Rfc5897
TitleIdentification of Communications Services in the Session Initiation Protocol (SIP)
AuthorJ. Rosenberg
DateJune 2010
Format:TXT, HTML
Status:INFORMATIONAL






Internet Engineering Task Force (IETF)                      J. Rosenberg
Request for Comments: 5897                                   jdrosen.net
Category: Informational                                        June 2010
ISSN: 2070-1721


               Identification of Communications Services
                in the Session Initiation Protocol (SIP)

Abstract

   This document considers the problem of service identification in the
   Session Initiation Protocol (SIP).  Service identification is the
   process of determining the user-level use case that is driving the
   signaling being utilized by the user agent (UA).  This document
   discusses the uses of service identification, and outlines several
   architectural principles behind the process.  It identifies perils
   when service identification is not done properly -- including fraud,
   interoperability failures, and stifling of innovation.  It then
   outlines a set of recommended practices for service identification.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc5897.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must



RFC 5897                    Service ID in SIP                  June 2010


   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1. Introduction ....................................................3
   2. Services and Service Identification .............................4
   3. Example Services ................................................6
      3.1. IPTV vs. Multimedia ........................................6
      3.2. Gaming vs. Voice Chat ......................................7
      3.3. Gaming vs. Voice Chat #2 ...................................7
      3.4. Configuration vs. Pager Messaging ..........................7
   4. Using Service Identification ....................................8
      4.1. Application Invocation in the User Agent ...................8
      4.2. Application Invocation in the Network ......................9
      4.3. Network Quality-of-Service Authorization ..................10
      4.4. Service Authorization .....................................10
      4.5. Accounting and Billing ....................................11
      4.6. Negotiation of Service ....................................11
      4.7. Dispatch to Devices .......................................11
   5. Key Principles of Service Identification .......................12
      5.1. Services Are a By-Product of Signaling ....................12
      5.2. Identical Signaling Produces Identical Services ...........13
      5.3. Do What I Say, Not What I Mean ............................14
      5.4. Declarative Service Identifiers Are Redundant .............15
      5.5. URIs Are Key for Differentiated Signaling .................15
   6. Perils of Declarative Service Identification ...................16
      6.1. Fraud .....................................................16
      6.2. Systematic Interoperability Failures ......................17
      6.3. Stifling of Service Innovation ............................18
   7. Recommendations ................................................20
      7.1. Use Derived Service Identification ........................20
      7.2. Design for SIP's Negotiative Expressiveness ...............20
      7.3. Presence ..................................................21
      7.4. Intra-Domain ..............................................21
      7.5. Device Dispatch ...........................................21
   8. Security Considerations ........................................22
   9. Acknowledgements ...............................................22
   10. Informative References ........................................22











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1.  Introduction

   The Session Initiation Protocol (SIP) [RFC3261] defines mechanisms
   for initiating and managing communications sessions between agents.
   SIP allows for a broad array of session types between agents.  It can
   manage audio sessions, ranging from low-bitrate voice-only up to
   multi-channel high-fidelity music.  It can manage video sessions,
   ranging from small, "talking-head" style video chat, up to high-
   definition multipoint video conferencing and ranging from low-
   bandwidth user-generated content, up to high-definition movie and TV
   content.  SIP endpoints can be anything -- adaptors that convert an
   old analog telephone to Voice over IP (VoIP), dedicated hardphones,
   fancy hardphones with rich displays and user entry capabilities,
   softphones on a PC, buddy-list and presence applications on a PC,
   dedicated videoconferencing peripherals, and speakerphones.

   This breadth of applicability is SIP's greatest asset, but it also
   introduces numerous challenges.  One of these is that, when an
   endpoint generates a SIP INVITE for a session, or receives one, that
   session can potentially be within the context of any number of
   different use cases and endpoint types.  For example, a SIP INVITE
   with a single audio stream could represent a Push-To-Talk session
   between mobile devices, a VoIP session between softphones, or audio-
   based access to stored content on a server.

   Each of these different use cases represents a different service.
   The service is the user-visible use case that is driving the behavior
   of the user agents and servers in the SIP network.

   The differing services possible with SIP have driven implementors and
   system designers to seek techniques for service identification.
   Service identification is the process of determining and/or signaling
   the specific use case that is driving the signaling being generated
   by a user agent.  At first glance, this seems harmless and easy
   enough.  It is tempting to define a new header, "Service-ID", for
   example, and have a user agent populate it with any number of well-
   known tokens that define what the service is.  It could then be
   consumed for any number of purposes.  A token placed into the
   signaling for this purpose is called a service identifier.

   Service identification and service identifiers, when used properly,
   can be beneficial.  However, when done improperly, service
   identification can lead to fraud, systemic interoperability failures,
   and a complete stifling of the innovation that SIP was meant to
   achieve.  The purpose of this document is to describe service
   identification in more detail and describe how these problems arise.





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   Section 2 begins by defining a service and the service identification
   problem.  Section 3 gives some concrete examples of services and why
   they can be challenging to identify.  Section 4 explores the ways in
   which a service identification can be utilized within a network.
   Next, Section 5 discusses the key architectural principles of service
   identification.  Section 6 describes what declarative service
   invocation is, and how it can lead to fraud, interoperability
   failures, and stifling of service innovation.

