . . . . 23
     3.3.  Call Control Actions on Multiple Dialogs . . . . . . . . . 24
       3.3.1.  Transfer . . . . . . . . . . . . . . . . . . . . . . . 24
       3.3.2.  Take . . . . . . . . . . . . . . . . . . . . . . . . . 25
       3.3.3.  Add  . . . . . . . . . . . . . . . . . . . . . . . . . 26

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       3.3.4.  Local Join . . . . . . . . . . . . . . . . . . . . . . 27
       3.3.5.  Insert . . . . . . . . . . . . . . . . . . . . . . . . 27
       3.3.6.  Split  . . . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.7.  Near-fork  . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.8.  Far fork . . . . . . . . . . . . . . . . . . . . . . . 28
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 29
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 30
   6.  Appendix A: Example Features . . . . . . . . . . . . . . . . . 30
     6.1.  Implementation of these features . . . . . . . . . . . . . 33
       6.1.1.  Barge-in . . . . . . . . . . . . . . . . . . . . . . . 34
       6.1.2.  Call Monitoring  . . . . . . . . . . . . . . . . . . . 34
       6.1.3.  Call Park  . . . . . . . . . . . . . . . . . . . . . . 35
       6.1.4.  Call Pickup  . . . . . . . . . . . . . . . . . . . . . 35
       6.1.5.  Click-to-dial  . . . . . . . . . . . . . . . . . . . . 35
       6.1.6.  Distinctive ring . . . . . . . . . . . . . . . . . . . 36
       6.1.7.  Intercom . . . . . . . . . . . . . . . . . . . . . . . 36
       6.1.8.  Music on Hold  . . . . . . . . . . . . . . . . . . . . 36
       6.1.9.  Pre-paid calling . . . . . . . . . . . . . . . . . . . 36
       6.1.10. Single Line Extension/Multiple Line Appearance . . . . 37
       6.1.11. Speakerphone paging  . . . . . . . . . . . . . . . . . 37
       6.1.12. Voice message screening  . . . . . . . . . . . . . . . 37
       6.1.13. Voice Portal . . . . . . . . . . . . . . . . . . . . . 38
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 38
   8.  Informative References . . . . . . . . . . . . . . . . . . . . 38
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41
   Intellectual Property and Copyright Statements . . . . . . . . . . 43

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1.  Motivation and Background

   The Session Initiation Protocol [RFC3261] (SIP) was defined for the
   initiation, maintenance, and termination of sessions or calls between
   one or more users.  However, despite its origins as a large-scale
   multiparty conferencing protocol, SIP is used today primarily for
   point to point calls.  This two-party configuration is the focus of
   the SIP specification and most of its extensions.

   This document defines a framework and requirements for call control
   and multi-party usage of SIP.  Most multi-party operations manipulate
   SIP dialogs (also known as call legs) or SIP conference media policy
   to cause participants in a conversation to perceive specific media
   relationships.  In other protocols that deal with the concept of
   calls, this manipulation is known as call control.  In addition to
   its dialog or policy manipulation aspect, "call control" also
   includes communicating information and events related to manipulating
   calls, including information and events dealing with session state
   and history, conference state, user state, and even message state.

   Based on input from the SIP community, the authors compiled the
   following set of goals for SIP call control and multiparty
   o  Define Primitives, Not Services.  Allow for a handful of robust
      yet simple mechanisms that can be combined to deliver features and
      services.  Throughout this document we refer to these simple
      mechanisms as "primitives".  Primitives should be sufficiently
      robust so that when they are combined with eachother, they can be
      used to build lots of services.  However, the goal is not to
      define a provably complete set of primitives.  Note that while the
      IETF will NOT standardize behavior or services, it may define
      example services for informational purposes, as in service
      examples [I-D.ietf-sipping-service-examples].
   o  Participant oriented.  The primitives should be designed to
      provide services that are oriented around the experience of the
      participants.  The authors observe that end users of features and
      services usually don't care how a media relationship is setup.
      Their ultimate experience is based only on the resulting media and
      other externally visible characteristics.
   o  Signaling Model independent: Support both a central control and a
      peer-to-peer feature invocation model (and combinations of the
      two).  Baseline SIP already supports a centralized control model
      described in 3pcc [RFC3725], and the SIP community has expressed a
      great deal of interest in peer-to-peer or distributed call control
      using primitives such as those defined in REFER [RFC3515],
      Replaces [RFC3891], and Join [RFC3911].

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   o  Mixing Model independent: The bulk of interesting multiparty
      applications involve mixing or combining media from multiple
      participants.  This mixing can be performed by one or more of the
      participants, or by a centralized mixing resource.  The experience
      of the participants should not depend on the mixing model used.
      While most examples in this document refer to audio mixing, the
      framework applies to any media type.  In this context a "mixer"
      refers to combining media of the same type in an appropriate,
      media-specific way.  This is consistent with model described in
      the SIP conferencing framework.
   o  Invoker oriented.  Only the user who invokes a feature or a
      service needs to know exactly which service is invoked or why.
      This is good because it allows new services to be created without
      requiring new primitives from all the participants; and it allows
      for much simpler feature authorization policies, for example, when
      participation spans organizational boundaries.  As discussed in
      section 3.8, this also avoids exponential state explosion when
      combining features.  The invoker only has to manage a user
      interface or API to prevent local feature interactions.  All the
      other participants simply need to manage the feature interactions
      of a much smaller number of primitives.
   o  Primitives make full use of URIs.  URIs are a very powerful
      mechanism for describing users and services.  They represent a
      plentiful resource that can be extremely expressive and easily
      routed, translated, and manipulated--even across organizational
      boundaries.  URIs can contain special parameters and informational
      headers that need only be relevant to the owner of the namespace
      (domain) of the URI.  Just as a user who selects an http: URL need
      not understand the significance and organization of the web site
      it references, a user may encounter a SIP URI that translates into
      an email-style group alias, that plays a pre-recorded message, or
      runs some complex call-handling logic.  Note that while this may
      seem paradoxical to the previous goal, both goals can be satisfied
      by the same model.
   o  Make use of SIP headers and SIP event packages to provide SIP
      entities with information about their environment.  These should
      include information about the status / handling of dialogs on
      other user agents, information about the history of other contacts
      attempted prior to the current contact, the status of
      participants, the status of conferences, user presence
      information, and the status of messages.
   o  Encourage service decomposition, and design to make use of
      standard components using well-defined, simple interfaces.  Sample
      components include a SIP mixer, recording service, announcement
      server, and voice dialog server.  (This is not an exhaustive

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   o  Include authentication, authorization, policy, logging, and
      accounting mechanisms to allow these primitives to be used safely
      among mutually untrusted participants.  Some of these mechanisms
      may be used to assist in billing, but no specific billing system
      will be endorsed.
   o  Permit graceful fallback to baseline SIP.  Definitions for new SIP
      call control extensions/primitives must describe a graceful way to
      fallback to baseline SIP behavior.  Support for one primitive must
      not imply support for another primitive.
   o  There is no desire or goal to reinvent traditional models, such as
      the model used the H.450 family of protocols, JTAPI, or the CSTA
      call model, as these other models do not share the design goals
      presented in this document.

2.  Key Concepts

   This section introduces a number of key concepts which will be used
   to describe and explain various call control operations and services
   in the remainder of this document.  This includes the conversation
   space model, signaling and mixing models, common components, and the
   use of URIs.

2.1.  "Conversation Space" Model

   This document introduces the concept of an abstract "conversation
   space" as a set of participants who believe they are all
   communicating among one another.  Each conversation space contains
   one or more participants.

   Participants are SIP User Agents that send original media to or
   terminate and receive media from other members of the conversation
   space.  Logically, every participant in the conversation space has
   access to all the media generated in that space (this is strictly
   true if all participants share a common media type).  A SIP User
   Agent that does not contribute or consume any media is NOT a
   participant; nor is a user agent that merely forwards, transcoders,
   mixes, or selects media originating elsewhere in the conversation
   space.  [Note that a conversation space consists of zero or more SIP
   calls or SIP conferences.  A conversation space is similar to the
   definition of a "call" in some other call models.]

   Participants may represent human users or non-human users (referred
   to as robots or automatons in this document).  Some participants may
   be hidden within a conversation space.  Some examples of hidden
   participants include: robots that generate tones, images, or
   announcements during a conference to announce users arriving and
   departing, a human call center supervisor monitoring a conversation

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   between a trainee and a customer, and robots that record media for
   training or archival purposes.