   Consequently, this document concludes that declarative service
   identification -- the process by which a user agent inserts a moniker
   into a message that defines the desired service, separate from
   explicit and well-defined protocol mechanisms -- is harmful.

   Instead of performing declarative service identification, this
   document recommends derived service identification, and gives several
   recommendations around it in Section 7:

   1.  The identity of a service should always be derived from the
       explicit signaling in the protocol messages and other contextual
       information, and never indicated by the user through a separate
       identifier placed into the message.

   2.  The process of service identification based on signaling messages
       must be designed to SIP's negotiative expressiveness, and
       therefore handle heterogeneity and not assume a fixed set of use
       cases.

   3.  Presence can help in providing URIs that can be utilized to
       connect to specific services, thereby creating explicit
       indications in the signaling that can be used to derive a service
       identity.

   4.  Service identities placed into signaling messages for the
       purposes of caching the service identity are strictly for intra-
       domain usage.

   5.  Device dispatch should be based on feature tags that map to well-
       defined SIP extensions and capabilities.  Service dispatch should
       not be based on abstract service identifiers.

2.  Services and Service Identification

   The problem of identifying services within SIP is not a new one.  The
   problem has been considered extensively in the context of presence.
   In particular, the presence data model for SIP [RFC4479] defines the
   concept of a service as one of the core notions that presence
   describes.  Services are described in Section 3.3 of RFC 4479.



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   Essentially, the service is the user-visible use case that is driving
   the behavior of the user agents and servers in the SIP network.
   Being user-visible means that there is a difference in user
   experience between two services that are different.  That user
   experience can be part of the call, or outside of the call.  Within a
   call, the user experience can be based on different media types (an
   audio call vs. a video chat), different content within a particular
   media type (stored content, such as a movie or TV session), different
   devices (a wireless device for "telephony" vs. a PC application for
   "voice chat"), different user interfaces (a buddy-list view of voice
   on a PC application vs. a software emulation of a hardphone),
   different communities that can be accessed (voice chat with other
   users that have the same voice chat client vs. voice communications
   with any endpoint on the Public Switched Telephone Network (PSTN)),
   or different applications that are invoked by the user (manually
   selecting a Push-To-Talk application from a wireless phone vs. a
   telephony application).  Outside of a call, the difference in user
   experience can be a billing one (cheaper for one service than
   another), a notification feature for one and not another (for
   example, an IM that gets sent whenever a user makes a call), and
   so on.

   In some cases, there is very little difference in the underlying
   technology that will support two different services, and in other
   cases, there are big differences.  However, for the purposes of this
   discussion, the key definition is that two services are distinct when
   there is a perceived difference by the user in the two services.

   This leads naturally to the desire to perform service identification.
   Service identification is defined as the process of:

   1.  determining the underlying service that is driving a particular
       signaling exchange,

   2.  associating that service with a service identifier, and

   3.  attaching that moniker to a signaling message (typically a SIP
       INVITE).

   Once service identification is performed, the service identifier can
   then be used for various purposes within the network.  Service
   identification can be done in the endpoints, in which case the UA
   would insert the moniker directly into the signaling message based on
   its awareness of the service.  Or, it can be done within a server in
   the network (such as a proxy), based on inspection of the SIP
   message, or based on hints placed into the message by the user.





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   When service identification is performed entirely by inspecting the
   signaling, this is called derived service identification.  When it is
   done based on knowledge possessed only by the invoking user agent, it
   is called declarative service identification.  Declarative service
   identification can only be done in user agents, by definition.

3.  Example Services

   It is very useful to consider several example services, especially
   ones that appear difficult to differentiate from each other.  In
   cases where it is hard to differentiate, service identification --
   and in particular, declarative service identification -- appears
   highly attractive (and indeed, required).

3.1.  IPTV vs. Multimedia

   IP Television (IPTV) is the usage of IP networks to access
   traditional television content, such as movies and shows.  SIP can be
   utilized to establish a session to a media server in a network, which
   then serves up multimedia content and streams it as an audio and
   video stream towards the client.  Whether SIP is ideal for IPTV is,
   in itself, a good question.  However, such a discussion is outside
   the scope of this document.

   Consider multimedia conferencing.  The user accesses a voice and
   video conference at a conference server.  The user might join in
   listen-only mode, in which case the user receives audio and video
   streams, but does not send.

   These two services -- IPTV and listen-only multimedia conferencing --
   clearly appear as different services.  They have different user
   experiences and applications.  A user is unlikely to ever be confused
   about whether a session is IPTV or listen-only multimedia
   conferencing.  Indeed, they are likely to have different software
   applications or endpoints for the two services.

   However, these two services look remarkably alike based on the
   signaling.  Both utilize audio and video.  Both could utilize the
   same codecs.  Both are unidirectional streams (from a server in the
   network to the client).  Thus, it would appear on the surface that
   there is no way to differentiate them, based on inspection of the
   signaling alone.









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3.2.  Gaming vs. Voice Chat

   Consider an interactive game, played between two users from their
   mobile devices.  The game involves the users sending each other game
   moves, using a messaging channel, in addition to voice.  In another
   service, users have a voice and IM chat conversation using a buddy-
   list application on their PC.

   In both services, there are two media streams -- audio and messaging.
   The audio uses the same codecs.  Both use the Message Session Relay
   Protocol (MSRP) [RFC4975].  In both cases, the caller would send an
   INVITE to the Address of Record (AOR) of the target user.  However,
   these represent fairly different services, in terms of user
   experience.

3.3.  Gaming vs. Voice Chat #2

   Consider a variation on the example in Section 3.2.  In this
   variation, two users are playing an interactive game between their
   phones.  However, the game itself is set up and controlled using a
   proprietary mechanism -- not using SIP at all.  However, the client
   application allows the user to chat with their opponent.  The chat
   session is a simple voice session set up between the players.

   Compare this with a basic telephone call between the two users.  Both
   involve a single audio session.  Both use the same codecs.  They
   appear to be identical.  However, different user experiences are
   needed.  For example, we desire traditional telephony features (such
   as call forwarding and call screening) to be applied in the telephone
   service, but not in the gaming chat service.

3.4.  Configuration vs. Pager Messaging

   The SIP MESSAGE method [RFC3428] provides a way to send one-shot
   messages to a particular AOR.  This specification is primarily aimed
   at Short Message Service (SMS)-style messaging, commonly found in
   wireless phones.  Receipt of a MESSAGE request would cause the
   messaging application on a phone to launch, allowing the user to
   browse the message history and respond.

   However, a MESSAGE request is sometimes used for the delivery of
   content to a device for other purposes.  For example, some providers
   use it to deliver configuration updates, such as new phone settings
   or parameters, or to indicate that a new version of firmware is
   available.  Though not designed for this purpose, the MESSAGE method
   gets used since, in existing wireless networks, SMS is used for this
   purpose, and the MESSAGE request is the SIP equivalent of SMS.




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   Consequently, the MESSAGE request sent to a phone can be for two
   different services.  One would require invocation of a messaging app,
   whereas the other would be consumed by the software in the phone,
   without any user interaction at all.

4.  Using Service Identification

   It is important to understand what the service identity would be
   utilized for, if known.  This section discusses the primary uses.
   These are application invocation in user agents and the network,
   Quality of Service authorization, service authorization, accounting
   and billing, service negotiation, and device dispatch.

4.1.  Application Invocation in the User Agent

   In some of the examples above, there were multiple software
   applications executing on the host.  One common way of achieving this
   is to utilize a common SIP user agent implementation that listens for
   requests on a single port.  When an incoming INVITE or MESSAGE
   arrives, it must be delivered to the appropriate application
   software.  When each service is bound to a distinct software
   application, it would seem that the service identity is needed to
   dispatch the message to the appropriate piece of software.  This is
   shown in Figure 1.

                    +---------------------------------+
                    |                                 |
                    | +-------------+ +-------------+ |
                    | |     UI      | |     UI      | |
                    | +-------------+ +-------------+ |
                    | +-------------+ +-------------+ |
                    | |             | |             | |
                    | |  Service 1  | |  Service 2  | |
                    | |             | |             | |
                    | +-------------+ +-------------+ |
                    | +-----------------------------+ |
                    | |                             | |
                    | |             SIP             | |
                    | |            Layer            | |
                    | |                             | |
                    | +-----------------------------+ |
                    |                                 |
                    +---------------------------------+

                             Physical Device

                                 Figure 1




RFC 5897                    Service ID in SIP                  June 2010


   The role of the SIP layer is to parse incoming messages, handle the
   SIP state machinery for transactions and dialogs, and then dispatch
   requests to the appropriate service.  This software architecture is
   analogous to the way web servers frequently work.  An HTTP server
   listens on port 80 for requests, and based on the HTTP Request-URI,
   dispatches the request to a number of disparate applications.  The
   same is happening here.  For the example services in Section 3.2, an
   incoming INVITE for the gaming service would be delivered to the
   gaming application software.  An incoming INVITE for the voice chat
   service would be delivered to the voice chat application software.
   The example in Section 3.3 is similar.  For the examples in
   Section 3.4, a MESSAGE request for user-to-user messaging would be
   delivered to the messaging or SMS app, and a MESSAGE request
   containing configuration data would be delivered to a configuration
   update application.

   Unlike the web, however, in all three use cases, the user initiating
   communications has (or appears to have -- more below) only a single
   identifier for the recipient -- their AOR.  Consequently, the SIP
   Request-URI cannot be used for dispatching, as it is identical in all
   three cases.

4.2.  Application Invocation in the Network

   Another usage of a service identifier would be to cause servers in
   the SIP network to provide additional processing, based on the
   service.  For example, an INVITE issued by a user agent for IPTV
   would pass through a server that does some kind of content rights
   management, authorizing whether the user is allowed to access that
   content.  On the other hand, an INVITE issued by a user for
   multimedia conferencing would pass through a server providing
   "traditional" telephony features, such as outbound call screening and
   call recording.  It would make no sense for the INVITE associated
   with IPTV to have outbound call screening and call recording applied,
   and it would make no sense for the multimedia conferencing INVITE to
   be processed by the content rights management server.  Indeed, in
   these cases, it's not just an efficiency issue (invoking servers when
   not needed), but rather, truly incorrect behavior can occur.  For
   example, if an outbound call screening application is set to block
   outbound calls to everything except for the phone numbers of friends
   and family, an IPTV request that gets processed by such a server
   would be blocked (as it's not targeted to the AOR of a friend or
   family member).  This would block a user's attempt to access IPTV
   services, when that was not the goal at all.