   Participants may also be active or passive.  Active participants are
   expected to be intelligent enough to leave a conversation space when
   they no longer desire to participate.  (An attentive human
   participant is obviously active.)  Some robotic participants (such as
   a voice messaging system, an instant messaging agent, or a voice
   dialog system) may be active participants if they can leave the
   conversation space when there is no human interaction.  Other robots
   (for example our tone generating robot from the previous example) are
   passive participants.  A human participant "on-hold" is passive.

   An example diagram of a conversation space can be shown as a "bubble"
   or ovals, or as a "set" in curly or square brace notation.  Each set,
   oval, or "bubble" represents a conversation space.  Hidden
   participants are shown in lowercase letters.

   Note that while the term "conversation" usually applies to oral
   exchange of information, we apply the conversation space model to any
   media exchange between participants.

   { A , B }                   [ A , b, C, D ]

      .-.                 .---.
     /   \               /     \
    /  A  \             / A   b \
   (       )           (         )
    \  B  /             \ C   D /
     \   /               \     /
      '-'                 '---'

2.2.  Relationship Between Conversation Space, SIP Dialogs, and SIP

   In SIP, a call is "an informal term that refers to some communication
   between peers, generally set up for the purposes of a multimedia
   conversation."  Obviously we cannot discuss normative behavior based
   on such an intentionally vague definition.  The concept of a
   conversation space is needed because the SIP definition of call is
   not sufficiently precise for the purpose of describing the user
   experience of multiparty features.

   Do any other definitions convey the correct meaning?  SIP, and SDP
   [RFC4566] both define a conference as "a multimedia session
   identified by a common session description."  A session is defined as
   "a set of multimedia senders and receivers and the data streams
   flowing from senders to receivers."  Both of these definitions are

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   heavily oriented toward multicast sessions with little
   differentiation among participants.  As such, neither is particularly
   useful for our purposes.  In fact, the definition of "call" in some
   call models is more similar to our definition of a conversation

   Some examples of the relationship between conversation spaces, SIP
   dialogs, and SIP sessions are listed below.  In each example, a human
   user will perceive that there is a single call.
   o  A simple two-party call is a single conversation space, a single
      session, and a single dialog.
   o  A locally mixed three-way call is two sessions and two dialogs.
      It is also a single conversation space.
   o  A simple dial-in audio conference is a single conversation space,
      but is represented by as many dialogs and sessions as there are
      human participants.
   o  A multicast conference is a single conversation space, a single
      session, and as many dialogs as participants.

2.3.  Signaling Models

   Obviously to make changes to a conversation space, you must be able
   to use SIP signaling to cause these changes.  Specifically there must
   be a way to manipulate SIP dialogs (call legs) to move participants
   into and out of conversation spaces.  Although this is not as
   obvious, there also must be a way to manipulate SIP dialogs to
   include non-participant user agents that are otherwise involved in a
   conversation space (ex: B2BUAs, 3pcc controllers, mixers,
   transcoders, translators, or relays).

   Implementations may setup the media relationships described in the
   conversation space model using a centralized control model.  One
   common way to implement this using SIP is known as 3rd Party Call
   Control (3pcc) and is described in 3pcc [RFC3725].  The 3pcc approach
   relies on only the following 3 primitive operations:
   o  Create a new dialog (INVITE)
   o  Modify a dialog (reINVITE)
   o  Destroy a dialog (BYE)

   The main advantage of the 3pcc approach is that it only requires very
   basic SIP support from end systems to support call control features.
   As such, third-party call control is a natural way to handle protocol
   conversion and mid-call features.  It also has the advantage and
   disadvantage that new features can/must be implemented in one place
   only (the controller), and neither requires enhanced client
   functionality, nor takes advantage of it.

   In addition, a peer-to-peer approach is discussed at length in this

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   draft.  The primary drawback of the peer-to-peer model is additional
   complexity in the end system and authentication and management
   models.  The benefits of the peer-to-peer model include:
   o  state remains at the edges
   o  call signaling need only go through participants involved (there
      are no additional points of failure)
   o  peers can take advantage of end-to-end message integrity or

   The peer-to-peer approach relies on additional "primitive"
   operations, some of which are identified here.
   o  Replace an existing dialog
   o  Join a new dialog with an existing dialog
   o  Locally perform media forking (multi-unicast)
   o  Ask another UA to send a request on your behalf

   The peer-to-peer approach also only results in a single SIP dialog,
   directly between the two UAs.  The 3pcc approach results in two SIP
   dialogs, between each UA and the controller.  As a result, the SIP
   features and extensions that will be used during the dialog are
   limited to the those understood by the controller.  As a result, in a
   situation where both the UAs support an advanced SIP feature but the
   controller does not, the feature will not be able to be used.

   Many of the features, primitives, and actions described in this
   document also require some type of media mixing, combining, or
   selection as described in the next section.

2.4.  Mixing Models

   SIP permits a variety of mixing models, which are discussed here
   briefly.  This topic is discussed more thoroughly in the SIP
   conferencing framework [RFC4353] and cc-conferencing [RFC4579].  SIP
   supports both tightly-coupled and loosely-coupled conferencing,
   although more sophisticated behavior is available in tightly-coupled
   conferences.  In a tightly-coupled conference, a single SIP user
   agent (called the focus) has a direct dialog relationship with each
   participant (and may control non participant user agents as well).
   In a loosely-coupled conference there is no coordinated signaling
   relationships among the participants.

   For brevity, only the two most popular conferencing models are
   significantly discussed in this document (local and centralized
   mixing).  Applications of the conversation spaces model to loosely-
   coupled multicast and distributed full unicast mesh conferences are
   left as an exercise for the reader.  Note that a distributed full
   mesh conference can be used for basic conferences, but does not
   easily allow for more complex conferencing actions like splitting,

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   merging, and sidebars.

   Call control features should be designed to allow a mixer (local or
   centralized) to decide when to reduce a conference back to a 2-party
   call, or drop all the participants (for example if only two
   automatons are communicating).  The actual heuristics used to release
   calls are beyond the scope of this document, but may depend on
   properties in the conversation space, such as the number of active,
   passive, or hidden participants; and the send-only, receive-only, or
   send-and-receive orientation of various participants.

2.4.1.  Tightly Coupled

   Tightly coupled conferences utilize a central point for signaling and
   authentication known as a focus [RFC4353].  The actual media can be
   centrally mixed or distributed.  (Single) End System Mixing

   The first model we call "end system mixing".  In this model, user A
   calls user B, and they have a conversation.  At some point later, A
   decides to conference in user C. To do this, A calls C, using a
   completely separate SIP call.  This call uses a different Call-ID,
   different tags, etc.  There is no call set up directly between B and
   C. No SIP extension or external signaling is needed.  A merely
   decides to locally join two dialogs.

      B     C
       \   /
        \ /

   A receives media streams from both B and C, and mixes them.  A sends
   a stream containing A's and C's streams to B, and a stream containing
   A's and B's streams to C. Basically, user A handles both signaling
   and media mixing.  Centralized Mixing

   In a centralized mixing model, all participants have a pairwise SIP
   and media relationship with the mixer.  Common applications of
   centralized mixing include ad-hoc conferences and scheduled dial-in
   or dial-out conferences.  In the figure below, the mixer M receives
   and sends media to participants A, B, C, D, and E.

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      B     C
       \   /
        \ /
         M --- A
        / \
       /   \
      D     E  Centralized Signaling, Distributed Media

   In this conferencing model, there is a centralized controller, as in
   the dial-in and dial-out cases.  However, the centralized server
   handles signaling only.  The media is still sent directly between
   participants, using either multicast or multi-unicast.  Participants
   perform their own mixing.  Multi-unicast is when a user sends
   multiple packets (one for each recipient, addressed to that
   recipient).  This is referred to as a "Decentralized Multipoint
   Conference" in [H.323].  Full mesh media with centralized mixing is
   another approach.