   Similarly, a MESSAGE request as described in Section 3.4 might need
   to pass through a message server for filtering when it is associated
   with chat, but not when it is associated with a configuration update.



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   Consider a filter that gets applied to MESSAGE requests, and that
   filter runs in a server in the network.  The filter operation
   prevents user Joe from sending messages to user Bob that contain the
   words "stock" or "purchase", due to some regulations that disallow
   Joe and Bob from discussing stock trading.  However, a MESSAGE for
   configuration purposes might contain an XML document that uses the
   token "stock" as some kind of attribute.  This configuration update
   would be discarded by the filtering server, when it should not have
   been.

4.3.  Network Quality-of-Service Authorization

   The IP network can provide differing levels of Quality of Service
   (QoS) to IP packets.  This service can include guaranteed throughput,
   latency, or loss characteristics.  Typically, the user agent will
   make some kind of QoS request, either using explicit signaling
   protocols (such as the Resource ReSerVation Protocol (RSVP)
   [RFC2205]) or through marking of a Diffserv value in packets.  The
   network will need to make a policy decision based on whether or not
   these QoS treatments are authorized.  One common authorization policy
   is to check if the user has invoked a service using SIP that they are
   authorized to invoke, and that this service requires the level of QoS
   treatment the user has requested.

   For example, consider IPTV and multimedia conferencing as described
   in Section 3.1.  IPTV is a non-real-time service.  Consequently,
   media traffic for IPTV would be authorized for bandwidth guarantees,
   but not for latency or loss guarantees.  On the other hand,
   multimedia conferencing is in real time.  Its traffic would require
   bandwidth, loss, and latency guarantees from the network.

   Consequently, if a user should make an RSVP reservation for a media
   stream, and ask for latency guarantees for that stream, the network
   would choose to be able to authorize it if the service was multimedia
   conferencing, but not if it was IPTV.  This would require the server
   performing the QoS authorization to know the service associated with
   the INVITE that set up the session.

4.4.  Service Authorization

   Frequently, a network administrator will want to authorize whether a
   user is allowed to invoke a particular service.  Not all users will
   be authorized to use all services that are provided.  For example, a
   user may not be authorized to access IPTV services, whereas they are
   authorized to utilize multimedia processing.  A user might not be
   able to utilize a multiplayer gaming service, whereas they are
   authorized to utilize voice chat services.




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   Consequently, when an INVITE arrives at a server in the network, the
   server will need to determine what the requested service is, so that
   the server can make an authorization decision.

4.5.  Accounting and Billing

   Service authorization and accounting/billing go hand in hand.  One of
   the primary reasons for authorizing that a user can utilize a service
   is that they are being billed differently based on the type of
   service.  Consequently, one of the goals of a service identity is to
   be able to include it in accounting records, so that the appropriate
   billing model can be applied.

   For example, in the case of IPTV, a service provider can bill based
   on the content (US $5 per movie, perhaps), whereas for multimedia
   conferencing, they can bill by the minute.  This requires the
   accounting streams to indicate which service was invoked for the
   particular session.

4.6.  Negotiation of Service

   In some cases, when the caller initiates a session, they don't
   actually know which service will be utilized.  Rather, they might
   choose to offer up all of the services they have available to the
   called party, and then let the called party decide, or let the system
   make a decision based on overlapping service capabilities.

   As an example, a user can do both the game and the voice chat service
   described in Section 3.2.  The user initiates a session to a target
   AOR, but the devices used by the target can only support voice chat.
   The called device returns, in its call acceptance, an indication that
   only voice chat can be used.  Consequently, voice chat gets utilized
   for the session.

4.7.  Dispatch to Devices

   When a user has multiple devices, each with varying capabilities in
   terms of service, it is useful to dispatch an incoming request to the
   right device based on whether the device can support the service that
   has been requested.

   For example, if a user initiates a gaming session with voice chat,
   and the target user has two devices -- one that can support the
   gaming service, and another that cannot -- the INVITE should be
   dispatched to the device that supports the gaming session.






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5.  Key Principles of Service Identification

   In this section, we describe several key principles of service
   identification:

   1.  Services are a by-product of signaling

   2.  Identical signaling produces identical services

   3.  Declarative service identification is an example of "Do What I
       Mean" (DWIM)

   4.  Declarative service identifiers are redundant

   5.  URIs are a key mechanism for producing differentiated signaling

5.1.  Services Are a By-Product of Signaling

   Declarative service identification -- the addition of a service
   identifier by clients in order to inform other entities of what the
   service is -- is a very compelling solution to solving the use cases
   described above.  It provides a clear way for each of the use cases
   to be differentiated.  On the other hand, derived service
   identification appears "hard", since the signaling appears to be the
   same for these different services.

   Declarative service identification misses a key point, which cannot
   be stressed enough, and which represents the core architectural
   principle to be understood here:

      A service is the byproduct of the signaling and the context around
      it (the user profile, time of day, and so on) -- the effects of
      the signaling message once it is launched into the network.  The
      service identity is therefore always derivable from the signaling
      and its context without additional identifiers.  In other words,
      derived service identification is always possible when signaling
      is being properly handled.

   When a user sends an INVITE request to the network and targets that
   request at an IPTV server, and includes the Session Description
   Protocol (SDP) for audio and video streaming, the *result* of sending
   such an INVITE is that an IPTV session occurs.  The entire purpose of
   the INVITE is to establish such a session, and therefore, invoke the
   service.  Thus, a service is not something that is different from the
   rest of the signaling message.  A service is what the user gets after
   the network and other user agents have processed a signaling message.





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   It may seem that delayed offers (SIP INVITE requests that lack SDP)
   make it impossible to perform derived service identification.  After
   all, in some of the cases above, the differentiation was done using
   the SDP in the request.  What if it's not there?  The answer is
   simple -- if it's not there, and the SDP is being offered by the
   called party, you cannot in fact know the service at the time of the
   INVITE.  That's the whole point of delayed offer -- to give the
   called party the chance to offer up what it wants for the session.
   In cases where service identification is needed at request time,
   delayed offer cannot be used.

5.2.  Identical Signaling Produces Identical Services

   This principle is a natural conclusion of the previous assertion.  If
   a service is the byproduct of signaling, how can a user have
   different experiences and different services when the signaling
   message is the same?  They cannot.

   But how can that be?  From the examples in Section 3, it would seem
   that there are services that are different, but have identical
   signaling.  If we hold true to the assertion, there is in fact only
   one logical conclusion:

      If two services are different, but their signaling appears to be
      the same, it is because one or more of the following is true:

      1.  there is in fact something different that has been overlooked

      2.  something has been implied from the signaling, when in fact it
          should have been signaled explicitly

      3.  the signaling mechanism should be changed so that there is, in
          fact, something that is different

   To illustrate this, let us take each of the example services in
   Section 3 and investigate whether there is, or should be, something
   different in the signaling in each case.

   IPTV vs. Multimedia Conferencing:  The two services described in
      Section 3.1 appear to have identical signaling.  They both involve
      audio and video streams, both of which are unidirectional.  Both
      might utilize the same codecs.  However, there is another
      important difference in the signaling -- the target URI.  In the
      case of IPTV, the request is targeted at a media server or to a
      particular piece of content to be viewed.  In the case of
      multimedia conferencing, the target is a conference server.  The
      administrator of the domain can therefore examine the Request-URI




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      and figure out whether it is targeted for a conference server or a
      content server, and use that to derive the service associated with
      the request.

   Gaming vs. Voice Chat:  Though both sessions involve MSRP and voice,
      and both are targeted to the same AOR of the called user, there is
      a difference.  The MSRP messages for the gaming session carry
      content that is game specific, whereas the MSRP messages for the
      voice chat are just regular text, meant for rendering to a user.
      Thus, the MSRP session in the SDP will indicate the specific
      content type that MSRP is carrying, and this type will differ in
      both cases.  Even if the game moves look like text, since they are
      being consumed by an automata, there is an underlying schema that
      dictates their content, and therefore, this schema represents the
      actual content type that should be signaled.

   Gaming vs. Voice Chat #2:  In this case, both sessions involve only
      voice, and both are targeted at the same AOR.  Indeed, there truly
      is nothing different -- if indeed the signaling works this way.
      However, there is an alternative mechanism for performing the
      signaling.  For the gaming session, the proprietary protocol can
      be used to exchange a URI that can be used to identify the voice
      chat function on the phone that is associated with the game (for
      example, a Globally Routable User Agent URI (GRUU) can be used
      [RFC5627]).  Indeed, the gaming chat is not targeting the USER --
      it's targeting the gaming instance on the phone.  Thus, if a
      special GRUU is used for the gaming chat, this makes the signaling
      different between these two services.

   Configuration vs. Pager Messaging:  Just as in the case of gaming vs.
      voice chat, the content type of the messages differentiates the
      service that occurs as a consequence of the messages.

5.3.  Do What I Say, Not What I Mean

   "Do What I Mean", abbreviated as DWIM, is a concept in computer
   science.  It is sometimes used to describe a function that tries to
   intelligently guess at what the user intended.  It is in contrast to
   "Do What I Say", or DWIS, which describes a function that behaves
   concretely based on the inputs provided.  Systems built on the DWIM
   concept can have unexpected behaviors, because they are driven by
   unstated rules.

   Declarative service identification is an example of DWIM.  The
   service identifier has no well-defined impact on the state machinery
   or protocols in the system; it has various side effects based on an
   assumption of what is meant by the service identifier.  Derived
   service identification, on the other hand, is an expression of the



RFC 5897                    Service ID in SIP                  June 2010


   principle of DWIS -- the behavior of the system is based entirely on
   the specifics of the protocol and are well defined by the protocol
   specification.  The service identifier is just a shorthand for
   summarizing things that are well defined by signaling.

   As a litmus test to differentiate the two cases, consider the
   following question.  If a request contained a service identifier, and
   that request were processed by a domain that didn't understand the
   concept of service identifiers at all, would the request be rejected
   if that service were not supported, or would it complete but do the
   wrong thing?  If it is the latter case, it's DWIM.  If it's the
   former, it's DWIS.

5.4.  Declarative Service Identifiers Are Redundant

   Because a declarative service identifier is, by definition, inside of
   the signaling message, and because the signaling itself completely
   defines the behavior of the service, another natural conclusion is
   that a declarative service identifier is redundant with the signaling
   itself.  It says nothing that could not or should not otherwise be
   derived from examination of the signaling.