2.4.2.  Loosely Coupled

   In these models, there is no point of central control of SIP
   signaling.  As in the "Centralized Signaling, Distributed Media" case
   above, all endpoints send media to all other endpoints.  Consequently
   every endpoint mixes their own media from all the other sources, and
   sends their own media to every other participant.  Large-Scale Multicast Conferences

   Large-scale multicast conferences were the original motivation for
   both the Session Description Protocol SDP [RFC4566] and SIP.  In a
   large- scale multicast conference, one or more multicast addresses
   are allocated to the conference.  Each participant joins those
   multicast groups, and sends their media to those groups.  Signaling
   is not sent to the multicast groups.  The sole purpose of the
   signaling is to inform participants of which multicast groups to
   join.  Large-scale multicast conferences are usually pre-arranged,
   with specific start and stop times.  However, multicast conferences
   do not need to be pre-arranged, so long as a mechanism exists to
   dynamically obtain a multicast address.  Full Distributed Unicast Conferencing

   In this conferencing model, each participant has both a pairwise
   media relationship and a pairwise signaling relationship with every
   other participant (a full mesh).  This model requires a mechanism to

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   maintain a consistent view of distributed state across the group.
   This is a classic hard problem in computer science.  Also, this model
   does not scale well for large numbers of participants. because for
   <n> participants the number of media and signaling relationships is
   approximately n-squared.  As a result, this model is not generally
   available in commercial implementations; to the contrary it is
   primarily the topic of research or experimental implementations.
   Note that this model assumes peer-to-peer signaling.

2.5.  Conveying Information and Events

   Participants should have access to information about the other
   participants in a conversation space, so that this information can be
   rendered to a human user or processed by an automaton.  Although some
   of this information may be available from the Request-URI or To,
   From, Contact, or other SIP headers, another mechanism of reporting
   this information is necessary.

   Many applications are driven by knowledge about the progress of calls
   and conferences.  In general these types of events allow for the
   construction of distributed applications, where the application
   requires information on dialog and conference state, but is not
   necessarily co-resident with an endpoint user agent or conference
   server.  For example, a focus involved in a conversation space may
   wish to provide URIs for conference status, and/or conference/floor

   The SIP Events [RFC3265] architecture defines general mechanisms for
   subscription to and notification of events within SIP networks.  It
   introduces the notion of a package that is a specific "instantiation"
   of the events mechanism for a well-defined set of events.

   Event packages are needed to provide the status of a user's dialogs,
   provide the status of conferences and its participants, provide user
   presence information, provide the status of registrations, and
   provide the status of user's messages.  While this is not an
   exhaustive list, these are sufficient to enable the sample features
   described in this document.

   The conference event package [RFC4575] allows users to subscribe to
   information about an entire tightly-coupled SIP conference.
   Notifications convey information about the participants such as: the
   SIP URI identifying each user, their status in the space (active,
   declined, departed), URIs to invoke other features (such as sidebar
   conversations), links to other relevant information (such as floor
   control policies), and if floor control policies are in place, the
   user's floor control status.  For conversation spaces created from
   cascaded conferences, conversation state can be gathered from

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   relevant foci and merged into a cohesive set of state.

   The dialog package [RFC4235] provides information about all the
   dialogs the target user is maintaining, what conversations the user
   in participating in, and how these are correlated.  Likewise the
   registration package [RFC3680] provides notifications when contacts
   have changed for a specific address-of-record.  The combination of
   these allows a user agent to learn about all conversations occurring
   for the entire registered contact set for an address-of-record.

   Note that user presence in SIP [RFC3856] has a close relationship
   with these later two event packages.  It is fundamental to the
   presence model that the information used to obtain user presence is
   constructed from any number of different input sources.  Examples of
   other such sources include calendaring information and uploads of
   presence documents.  These two packages can be considered another
   mechanism that allows a presence agent to determine the presence
   state of the user.  Specifically, a user presence server can act as a
   subscriber for the dialog and registration packages to obtain
   additional information that can be used to construct a presence

   The multi-party architecture may also need to provide a mechanism to
   get information about the status /handling of a dialog (for example,
   information about the history of other contacts attempted prior to
   the current contact).  Finally, the architecture should provide ample
   opportunities to present informational URIs that relate to calls,
   conversations, or dialogs in some way.  For example, consider the SIP
   Call-Info header, or Contact headers returned in a 300-class
   response.  Frequently additional information about a call or dialog
   can be fetched via non-SIP URIs.  For example, consider a web page
   for package tracking when calling a delivery company, or a web page
   with related documentation when joining a dial-in conference.  The
   use of URIs in the multiparty framework is discussed in more detail
   in Section 3.7.

   Finally the interaction of SIP with stimulus-signaling-based
   applications, that allow a user agent to interact with an application
   without knowledge of the semantics of that application, is discussed
   in the SIP application interaction framework
   [I-D.ietf-sipping-app-interaction-framework].  Stimulus signaling can
   occur to a user interface running locally with the client, or to a
   remote user interface, through media streams.  Stimulus signaling
   encompasses a wide range of mechanisms, ranging from clicking on
   hyperlinks, to pressing buttons, to traditional Dual Tone Multi
   Frequency (DTMF) input.  In all cases, stimulus signaling is
   supported through the use of markup languages, which play a key role
   in that framework.

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2.6.  Componentization and Decomposition

   This framework proposes a decomposed component architecture with a
   very loose coupling of services and components.  This means that a
   service (such as a conferencing server or an auto-attendant) need not
   be implemented as an actual server.  Rather, these services can be
   built by combining a few basic components in straightforward or
   arbitrarily complex ways.

   Since the components are easily deployed on separate boxes, by
   separate vendors, or even with separate providers, we achieve a
   separation of function that allows each piece to be developed in
   complete isolation.  We can also reuse existing components for new
   applications.  This allows rapid service creation, and the ability
   for services to be distributed across organizational domains anywhere
   in the Internet.

   For many of these components it is also desirable to discover their
   capabilities, for example querying the ability of a mixer to host a
   10 dialog conference, or to reserve resources for a specific time.
   These actions could be provided in the form of URIs, provided there
   is an a priori means of understanding their semantics.  For example
   if there is a published dictionary of operations, a way to query the
   service for the available operations and the associated URIs, the URI
   can be the interface for providing these service operations.  This
   concept is described in more detail in the context of dialog
   operations in Section 3.

2.6.1.  Media Intermediaries

   Media Intermediaries are not participants in any conversation space,
   although an entity that is also a media translator may also have a
   co-located participant component (for example a mixer that also
   announces the arrival of a new participant; the announcement portion
   is a participant, but the mixer itself is not).  Media intermediaries
   should be as transparent as possible to the end users--offering a
   useful, fundamental service; without getting in the way of new
   features implemented by participants.  Some common media
   intermediaries are described below.

2.6.2.  Mixer

   A SIP mixer is a component that combines media from all dialogs in
   the same conversation in a media specific way.  For example, the
   default combining for an audio conference might be an N-1
   configuration, while a text mixer might interleave text messages on a
   per-line basis.  More details about how to manipulate the media
   policy used by mixers is being discussed in the XCON Working Group.

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2.6.3.  Transcoder

   A transcoder translates media from one encoding or format to another
   (for example, GSM voice to G.711, MPEG2 to H.261, or text/html to
   text/plain), or from one media type to another (for example text to
   speech).  A more thorough discussion of transcoding is described in
   SIP transcoding services invocation

2.6.4.  Media Relay

   A media relay terminates media and simply forwards it to a new
   destination without changing the content in any way.  Sometimes media
   relays are used to provide source IP address anonymity, to facilitate
   middlebox traversal, or to provide a trusted entity where media can
   be forcefully disconnected.

2.6.5.  Queue Server

   A queue server is a location where calls can be entered into one of
   several FIFO (first-in, first-out) queues.  A queue server would
   subscribe to the presence of groups or individuals who are interested
   in its queues.  When detecting that a user is available to service a
   queue, the server redirects or transfers the last call in the
   relevant queue to the available user.  On a queue-by-queue basis,
   authorized users could also subscribe to the call state (dialog
   information) of calls within a queue.  Authorized users could use
   this information to effectively pluck (take) a call out of the queue
   (for example by sending an INVITE with a Replaces header to one of
   the user agents in the queue).

2.6.6.  Parking Place

   A parking place is a location where calls can be terminated
   temporarily and then retrieved later.  While a call is "parked", it
   can receive media "on-hold" such as music, announcements, or
   advertisements.  Such a service could be further decomposed such that
   announcements or music are handled by a separate component.

2.6.7.  Announcements and Voice Dialogs

   An announcement server is a server that can play digitized media
   (frequently audio), such as music or recorded speech.  These servers
   are typically accessible via SIP, HTTP, or RTSP.  An analogous
   service is a recording service that stores digitized media.  A
   convention for specifying announcements in SIP URIs is described in
   [RFC4240].  Likewise the same server could easily provide a service
   that records digitized media.

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   A "voice dialog" is a model of spoken interactive behavior between a
   human and an automaton that can include synthesized speech, digitized
   audio, recognition of spoken and DTMF key input, recording of spoken
   input, and interaction with call control.  Voice dialogs frequently
   consist of forms or menus.  Forms present information and gather
   input; menus offer choices of what to do next.

   Spoken dialogs are a basic building block of applications that use
   voice.  Consider for example that a voice mail system, the
   conference-id and passcode collection system for a conferencing
   system, and complicated voice portal applications all require a voice
   dialog component.  Text-to-Speech and Automatic Speech Recognition

   Text-to-Speech (TTS) is a service that converts text into digitized
   audio.  TTS is frequently integrated into other applications, but
   when separated as a component, it provides greater opportunity for
   broad reuse.  Automatic Speech Recognition (ASR) is a service that
   attempts to decipher digitized speech based on a proposed grammar.
   Like TTS, ASR services can be embedded, or exposed so that many
   applications can take advantage of such services.  A standardized
   (decomposed) interface to access standalone TTS and ASR services is
   currently being developed in the SPEECHSC Working Group.  VoiceXML

   VoiceXML is a W3C recommendation that was designed to give authors
   control over the spoken dialog between users and applications.  The
   application and user take turns speaking: the application prompts the
   user, and the user in turn responds.  Its major goal is to bring the
   advantages of web-based development and content delivery to
   interactive voice response applications.  We believe that VoiceXML
   represents the ideal partner for SIP in the development of
   distributed IVR servers.  VoiceXML is an XML based scripting language
   for describing IVR services at an abstract level.  VoiceXML supports
   DTMF recognition, speech recognition, text-to-speech, and playing out
   of recorded media files.  The results of the data collected from the
   user are passed to a controlling entity through an HTTP POST
   operation.  The controller can then return another script, or
   terminate the interaction with the IVR server.

   A VoiceXML server also need not be implemented as a monolithic
   server.  Below is a diagram of a VoiceXML browser that is split into
   media and non-media handling parts.  The VoiceXML interpreter handles
   SIP dialog state and state within a VoiceXML document, and sends
   requests to the media component over another protocol.

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                       |             |
                       | VoiceXML    |
                       | Interpreter |
                       | (signaling) |
                         ^          ^
                         |          |
                     SIP |          | RTSP
                         |          |
                         |          |
                         v          v
            +-------------+        +-------------+
            |             |        |             |
            |  SIP UA     |   RTP  | RTSP Server |
            |             |<------>|   (media)   |
            |             |        |             |
            +-------------+        +-------------+

                   Figure : Decomposed VoiceXML Server

2.7.  Use of URIs

   All naming in SIP uses URIs.  URIs in SIP are used in a plethora of
   contexts: the Request-URI; Contact, To, From, and *-Info headers;
   application/uri bodies; and embedded in email, web pages, instant
   messages, and ENUM records.  The request-URI identifies the user or
   service that the call is destined for.

   SIP URIs embedded in informational SIP headers, SIP bodies, and non-
   SIP content can also specify methods, special parameters, headers,
   and even bodies.  For example:;method=REFER?Refer-To=

   Throughout this draft we discuss call control primitive operations.
   One of the biggest problems is defining how these operations may be
   invoked.  There are a number of ways to do this.  One way is to
   define the primitives in the protocol itself such that SIP methods
   (for example REFER) or SIP headers (for example Replaces) indicate a
   specific call control action.  Another way to invoke call control
   primitives is to define a specific Request-URI naming convention.
   Either these conventions must be shared between the client (the
   invoker) and the server, or published by or on behalf of the server.
   The former involves defining URI construction techniques (e.g.  URI

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   parameters and/or token conventions) as proposed in [RFC4240].  The
   latter technique usually involves discovering the URI via a SIP event
   package, a web page, a business card, or an Instant Message.  Yet
   another means to acquire the URIs is to define a dictionary of
   primitives with well-defined semantics and provide a means to query
   the named primitives and corresponding URIs that may be invoked on
   the service or dialogs.

2.7.1.  Naming Users in SIP

   An address-of-record, or public SIP address, is a SIP (or SIPS) URI
   that points to a domain with a location server that can map the URI
   to set of Contact URIs where the user might be available.  Typically
   the Contact URIs are populated via registration.

   Address of Record               Contacts ->

   Callee Capabilities [RFC3840] defines a set of additional parameters
   to the Contact header that define the characteristics of the user
   agent at the specified URI.  For example, there is a mobility
   parameter that indicates whether the UA is fixed or mobile.  When a
   user agent registers, it places these parameters in the Contact
   headers to characterize the URIs it is registering.  This allows a
   proxy for that domain to have information about the contact addresses
   for that user.

   When a caller sends a request, it can optionally request Caller
   Preferences [RFC3841], by including the Accept-Contact, Request-
   Disposition, and Reject-Contact headers that request certain handling
   by the proxy in the target domain.  These headers contain preferences
   that describe the set of desired URIs to which the caller would like
   their request routed.  The proxy in the target domain matches these
   preferences with the Contact characteristics originally registered by
   the target user.  The target user can also choose to run arbitrarily
   complex "Find-me" feature logic on a proxy in the target domain.

   There is a strong asymmetry in how preferences for callers and
   callees can be presented to the network.  While a caller takes an
   active role by initiating the request, the callee takes a passive
   role in waiting for requests.  This motivates the use of callee-
   supplied scripts and caller preferences included in the call request.
   This asymmetry is also reflected in the appropriate relationship
   between caller and callee preferences.  A server for a callee should
   respect the wishes of the caller to avoid certain locations, while

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   the preferences among locations has to be the callee's choice, as it
   determines where, for example, the phone rings and whether the callee
   incurs mobile telephone charges for incoming calls.

   SIP User Agent implementations are encouraged to make intelligent
   decisions based on the type of participants (active/passive, hidden,
   human/robot) in a conversation space.  This information is conveyed
   via the dialog package or in a SIP header parameter communicated
   using an appropriate SIP header.  For example, a music on hold
   service may take the sensible approach that if there are two or more
   unhidden participants, it should not provide hold music; or that it
   will not send hold music to robots.

   Multiple participants in the same conversation space may represent
   the same human user.  For example, the user may use one participant
   for video, chat, and whiteboard media on a PC and another for audio
   media on a SIP phone.  In this case, the address-of-record is the
   same for both user agents, but the Contacts are different.  In
   addition, human users may add robot participants that act on their
   behalf (for example a call recording service, or a calendar
   announcement reminder).  Call Control features in SIP should continue
   to function as expected in such an environment.

2.7.2.  Naming Services with SIP URIs

   A critical piece of defining a session level service that can be
   accessed by SIP is defining the naming of the resources within that
   service.  This point cannot be overstated.

   In the context of SIP control of application components, we take
   advantage of the fact that the left-hand-side of a standard SIP URI
   is a user part.  Most services may be thought of as user automatons
   that participate in SIP sessions.  It naturally follows that the user
   part should be utilized as a service indicator.

   For example, media servers commonly offer multiple services at a
   single host address.  Use of the user part as a service indicator
   enables service consumers to direct their requests without ambiguity.
   It has the added benefit of enabling media services to register their
   availability with SIP Registrars just as any "real" SIP user would.
   This maintains consistency and provides enhanced flexibility in the
   deployment of media services in the network.

   There has been much discussion about the potential for confusion if
   media services URIs are not readily distinguishable from other types
   of SIP UAs.  The use of a service namespace provides a mechanism to
   unambiguously identify standard interfaces while not constraining the
   development of private or experimental services.

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   In SIP, the Request-URI identifies the user or service that the call
   is destined for.  The great advantage of using URIs (specifically,
   the SIP Request-URI) as a service identifier comes because of the
   combination of two facts.  First, unlike in the PSTN, where the
   namespace (dialable telephone numbers) are limited, URIs come from an
   infinite space.  They are plentiful, and they are free.  Secondly,
   the primary function of SIP is call routing through manipulations of
   the Request-URI.  In the traditional SIP application, this URI
   represents a person.  However, the URI can also represent a service,
   as we propose here.  This means we can apply the routing services SIP
   provides to routing of calls to services.  The result - the problem
   of service invocation and service location becomes a routing problem,
   for which SIP provides a scalable and flexible solution.  Since there
   is such a vast namespace of services, we can explicitly name each
   service in a finely granular way.  This allows the distribution of
   services across the network.  For further discussion about services
   and SIP URIs, see RFC 3087 [RFC3087]

   Consider a conferencing service, where we have separated the names of
   ad-hoc conferences from scheduled conferences, we can program proxies
   to route calls for ad-hoc conferences to one set of servers, and
   calls for scheduled ones to another, possibly even in a different
   provider.  In fact, since each conference itself is given a URI, we
   can distribute conferences across servers, and easily guarantee that
   calls for the same conference always get routed to the same server.
   This is in stark contrast to conferences in the telephone network,
   where the equivalent of the URI - the phone number - is scarce.  An
   entire conferencing provider generally has one or two numbers.
   Conference IDs must be obtained through IVR interactions with the
   caller, or through a human attendant.  This makes it difficult to
   distribute conferences across servers all over the network, since the
   PSTN routing only knows about the dialed number.