5.5.  URIs Are Key for Differentiated Signaling

   In the IPTV example and in the second gaming example, it was
   ultimately the Request-URI that was (or should be) different between
   the two services.  This is important.  In many cases where services
   appear the same, it is because the resource that is being targeted is
   not, in fact, the user.  Rather, it is a resource that is linked with
   the user.  This resource might be an instance of a software
   application on the particular device of a user, or a resource in the
   network that acts on behalf of the user.

   The Request-URI is an infinitely large namespace for identifying
   these resources.  It is an ideal mechanism for providing
   differentiation when there would otherwise be none.

   Returning again to the example in Section 3.3, we can see that it
   does make more sense to target the gaming chat session at a software
   instance on the user's phone, rather than at the user themselves.
   The gaming chat session should really only go to the phone on which
   the user is playing the game.  The software instance does indeed live
   only on that phone, whereas the user themselves can be contacted in
   many ways.  We don't want telephony features invoked for the gaming
   chat session, because those features only make sense when someone is
   trying to communicate with the USER.  When someone is trying to





RFC 5897                    Service ID in SIP                  June 2010


   communicate with a software instance that acts on behalf of the user,
   a different set of rules apply, since the target of the request is
   completely different.

6.  Perils of Declarative Service Identification

   Based on these principles, several perils of declarative service
   identification can be described.  They are:

   1.  Declarative service identification can be used for fraud

   2.  Declarative service identification can hurt interoperability

   3.  Declarative service identification can stifle service innovation

6.1.  Fraud

   Declarative service identification can lead to fraud.  If a provider
   uses the service identifier for billing and accounting purposes, or
   for authorization purposes, it opens an avenue for attack.  The user
   can construct the signaling message so that its actual effect (which
   is the service the user will receive), is what the user desires, but
   the user places a service identifier into the request (which is what
   is used for billing and authorization) that identifies a cheaper
   service, or one that the user is not authorized to receive.  In such
   a case, the user will receive service, and not be billed properly for
   it.

   If, however, the domain administrator derived the service identifier
   from the signaling itself (derived service identification), the user
   cannot lie.  If they did lie, they wouldn't get the desired service.

   Consider the example of IPTV vs. multimedia conferencing.  If
   multimedia conferencing is cheaper, the user could send an INVITE for
   an IPTV session, but include a service identifier that indicates
   multimedia conferencing.  The user gets the service associated with
   IPTV, but at the cost of multimedia conferencing.

   This same principle shows up in other places -- for example, in the
   identification of an emergency services call [ECRIT-FRAMEWORK].  It
   is desirable to give emergency services calls special treatment, such
   as being free and authorized even when the user cannot otherwise make
   calls, and to give them priority.  If emergency calls were indicated
   through something other than the target of the call being an
   emergency services URN [RFC5031], it would open an avenue for fraud.
   The user could place any desired URI in the request-URI, and indicate
   separately, through a declarative identifier, that the call is an
   emergency services call.  This would then get special treatment but



RFC 5897                    Service ID in SIP                  June 2010


   of course would get routed to the target URI.  The only way to
   prevent this fraud is to consider an emergency call as any call whose
   target is an emergency services URN.  Thus, the service
   identification here is based on the target of the request.  When the
   target is an emergency services URN, the request can get special
   treatment.  The user cannot lie, since there is no way to separately
   indicate that this is an emergency call, besides targeting it to an
   emergency URN.

6.2.  Systematic Interoperability Failures

   How can declarative service identification cause loss of
   interoperability?  When an identifier is used to drive functionality
   -- such as dispatch on the phones, in the network, or QoS
   authorization -- it means that the wrong thing can happen when this
   field is not set properly.  Consider a user in domain 1, calling a
   user in domain 2.  Domain 1 provides the user with a service they
   call "voice chat", which utilizes voice and IM for real-time
   conversation, driven off of a buddy-list application on a PC.
   Domain 2 provides their users with a service they call "text
   telephony", which is a voice service on a wireless device that also
   allows the user to send text messages.  Consider the case where
   domain 1 and domain 2 both have their user agents insert a service
   identifier into the request, and then use that to perform QoS
   authorization, accounting, and invocation of applications in the
   network and in the device.  The user in domain 1 calls the user in
   domain 2, and inserts the identifier "Voice Chat" into the INVITE.
   When this arrives at the server in domain 2, the service identifier
   is unknown.  Consequently, the request does not get the proper QoS
   treatment, even if the call itself will succeed.

   If, on the other hand, derived service identification were used, the
   service identifier could be removed by domain 2, and then recomputed
   based on the signaling to match its own notion of services.  In this
   case, domain 2 could derive the "text telephony" identifier, and the
   request completes successfully.

   Declarative service identification, used between domains, causes
   interoperability failures unless all interconnected domains agree on
   exactly the same set of services and how to name them.  Of course,
   lack of service identifiers does not guarantee service
   interoperability.  However, SIP was built with rich tools for
   negotiation of capabilities at a finely granular level.  One user
   agent can make a call using audio and video, but if the receiving UA
   only supports audio, SIP allows both sides to negotiate down to the
   lowest common denominator.  Thus, communication is still provided.
   As another example, if one agent initiates a Push-To-Talk session
   (which is audio with a companion floor control mechanism), and the



RFC 5897                    Service ID in SIP                  June 2010


   other side only did regular audio, SIP would be able to negotiate
   back down to a regular voice call.  As another example, if a calling
   user agent is running a high-definition video conferencing endpoint,
   and the called user agent supports just a regular video endpoint, the
   codecs themselves can negotiate downward to a lower rate, picture
   size, and so on.  Thus, interoperability is achieved.  Interestingly,
   the final "service" may no longer be well characterized by the
   service identifier that would have been placed in the original
   INVITE.  For example, in this case, if the original INVITE from the
   caller had contained the service identifier "hi-fi video", but the
   video gets negotiated down to a lower rate and picture size, the
   service identifier is no longer really appropriate.  That is why
   services need to be derived by signaling -- because the signaling
   itself provides negotiation and interoperability between different
   domains.