   For more examples, consider the URI conventions of RFC 4240 [RFC4240]
   for media servers and RFC 4458 [RFC4458] for voicemail and IVR

   In practical applications, it is important that an invoker does not
   necessarily apply semantic rules to various URIs it did not create.
   Instead, it should allow any arbitrary string to be provisioned, and
   map the string to the desired behavior.  The administrator of a
   service may choose to provision specific conventions or mnemonic
   strings, but the application should not require it.  In any large
   installation, the system owner is likely to have pre-existing rules
   for mnemonic URIs, and any attempt by an application to define its
   own rules may create a conflict.  Implementations should allow an
   arbitrary mix of URIs from these schemes, or any other scheme that
   renders valid SIP URIs to be provisioned, rather than enforce only

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   one particular scheme.

   As we have shown, SIP URIs represent an ideal, flexible mechanism for
   describing and naming service resources, regardless if the resources
   are queues, conferences, voice dialogs, announcements, voicemail
   treatments, or phone features.

2.8.  Invoker Independence

   With functional signaling, only the invoker of features in SIP need
   to know exactly which feature they are invoking.  One of the primary
   benefits of this approach is that combinations of functional features
   work in SIP call control without requiring complex feature
   interaction matrices.  For example, let us examine the combination of
   a "transfer" of a call that is "conferenced".

   Alice calls Bob. Alice silently "conferences in" her robotic
   assistant Albert as a hidden party.  Bob transfers Alice to Carol.
   If Bob asks Alice to Replace her leg with a new one to Carol then
   both Alice and Albert should be communicating with Carol

   Using the peer-to-peer model, this combination of features works fine
   if A is doing local mixing (Alice replaces Bob's dialog with
   Carol's), or if A is using a central mixer (the mixer replaces Bob's
   dialog with Carol's).  A clever implementation using the 3pcc model
   can generate similar results.

   New extensions to the SIP Call Control Framework should attempt to
   preserve this property.

2.9.  Billing issues

   Billing in the PSTN is typically based on who initiated a call.  At
   the moment billing in a SIP network is neither consistent with
   itself, nor with the PSTN.  (A billing model for SIP should allow for
   both PSTN-style billing, and non-PSTN billing.)  The example below
   demonstrates one such inconsistency.

   Alice places a call to Bob. Alice then blind transfers Bob to Carol
   through a PSTN gateway.  In current usage of REFER, Bob may be billed
   for a call he did not initiate (his UA originated the outgoing dialog
   however).  This is not necessarily a terrible thing, but it
   demonstrates a security concern (Bob must have appropriate local
   policy to prevent fraud).  Also, Alice may wish to pay for Bob's
   session with Carol.  There should be a way to signal this in SIP.

   Likewise a Replacement call may maintain the same billing

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   relationship as a Replaced call, so if Alice first calls Carol, then
   asks Bob to Replace this call, Alice may continue to receive a bill.

   Further work in SIP billing should define a way to set or discover
   the direction of billing.

3.  Catalog of call control actions and sample features

   Call control actions can be categorized by the dialogs upon which
   they operate.  The actions may involve a single or multiple dialogs.
   These dialogs can be early or established.  Multiple dialogs may be
   related in a conversation space to form a conference or other
   interesting media topologies.

   It should be noted that it is desirable to provide a means by which a
   party can discover the actions that may be performed on a dialog.
   The interested party may be independent or related to the dialogs.
   One means of accomplishing this is through the ability to define and
   obtain URIs for these actions as described in section .

   Below are listed several call control "actions" that establish or
   modify dialogs and relate the participants in a conversation space.
   The names of the actions listed are for descriptive purposes only
   (they are not normative).  This list of actions is not meant to be

   In the examples, all actions are initiated by the user "Alice"
   represented by UA "A".

3.1.  Remote Call Control Actions on Early Dialogs

   The following are a set of actions that may be performed on a single
   early dialog.  These actions can be thought of as a set of remote
   control operations.  For example an automaton might perform the
   operation on behalf of a user.  Alternatively a user might use the
   remote control in the form of an application to perform the action on
   the early dialog of a UA that may be out of reach.  All of these
   actions correspond to telling the UA how to respond to a request to
   establish an early dialog.  These actions provide useful
   functionality for PDA, PC and server based applications that desire
   the ability to control a UA.  A proposed mechanism for this type of
   functionality is described in Remote Call Control

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3.1.1.  Remote Answer

   A dialog is in some early dialog state such as 180 Ringing.  It may
   be desirable to tell the UA to answer the dialog.  That is tell it to
   send a 200 Ok response to establish the dialog.

3.1.2.  Remote Forward or Put

   It may be desirable to tell the UA to respond with a 3xx class
   response to forward an early dialog to another UA.

3.1.3.  Remote Busy or Error Out

   It may be desirable to instruct the UA to send an error response such
   as 486 Busy Here.

3.2.  Remote Call Control Actions on Single Dialogs

   There is another useful set of actions that operate on a single
   established dialog.  These operations are useful in building
   productivity applications for aiding users to control their phone.
   For example a Customer Relationship Management (CRM) application that
   sets up calls for a user eliminating the need for the user to
   actually enter an address.  These operations can also be thought of a
   remote control actions.  A proposed mechanism for this type of
   functionality is described in Remote Call Control

3.2.1.  Remote Dial

   This action instructs the UA to initiate a dialog.  This action can
   be performed using the REFER method.

3.2.2.  Remote On and Off Hold

   This action instructs the UA to put an established dialog on hold.
   Though this operation can conceptually be performed with the REFER
   method, there is no semantics defined as to what the referred party
   should do with the SDP.  There is no way to distinguish between the
   desire to go on or off hold on a per media stream basis.

3.2.3.  Remote Hangup

   This action instructs the UA to terminate an early or established
   dialog.  A REFER request with the following Refer-To URI and Target-
   Dialog header field [RFC4538] performs this action.  Note: this
   example does not show the full set of header fields.

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   REFER SIP/2.0
   Target-Dialog: 13413098;local-tag=879738;remote-tag=023214

3.3.  Call Control Actions on Multiple Dialogs

   These actions apply to a set of related dialogs.

3.3.1.  Transfer

   This section describes how call transfer can be achieved using
   centralized (3pcc) and peer-to-peer (REFER) approaches.

   The conversation space changes as follows:

    before            after
   { A , B }  -->   { C , B }

   A replaces itself with C.

   To make this happen using the peer-to-peer approach, "A" would send
   two SIP requests.  A shorthand for those requests is shown below:

   REFER B  Refer-To:C
   BYE B

   To make this happen instead using the 3pcc approach, the controller
   sends requests represented by the shorthand below:

   INVITE C (w/SDP of B)
   reINVITE B (w/SDP of C)
   BYE A

   Features enabled by this action:

   - blind transfer
   - transfer to a central mixer (some type of conference or forking)
   - transfer to park server (park)
   - transfer to music on hold or announcement server
   - transfer to a "queue"
   - transfer to a service (such as Voice Dialogs service)
   - transition from local mixer to central mixer

   This action is frequently referred to as "completing an attended
   transfer".  It is described in more detail in cc-transfer

   Note that if a transfer requires URI hiding or privacy, then the 3pcc

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   approach can more easily implement this.  For example, if the URI of
   C needs to be hidden from B, then the use of 3pcc helps accomplish

3.3.2.  Take

   The conversation space changes as follows:

   { B , C } --> { B , A }

   A forcibly replaces C with itself.  In most uses of this primitive, A
   is just "un-replacing" itself.

   Using the peer-to-peer approach, "A" sends:

    INVITE B  Replaces: <dialog between B and C>

   Using the 3pcc approach (all requests sent from controller)

    INVITE A (w/SDP of B)
    reINVITE B (w/SDP of A)
    BYE C

   Features enabled by this action:

   - transferee completes an attended transfer
   - retrieve from central mixer (not recommended)
   - retrieve from music on hold or park
   - retrieve from queue
   - call center take
   - voice portal resuming ownership of a call it originated
   - answering-machine style screening (pickup)
   - pickup of a ringing call (i.e. early dialog)

   Note: that pick up of a ringing call has perhaps some interesting
   additional requirements.  First of all it is an early dialog as
   opposed to an established dialog.  Secondly the party which is to
   pickup the call may only wish to do so only while it is an early
   dialog.  That is in the race condition where the ringing UA accepts
   just before it receives signaling from the party wishing to take the
   call, the taking party wishes to yield or cancel the take.  The goal
   is to avoid yanking an answered call from the called party.

   This action is described in Replaces [RFC3891] and in cc-transfer

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3.3.3.  Add

   Note that the following 4 actions are described in cc-conferencing

   This is merely adding a participant to a SIP conference.  The
   conversation space changes as follows:

   { A , B } --> { A , B , C }

   A adds C to the conversation.

   Using the peer-to-peer approach, adding a party using local mixing
   requires no signaling.  To transition from a 2-party call or a
   locally mixed conference to centrally mixing A could send the
   following requests:

    REFER B  Refer-To: conference-URI
    INVITE conference-URI
    BYE B

   To add a party to a conference:

    REFER C  Refer-To: conference-URI
    REFER conference-URI  Refer-To: C

   Using the 3pcc approach to transition to centrally mixed, the
   controller would send:

    INVITE mixer leg 1 (w/SDP of A)
    INVITE mixer leg 2 (w/SDP of B)
    INVITE C (late SDP)
    reINVITE A (w/SDP of mixer leg 1)
    reINVITE B (w/SDP of mixer leg 2)
    INVITE mixer leg3 (w/SDP of C)

   To add a party to a SIP conference:

    INVITE C (late SDP)
    INVITE conference-URI (w/SDP of C)

   Features enabled:

   - standard conference feature
   - call recording
   - answering-machine style screening (screening)

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3.3.4.  Local Join

   The conversation space changes like this:

   { A , B } , { A , C }  -->  { A , B , C }

           or like this

   { A , B } , { C , D }  -->  { A , B , C , D }

   A takes two conversation spaces and joins them together into a single

   Using the peer-to-peer approach, A can mix locally, or REFER the
   participants of both conversation spaces to the same central mixer
   (as in 3.3.5).

   For the 3pcc approach, the call flows for inserting participants, and
   joining and splitting conversation spaces are tedious yet
   straightforward, so these are left as an exercise for the reader.

   Features enabled:

   - standard conference feature
   - leaving a sidebar to rejoin a larger conference

3.3.5.  Insert

   The conversation space changes like this:

   { B , C } --> { A , B , C }

   A inserts itself into a conversation space.

   A proposed mechanism for signaling this using the peer-to-peer
   approach is to send a new header in an INVITE with "joining"
   [RFC3911] semantics.  For example:

   INVITE B Join: <dialog id of B and C>

   If B accepted the INVITE, B would accept responsibility to setup the
   dialogs and mixing necessary (for example: to mix locally or to
   transfer the participants to a central mixer)

   Features enabled:

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   - barge-in
   - call center monitoring
   - call recording

3.3.6.  Split

   { A , B , C , D } --> { A , B } , { C , D }

   If using a central conference with peer-to-peer

    REFER C  Refer-To: conference-URI (new URI)
    REFER D  Refer-To: conference-URI (new URI)
    BYE C
    BYE D

   Features enabled:

   - sidebar conversations during a larger conference

3.3.7.  Near-fork

   A participates in two conversation spaces simultaneously:

   { A, B } --> { B , A } & { A , C }

   A is a participant in two conversation spaces such that A sends the
   same media to both spaces, and renders media from both spaces,
   presumably by mixing or rendering the media from both.  We can define
   that A is the "anchor" point for both forks, each of which is a
   separate conversation space.

   This action is purely local implementation (it requires no special
   signaling).  Local features such as switching calls between the
   background and foreground are possible using this media relationship.

3.3.8.  Far fork

   The conversation space diagram...

   { A, B } --> { A , B } & { B , C }

   A requests B to be the "anchor" of two conversation spaces.

   This is easily setup by creating a conference with two sub-
   conferences and setting the media policy appropriately such that B is
   a participant in both.  Media forking can also be setup using 3pcc as
   described in Section 5.1 of RFC3264 [RFC3264] (an offer/answer model
   for SDP).  The session descriptions for forking are quite complex.

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   Controllers should verify that endpoints can handle forked-media, for
   example using prior configuration.

   Features enabled:

   - barge-in
   - voice portal services
   - whisper
   - hotword detection
   - sending DTMF somewhere else

4.  Security Considerations

   Call Control primitives provide a powerful set of features that can
   be dangerous in the hands of an attacker.  To complicate matters,
   call control primitives are likely to be automatically authorized
   without direct human oversight.

   The class of attacks that are possible using these tools include the
   ability to eavesdrop on calls, disconnect calls, redirect calls,
   render irritating content (including ringing) at a user agent, cause
   an action that has billing consequences, subvert billing (theft-of-
   service), and obtain private information.  Call control extensions
   must take extra care to describe how these attacks will be prevented.

   We can also make some general observations about authorization and
   trust with respect to call control.  The security model is
   dramatically dependent on the signaling model chosen (see section

   Let us first examine the security model used in the 3pcc approach.
   All signaling goes through the controller, which is a trusted entity.
   Traditional SIP authentication and hop-by-hop encryption and message
   integrity work fine in this environment, but end-to-end encryption
   and message integrity may not be possible.

   When using the peer-to-peer approach, call control actions and
   primitives can be legitimately initiated by a) an existing
   participant in the conversation space, b) a former participant in the
   conversation space, or c) an entity trusted by one of the
   participants.  For example, a participant always initiates a
   transfer; a retrieve from Park (a take) is initiated on behalf of a
   former participant; and a barge-in (insert or far-fork) is initiated
   by a trusted entity (an operator for example).

   Authenticating requests by an existing participant or a trusted
   entity can be done with baseline SIP mechanisms.  In the case of

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   features initiated by a former participant, these should be protected
   against replay attacks by using a unique name or identifier per
   invocation.  The Replaces header exhibits this behavior as a by-
   product of its operation (once a Replaces operation is successful,
   the dialog being Replaced no longer exists).  For other requests, a
   "one-time" Request-URI may be provided to the feature invoker.

   To authorize call control primitives that trigger special behavior
   (such as an INVITE with Replaces or Join semantics), the receiving
   user agent may have trouble finding appropriate credentials with
   which to challenge or authorize the request, as the sender may be
   completely unknown to the receiver, except through the introduction
   of a third party.  These credentials need to be passed transitively
   in some way or fetched in an event body, for example.

5.  IANA Considerations

   This document required no action by IANA.

6.  Appendix A: Example Features

   Primitives are defined in terms of their ability to provide features.
   These example features should require an amply robust set of services
   to demonstrate a useful set of primitives.  They are described here
   briefly.  Note that the descriptions of these features are non-
   normative.  Some of these features are used as examples in section 6
   to demonstrate how some features may require certain media
   relationships.  Note also that this document describes a mixture of
   both features originating in the world of telephones, and features
   that are clearly Internet oriented.

   Example Feature Definitions:

   Attended Transfer - The transferring party establishes a session with
   the transfer target before completing the transfer.

   Auto Answer - Calls to a certain address or location answer
   immediately via a speakerphone.

   Automatic Callback: Alice calls Bob, but Bob is busy.  Alice would
   like Bob to call her automatically when he is available.  When Bob
   hangs up, Alice's phone rings.  When Alice answers, Bob's phone
   rings.  Bob answers and they talk.

   Barge-in - Carol interrupts Alice who has a call in-progress call
   with Bob. In some variations, Alice forcibly joins a new conversation

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   with Carol, in other variations, all three parties are placed in the
   same conversation (basically a 3-way conference).

   Blind Transfer - Alice is in a conversation with Bob. Alice asks Bob
   to contact Carol, but makes no attempt to contact Carol
   independently.  In many implementations, Alice does not verify Bob's
   success or failure in contacting Carol.

   Call Forwarding - Before a dialog is accepted it is redirected to
   another location, for example, because the originally intended
   recipient is busy, does not answer, is disconnected from the network,
   configured all requests to go somewhere else.