   This illustrates another key aspect of the interoperability problem.
   Declarative service identification will result in inconsistencies
   between its service identifiers and the results of any SIP
   negotiation that might otherwise be applied in the session.

   When a service identifier becomes something that both proxies and the
   user agent need to understand in order to properly treat a request
   (which is the case for declarative service identification), it
   becomes equivalent to including a token in the Proxy-Require and
   Require header fields of every single SIP request.  The very reason
   that [RFC4485] frowns upon usage of Require and certainly Proxy-
   Require is the huge impact on interoperability it causes.  It is for
   this same reason that declarative service identification needs to be
   avoided.

6.3.  Stifling of Service Innovation

   The probability that any two service providers end up with the same
   set of services, and give those services the same names, becomes
   smaller and smaller as the number of providers grow.  Indeed, it
   would almost certainly require a centralized authority to identify
   what the services are, how they work, and what they are named.  This,
   in turn, leads to a requirement for complete homogeneity in order to
   facilitate interconnection.  Two providers cannot usefully
   interconnect unless they agree on the set of services they are
   offering to their customers and each do the same thing.  This is
   because each provider has become dependent on inclusion of the proper
   service identifier in the request, in order for the overall treatment
   of the request to proceed correctly.  This is, in a very real sense,
   anathema to the entire notion of SIP, which is built on the idea that
   heterogeneous domains can interconnect and still get
   interoperability.



RFC 5897                    Service ID in SIP                  June 2010


   Declarative service identification leads to a requirement for
   homogeneity in service definitions across providers that
   interconnect, ruining the very service heterogeneity that SIP was
   meant to bring.

   Indeed, Metcalfe's Law says that the value of a network grows with
   the square of the number of participants.  As a consequence of this,
   once a bunch of large domains did get together, agree on a set of
   services, and then agree on a set of well-known identifiers for those
   services, it would force other providers to also deploy the same
   services, in order to obtain the value that interconnection brings.
   This, in turn, will stifle innovation, and quickly force the set of
   services in SIP to become fixed and never expand beyond the ones
   initially agreed upon.  This, too, is anathema to the very framework
   on which SIP is built, and defeats much of the purpose of why
   providers have chosen to deploy SIP in their own networks.

   Consider the following example.  Several providers get together and
   standardize on a bunch of service identifiers.  One of these uses
   audio and video (say, "multimedia conversation").  This service is
   successful and is widely utilized.  Endpoints look for this
   identifier to dispatch calls to the right software applications, and
   the network looks for it to invoke features, perform accounting, and
   provide QoS.  A new provider gets the idea for a new service (say,
   "avatar-enhanced multimedia conversation").  In this service, there
   is audio and video, but there is a third stream, which renders an
   avatar.  A caller can press buttons on their phone, to cause the
   avatar on the other person's device to show emotion, make noise, and
   so on.  This is similar to the way emoticons are used today in IM.
   This service is enabled by adding a third media stream (and
   consequently, a third m-line) to the SDP.

   Normally, this service would be backwards-compatible with a regular
   audio-video endpoint, which would just reject the third media stream.
   However, because a large network has been deployed that is expecting
   to see the token, "multimedia conversation" and its associated audio+
   video service, it is nearly impossible for the new provider to roll
   out this new service.  If they did, it would fail completely, or
   partially fail, when their users call users in other provider
   domains.











RFC 5897                    Service ID in SIP                  June 2010


7.  Recommendations

   From these principles, several recommendations can be made.

7.1.  Use Derived Service Identification

   Derived service identification -- where an identifier for a service
   is obtained by inspection of the signaling and of other contextual
   data (such as subscriber profile) -- is reasonable, and when done
   properly, does not lead to the perils described above.  However,
   declarative service identification -- where user agents indicate what
   the service is, separate from the rest of the signaling -- leads to
   the perils described above.

   If it appears that the signaling currently defined in standards is
   not sufficient to identify the service, it may be due to lack of
   sufficient signaling to convey what is needed, or may be because
   request URIs should be used for differentiation and they are not
   being used.  By applying the litmus tests described in Section 5.3,
   network designers can determine whether or not the system is
   attempting to perform declarative service identification.

7.2.  Design for SIP's Negotiative Expressiveness

   One of SIP's key strengths is its ability to negotiate a common view
   of a session between participants.  This means that the service that
   is ultimately received can vary wildly, depending on the types of
   endpoints in the call and their capabilities.  Indeed, this fact
   becomes even more evident when calls are set up between domains.

   As such, when performing derived service identification, domains
   should be aware that sessions may arrive from different networks and
   different endpoints.  Consequently, the service identification
   algorithm must be complete -- meaning it computes the best answer for
   any possible signaling message that might be received and any session
   that might be set up.