   Call Monitoring - A call center supervisor joins an in-progress call
   for monitoring purposes.

   Call Park - A call participant parks a call (essentially puts the
   call on hold), and then retrieves it at a later time (typically from
   another location).

   Call Pickup - A party picks up a call that was ringing at another
   location.  One variation allows the caller to choose which location,
   another variation just picks up any call in that user's "pickup

   Call Return - Alice calls Bob. Bob misses the call or is disconnected
   before he is finished talking to Alice.  Bob invokes Call return that
   calls Alice, even if Alice did not provide her real identity or
   location to Bob.

   Call Waiting - Alice is in a call, then receives another call.  Alice
   can place the first call on hold, and talk with the other caller.
   She can typically switch back and forth between the callers.

   Click-to-dial - Alice looks in her company directory for Bob. When
   she finds Bob, she clicks on a URI to call him.  Her phone rings (or
   possibly answers automatically), and when she answers, Bob's phone

   Conference Call - Three or more active, visible participants in the
   same conversation space.

   Consultative transfer - the transferring party establishes a session
   with the target and mixes both sessions together so that all three
   parties can participate, then disconnects leaving the transferee and
   transfer target with an active session.

   Distinctive ring - Incoming calls have different ring cadences or

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   sample sounds depending on the From party, the To party, or other

   Do Not Disturb - Alice selects the Do Not Disturb option.  Calls to
   her either ring briefly or not at all and are forwarded elsewhere.
   Some variations allow specially authorized callers to override this
   feature and ring Alice anyway.

   Find-Me - Alice sets up complicated rules for how she can be reached
   (possibly using CPL (Call Processing Language) [RFC3880], presence
   RFC3856 [RFC3264], or other factors).  When Bob calls Alice, his call
   is eventually routed to a temporary Contact where Alice happens to be

   Hotline - Alice picks up a phone and is immediately connected to the
   technical support hotline, for example.

   IM Conference Alerts: A user receives an notification as an Instant
   Message whenever someone joins a conference they are also in.

   Inbound Call Screening - Alice doesn't want to receive calls from
   Matt.  Inbound Screening prevents Matt from disturbing Alice.  In
   some variations this works even if Matt hides his identity.

   Intercom - Alice typically presses a button on a phone that
   immediately connects to another user or phone and causes that phone
   to play her voice over its speaker.  Some variations immediately
   setup two-way communications, other variations require another button
   to be pressed to enable a two-way conversation.

   Message Waiting - Bob calls Alice when she steps away from her phone,
   when she returns a visible or audible indicator conveys that someone
   has left her a voicemail message.  The message waiting indication may
   also convey how many messages are waiting, from whom, what time, and
   other useful pieces of information.

   Music on Hold - When Alice places a call with Bob on hold, it
   replaces its audio with streaming content such as music,
   announcements, or advertisements.

   Outbound Call Screening - Alice is paged and unknowingly calls a PSTN
   pay-service telephone number in the Caribbean, but local policy
   blocks her call, and possibly informs her why.

   Pre-paid calling - Alice pays for a certain currency or unit amount
   of calling value.  When she places a call, she provides her account
   number somehow.  If her account runs out of calling value during a
   call her call is disconnected or redirected to a service where she

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   can purchase more calling value.

   Presence-Enabled Conferencing: Alice wants to set up a conference
   call with Bob and Cathy when they all happen to be available (rather
   than scheduling a predefined time).  The server providing the
   application monitors their status, and calls all three when they are
   all "online", not idle, and not in another call.

   Single Line Extension/Multiple Line Appearance -- A group of phones
   are all treated as "extensions" of a single line.  A call for one
   rings them all.  As soon as one answers, the others stop ringing.  If
   any extension is actively in a conversation, another extension can
   "pick up" and immediately join the conversation.  This emulates the
   behavior of a home telephone line with multiple phones.

   Speakerphone paging - Alice calls the paging address and speaks.  Her
   voice is played on the speaker of every idle phone in a preconfigured
   group of phones.

   Speed dial - Alice dials an abbreviated number, or enters an alias,
   or presses a special speed dial button representing Bob. Her action
   is interpreted as if she specified the full address of Bob.

   Voice message screening - Bob calls Alice.  Alice is screening her
   calls, so Bob hears Alice's voicemail greeting.  Alice can hear Bob
   leave his message.  If she decides to talk to Bob, she can take the
   call back from the voicemail system, otherwise she can let Bob leave
   a message.  This emulates the behavior of a home telephone answering

   Voice Portal - A service that allows users to access a portal site
   using spoken dialog interaction.  For example, Alice needs to
   schedule a working dinner with her co-worker Carol.  Alice uses a
   voice portal to check Carol's flight schedule, find a restaurant near
   her hotel, make a reservation, get directions there, and page Carol
   with this information.

   Whispered call waiting - Alice is in a conversation with Bob. Carol
   calls Alice.  Either Carol can "whisper" to Alice directly ("Can you
   get lunch in 15 minutes?"), or an automaton whispers to Alice
   informing her that Carol is trying to reach her.

6.1.  Implementation of these features

   Example Features:

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 Attended Transfer        [I-D.ietf-sipping-cc-transfer]
 Auto Answer              [I-D.ietf-sip-answermode]
 Automatic Callback       Two person presence-based conference
 Barge-in                 Section 6.1.1
 Blind Transfer           [I-D.ietf-sipping-cc-transfer]
 Call Forwarding          Proxy or Local implementation
 Call Hold                [I-D.ietf-sipping-service-examples]
 Call Monitoring          Section 6.1.2
 Call Park                Sec 6.1.3, [I-D.ietf-sipping-service-examples]
 Call Pickup              Sec 6.1.4, [I-D.ietf-sipping-service-examples]
 Call Return              Proxy feature
 Call Waiting             Local Implementation
 Click-to-dial            Sec 6.1.5, [I-D.ietf-sipping-service-examples]
 Conference Call          [RFC4579]
 Conferencing             [RFC4579], [RFC3856]
 Consultative transfer    [I-D.ietf-sipping-cc-transfer]
 Distinctive ring         Section 6.1.6, Proxy or Local implementation
 Do Not Disturb           [RFC3856]
 Find-Me                  Proxy service based on presence
 Hotline                  Local Implementation
 IM Conference Alerts     Subscribe to conference status
 Inbound Call Screening   Proxy or Local implementation
 Intercom                 Section 6.1.7, [I-D.ietf-sip-answermode]
 Message Waiting          [RFC3842]
 Multiple Appearances     Section 6.1.10
 Music on Hold            Sec 6.1.8, [I-D.ietf-sipping-service-examples]
 Outbound Call Screening  Proxy feature
 Pre-Paid Calling         Section 6.1.9
 Single Line Extension    Section 6.1.10
 Speakerphone paging      Section 6.1.11, Speed dial + Auto Answer
 Speed dial               Local Implementation
 Voice Message Screening  Section 6.1.12
 Voice Portal             Section 6.1.13
 Whispered call waiting   Local implementation

6.1.1.  Barge-in

   Barge-in works the same as call monitoring except that it must
   indicate that the send media stream to be mixed so that all of the
   other parties can hear the stream from UA barging in.

6.1.2.  Call Monitoring

   Call monitoring is a Join operation.  The monitoring UA sends a Join
   to the dialog it wants to listen to.  It is able to discover the
   dialog via the dialog state on the monitored UA.  The monitoring UA

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   sends SDP in the INVITE that indicates receive only media.  As the UA
   is monitoring only it does not matter whether the UA indicates it
   wishes the send stream be mix or point to point.

6.1.3.  Call Park

   Call park requires the ability to: put a dialog some place, advertise
   it to users in a pickup group and to uniquely identify it in a means
   that can be communicated (including human voice).  The dialog can be
   held locally on the UA parking the dialog or alternatively
   transferred to the park service for the pickup group.  The parked
   dialog then needs to be labeled (e.g. orbit 12) in a way that can be
   communicated to the party that is to pick up the call.  The UAs in
   the pick up group discovers the parked dialog(s) via the dialog
   package from the park service.  If the dialog is parked locally the
   park service merely aggregates the parked call states from the set of
   UAs in the pickup up group.

6.1.4.  Call Pickup

   There are two different features that are called call pickup.  The
   first is the pickup of a parked dialog.  The UA from which the dialog
   is to be picked up subscribes to the dialog state of the park service
   or the UA that has locally parked the dialog.  Dialogs that are
   parked should be labeled with an identifier.  The labels are used by
   the UA to allow the user to indicate which dialog is to be picked up.
   The UA picking up the call invoked the URI in the call state that is
   labeled as replace-remote.