   In a homogeneous environment, the process of service identification
   is easy.  The service provider will know the set of services they are
   providing, and based on the specific call flows for each specific
   service, can construct rules to differentiate one service from
   another.  However, when different providers interconnect, or when
   different endpoints are introduced, assumptions about what services
   are used, and how they are signaled, no longer apply.  To provide the
   best user experience possible, a provider doing service
   identification needs to perform a "best-match" operation, such that





RFC 5897                    Service ID in SIP                  June 2010


   any legal SIP signaling -- not just the specific call flows running
   within their own network amongst a limited set of endpoints -- is
   mapped to the appropriate service.

7.3.  Presence

   Presence can help a great deal with providing unique URIs for
   different services.  When a user wishes to contact another user, and
   knows only the AOR for the target (which is usually the case), the
   user can fetch the presence document for the target.  That document,
   in turn, can contain numerous service URIs for contacting the target
   with different services.  Those URIs can then be used in the Request-
   URI for differentiation.  When possible, this is the best solution to
   the problem.

7.4.  Intra-Domain

   Service identifiers themselves are not bad; derived service
   identification allows each domain to cache the results of the service
   identification process for usage by another network element within
   the same domain.  However, service identifiers are fundamentally
   useful within a particular domain, and any such header must be
   stripped at a network boundary.  Consequently, the process of service
   identification and their associated service identifiers is always an
   intra-domain operation.

7.5.  Device Dispatch

   Device dispatch should be done following the principles of [RFC3841],
   using implicit preferences based on the signaling.  For example,
   [RFC5688] defines a new UA capability that can be used to dispatch
   requests based on different types of application media streams.

   However, it is a mistake to try and use a service identifier as a UA
   capability.  Consider a service called "multimedia telephony", which
   adds video to the existing PSTN experience.  A user has two devices,
   one of which is used for multimedia telephony and the other strictly
   for a voice-assisted game.  It is tempting to have the telephony
   device include a UA capability [RFC3840] called "multimedia
   telephony" in its registration.  A calling multimedia telephony
   device can then include the Accept-Contact header field [RFC3841]
   containing this feature tag.  The proxy serving the called party,
   applying the basic algorithms of [RFC3841], will correctly route the
   call to the terminating device.

   However, if the calling party is not within the same domain, and the
   calling domain does not know about or use this feature tag, there
   will be no Accept-Contact header field, even if the calling party was



RFC 5897                    Service ID in SIP                  June 2010


   using a service that is a good match for "multimedia telephony".  In
   such a case, the call may be delivered to both devices, but it will
   yield a poorer user experience.  That's because device dispatch was
   done using declarative service identification.

   The best way to avoid this problem is to use feature tags that can be
   matched to well-defined signaling features -- media types, required
   SIP extensions, and so on.  In particular, the golden rule is that
   the granularity of feature tags must be equivalent to the granularity
   of individual features that can be signaled in SIP.

8.  Security Considerations

   Oftentimes, the service associated with a request is utilized for
   purposes such as authorization, accounting, and billing.  When
   service identification is not done properly, the possibility of
   unauthorized service use and network fraud is introduced.  It is for
   this reason, discussed extensively in Section 6.1, that the usage of
   declarative service identifiers inserted by a UA is not recommended.

9.  Acknowledgements

   This document is based on discussions with Paul Kyzivat and
   Andrew Allen, who contributed significantly to the ideas here.  Much
   of the content in this document is a result of discussions amongst
   participants in the SIPPING mailing list, including Dean Willis,
   Tom Taylor, Eric Burger, Dale Worley, Christer Holmberg, and
   John Elwell, amongst many others.  Thanks to Spencer Dawkins,
   Tolga Asveren, Mahesh Anjanappa, and Claudio Allochio for reviews of
   this document.

10.  Informative References

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

   [RFC4479]  Rosenberg, J., "A Data Model for Presence", RFC 4479,
              July 2006.

   [RFC4485]  Rosenberg, J. and H. Schulzrinne, "Guidelines for Authors
              of Extensions to the Session Initiation Protocol (SIP)",
              RFC 4485, May 2006.

   [RFC4975]  Campbell, B., Mahy, R., and C. Jennings, "The Message
              Session Relay Protocol (MSRP)", RFC 4975, September 2007.




RFC 5897                    Service ID in SIP                  June 2010


   [RFC5031]  Schulzrinne, H., "A Uniform Resource Name (URN) for
              Emergency and Other Well-Known Services", RFC 5031,
              January 2008.

   [ECRIT-FRAMEWORK]
              Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
              "Framework for Emergency Calling using Internet
              Multimedia", Work in Progress, July 2009.

   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent URIs (GRUUs) in the Session Initiation Protocol
              (SIP)", RFC 5627, October 2009.

   [RFC5688]  Rosenberg, J., "A Session Initiation Protocol (SIP) Media
              Feature Tag for MIME Application Subtypes", RFC 5688,
              January 2010.

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

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

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

   [RFC2205]  Braden, B., Zhang, L., Berson, S., Herzog, S., and S.
              Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1
              Functional Specification", RFC 2205, September 1997.

Author's Address

   Jonathan Rosenberg
   jdrosen.net
   Monmouth, NJ
   USA

   EMail: jdrosen@jdrosen.net
   URI:   http://www.jdrosen.net