   The other call pickup feature involves picking up an early dialog
   (typically ringing).  This feature uses some of the same primitives
   as the pick up of a parked call.  The call state of the UA ringing
   phone is advertised using the dialog package.  The UA that is to
   pickup the early dialog subscribes either directly to the ringing UA
   or to a service aggregating the states for UAs in the pickup group.
   The call state identifies early dialogs.  The UA uses the call
   state(s) to help the user choose which early dialog that is to be
   picked up.  The UA then invokes the URI in the call state labeled as

6.1.5.  Click-to-dial

   The application or server that hosts the click-to-dial application
   captures the URI to be dialed and can setup the call using 3pcc or
   can send a REFER request to the UA that is to dial the address.  As
   users sometimes change their mind or wish to give up listing to a
   ringing or voicemail answered phone, this application illustrates the
   need to also have the ability to remotely hangup a call.

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6.1.6.  Distinctive ring

   The target UA either makes a local decision based on information in
   an incoming INVITE (To, From, Contact, Request-URI) or trusts an
   Alert-Info header provided by the caller or inserted by a trusted
   proxy.  In the latter case, the UA fetches the content described in
   the URI (typically via http) and renders it to the user.

6.1.7.  Intercom

   The UA initiates a dialog using INVITE and the Answer-Mode: Auto
   header field as described in [I-D.ietf-sip-answermode].  The called
   UA accepts the INVITE with a 200 OK and automatically enables the

   Alternatively this can be a local decision for the UA to answer based
   upon called party identification.

6.1.8.  Music on Hold

   Music on hold can be implemented a number of ways.  One way is to
   transfer the held call to a holding service.  When the UA wishes to
   take the call off hold it basically performs a take on the call from
   the holding service.  This involves subscribing to call state on the
   holding service and then invoking the URI in the call state labeled
   as replace-remote.

   Alternatively music on hold can be performed as a local mixing
   operation.  The UA holding the call can mix in the music from the
   music service via RTP (i.e. an additional dialog) or RTSP or other
   streaming media source.  This approach is simpler (i.e. the held
   dialog does not move so there is less chance of loosing them) from a
   protocol perspective, however it does use more LAN bandwidth and
   resources on the UA.

6.1.9.  Pre-paid calling

   For prepaid calling, the user's media always passes through a device
   that is trusted by the pre-paid provider.  This may be the other
   endpoint (for example a PSTN gateway).  In either case, an
   intermediary proxy or B2BUA can periodically verify the amount of
   time available on the pre-paid account, and use the session-timer
   extension to cause the trusted endpoint (gateway) or intermediary
   (media relay) to send a reINVITE before that time runs out.  During
   the reINVITE, the SIP intermediary can re-verify the account and
   insert another session-timer header.

   Note that while most pre-paid systems on the PSTN use an IVR to

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   collect the account number and destination, this isn't strictly
   necessary for a SIP-originated prepaid call.  SIP requests and SIP
   URIs are sufficiently expressive to convey the final destination, the
   provider of the prepaid service, the location from which the user is
   calling, and the prepaid account they want to use.  If a pre-paid IVR
   is used, the mechanism described below (Voice Portals) can be
   combined as well.

6.1.10.  Single Line Extension/Multiple Line Appearance

   Incoming calls ring all the extensions through basic parallel
   forking.  Each extension subscribes to dialog events from each other
   extension.  While one user has an active call, any other UA extension
   can insert itself into that conversation (it already knows the dialog
   information) in the same way as barge-in.

   Standardization work to allow line appearance numbers to be
   coordinated across a group of UAs is currently underway.

6.1.11.  Speakerphone paging

   Speakerphone paging can be implemented using either multicast or
   through a simple multipoint mixer.  In the multicast solution the
   paging UA sends a multicast INVITE with send only media in the SDP
   (see also RFC3264).  The automatic answer and enabling of the
   speakerphone is a locally configured decision on the paged UAs.  The
   paging UA sends RTP via the multicast address indicated in the SDP.

   The multipoint solution is accomplished by sending an INVITE to the
   multipoint mixer.  The mixer is configured to automatically answer
   the dialog.  The paging UA then sends REFER requests for each of the
   UAs that are to become paging speakers (The UA is likely to send out
   a single REFER that is parallel forked by the proxy server).  The UAs
   performing as paging speakers are configured to automatically answer
   based upon caller identification (e.g.  To field, URI or Referred-To

   Finally as a third option, the user agent can send a mass-invitation
   request to a conference server, which would create a conference and
   send INVITEs containing the Answer-Mode: Auto header field to all
   user agents in the paging group.

6.1.12.  Voice message screening

   At first, this is the same as call monitoring.  In this case the
   voicemail service is one of the UAs.  The UA screening the message
   monitors the call on the voicemail service, and also subscribes to
   dialog information.  If the user screening their messages decides to

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   answer, they perform a Take from the voicemail system (for example,
   send an INVITE with Replaces to the UA leaving the message)

6.1.13.  Voice Portal

   A voice portal is essentially a complex collection of voice dialogs
   used to access interesting content.  One of the most desirable call
   control features of a Voice Portal is the ability to start a new
   outgoing call from within the context of the Portal (to make a
   restaurant reservation, or return a voicemail message for example).
   Once the new call is over, the user should be able to return to the
   Portal by pressing a special key, using some DTMF sequence (ex: a
   very long pound or hash tone), or by speaking a hotword (ex: "Main

   In order to accomplish this, the Voice Portal starts with the
   following media relationship:

   { User , Voice Portal }

   The user then asks to make an outgoing call.  The Voice Portal asks
   the User to perform a Far-Fork.  In other words the Voice Portal
   wants the following media relationship:

           { Target , User }  &  { User , Voice Portal }

   The Voice Portal is now just listening for a hotword or the
   appropriate DTMF.  As soon as the user indicates they are done, the
   Voice Portal takes the call from the old Target, and we are back to
   the original media relationship.

   This feature can also be used by the account number and phone number
   collection menu in a pre-paid calling service.  A user can press a
   DTMF sequence that presents them with the appropriate menu again.

7.  Acknowledgements

   The authors would like to acknowledge Ben Campbell for his
   contributions to the document and thank AC Mahendran, John Elwell,
   and Xavier Marjou for their detailed Working Group review of the

8.  Informative References

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.

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              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

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

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

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

              Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and
              K. Summers, "Session Initiation Protocol Service
              Examples", draft-ietf-sipping-service-examples-14 (work in
              progress), February 2008.

   [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.

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

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

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

   [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.

   [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.

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   [RFC4353]  Rosenberg, J., "A Framework for Conferencing with the
              Session Initiation Protocol (SIP)", RFC 4353,
              February 2006.

              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.

              Camarillo, G., "Framework for Transcoding with the Session
              Initiation Protocol (SIP)",
              draft-ietf-sipping-transc-framework-05 (work in progress),
              December 2006.

              Sparks, R., "Session Initiation Protocol Call Control -
              Transfer", draft-ietf-sipping-cc-transfer-09 (work in
              progress), December 2007.

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

   [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.

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

              Jennings, C. and R. Mahy, "Remote Call Control in the
              Session Initiation Protocol (SIP) using the REFER  method
              and the session-oriented dialog package",
              draft-mahy-sip-remote-cc-05 (work in progress),
              March 2007.

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

   [RFC4458]  Jennings, C., Audet, F., and J. Elwell, "Session
              Initiation Protocol (SIP) URIs for Applications such as

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              Voicemail and Interactive Voice Response (IVR)", RFC 4458,
              April 2006.

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

   [RFC3880]  Lennox, J., Wu, X., and H. Schulzrinne, "Call Processing
              Language (CPL): A Language for User Control of Internet
              Telephony Services", RFC 3880, October 2004.

              Willis, D. and A. Allen, "Requesting Answering Modes for
              the Session Initiation Protocol (SIP)",
              draft-ietf-sip-answermode-06 (work in progress),
              September 2007.

   [RFC3842]  Mahy, R., "A Message Summary and Message Waiting
              Indication Event Package for the Session Initiation
              Protocol (SIP)", RFC 3842, August 2004.

Authors' Addresses

   Rohan Mahy
   345 Encincal Street
   Santa Cruz, CA


   Robert Sparks
   Estacado Systems


   Jonathan Rosenberg
   Cisco Systems


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   Dan Petrie


   Alan Johnston (editor)


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