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In: IS-auth-wait
Network Working Group                                       J. Haluska
Internet Draft                                            R. Berkowitz
Expires: August 2007                                          P. Roder
                                                             W. Downum
                                           Telcordia Technologies, Inc.
                                                              R. Ahern
                                     AT&T Customer Information Services
                                                            P. Lum Lung
                                     Qwest Communications International
                                                          N. Costantino
                                             Soleo Communications, Inc.
                                                           C. Blackwell
                                                           J. Mellinger
                                                                Verizon
                                                               D. Scott
                                                              VoltDelta
                                                      February 15, 2007



   Considerations for Information Services and Operator Services Using
                                   SIP


            draft-haluska-sipping-directory-assistance-02.txt




Status of this Memo

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



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

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

   This Internet-Draft will expire on August 15, 2007.

Abstract

   Information Services are services whereby information is provided
   in response to user requests, and may include involvement of a
   human or automated agent. A popular existing Information Service is
   Directory Assistance (DA). Moving ahead, Information Services
   providers envision exciting multimedia services that support
   simultaneous voice and data interactions with full operator backup
   at any time during the call. Information Services providers are
   planning to migrate to SIP based platforms, which will enable such
   advanced services, while continuing to support traditional DA
   services. This document aims to identify how such services can be
   implemented using existing or currently proposed SIP mechanisms,
   and to provide a set of Best Current Practices to facilitate
   interoperability.



Table of Contents


   1. Introduction..................................................3
   2. Terminology...................................................4
   3. High Level Requirements.......................................7
   4. Service Description..........................................11
   5. OISP Internal Architecture...................................15
   6. General Approach.............................................16
   7. Signaling Mechanisms.........................................18
      7.1. Calling Party's Identity................................18
      7.2. Provider Identification.................................20
         7.2.1. Home Provider......................................20
         7.2.2. Last Hop Provider..................................21
         7.2.3. Arbitrary Traversed Provider.......................22
      7.3. Originating Station Type................................24
      7.4. Trunk Group Identifier..................................25
      7.5. Dialed Digits...........................................26
      7.6. Retargeting to Identify the Desired Service.............27
      7.7. Charge Number...........................................27
      7.8. Passing Whisper.........................................28


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      7.9. Calling Equipment Capabilities and Characteristics......31
      7.10. Media Server Returning Data to the Application Server..32
      7.11. Service Discovery......................................32
   8. Call Flow....................................................33
   9. VoIP Information Services - Summary and Next Steps...........42
   10. Security Considerations.....................................42
   11. IANA Considerations.........................................43
   12. References..................................................44
      12.1. Normative References...................................44
      12.2. Informative References.................................44
   Author's Addresses..............................................46



1. Introduction

   Information Services are services whereby information is provided
   in response to user requests. This may include involvement of a
   human or automated agent. Information Services may include call
   completion to a requested telephone number and other extensions
   provided on behalf of the owner of the information, such as
   assistance with purchases. The users normally access the
   Information Services by dialing a Directory Assistance (DA) dialing
   sequence and verbally requesting an operator or automated system
   for the information. The users may also request information through
   other access methods, such as chat (IM), email, Web (HTTP) or SMS
   initiated requests. The Information may be delivered to the user
   via any mode, such as verbal announcements, chat (IM), email, Web
   (HTTP), MMS, or SMS.

   A popular existing Information Service is Directory Assistance
   (DA).  DA is a well known service in today's PSTN, and is generally
   identified with "411" or "NPA-555-1212" type services in North
   America.  Today's DA services provide a user with telephone number
   associated with a name and locality provided by the user, can
   complete the call for the user, and can send SMS with the listing
   to the user's wireless phone. Other Information Services provide
   the user with a wide range of information, such as movie listings
   and the weather.

   Moving ahead, Information Services providers envision exciting
   multimedia services that support simultaneous voice and data
   interactions with full operator backup at any time during the call.
   For instance, a directions Information Service may announce and
   display directions to the requested listing, with the option for
   the caller to request transfer to an operator with the latest call
   context information.


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   Information Services providers are planning to migrate to SIP based
   platforms, which will enable such advanced services, while
   continuing to support traditional DA services.

   Implementing Information Services with SIP will require the
   exchange of certain information which is not normally exchanged for
   other types of calls. This document aims to identify such
   information, and stimulate discussion about how this information
   could be exchanged.  Existing mechanisms will be used where
   appropriate, and currently existing proposals would be favored over
   new extensions. It is intended to provide a Best Current Practices
   document to facilitate interoperability.

   In the current North American PSTN, Operator Services use signaling
   similar to current Directory Assistance services. Thus mechanisms
   developed for Information Services, which include Directory
   Assistance services, are expected to be useful in implementing
   Operator Services as well. This document aims to identify how such
   services can be implemented using existing or currently proposed
   SIP mechanisms, and to provide a set of Best Current Practices to
   facilitate interoperability.





2. Terminology

   This section defines terms that will be used to discuss Information
   Services.

   Application Server (AS) - An Application Server is a server
   providing value added services. It may influence and impact SIP
   sessions on behalf of the services supported by the service
   provider's network.

   Back End Automation - Back End Automation refers to automation of
   the function that provides listing information to the caller. This
   includes playing a verbal announcement with the listing
   information, and may also include prompting the user for call
   completion service.

   Branding - Branding is a service where customized announcements are
   provided to the caller to identify the service provider. For
   example, if the service is provided to a Home Provider's
   subscribers by a third party provider, branded service might
   include a message thanking them for using that Home Provider. Thus


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   the user experience is that the service is provided by their Home
   Provider rather than some third party. Branding can be influenced
   by a number of factors, including but not limited to the identity
   of the caller's Home Provider, or of other providers involved in
   the call.

   Call Completion - Call Completion is a service where a call is
   initiated by the provider on behalf of the user. For example, in
   the DA service, once the DA provider has identified the requested
   listing, it may offer to complete the call for the caller, usually
   for some additional fee. This relieves the user from having to
   remember the number and then dial it.

   DA Provider -The DA provider is the provider of DA services to end
   users. Since DA services are a subset of IS services, a DA provider
   is also an IS provider, and the definition of IS provider holds
   true for DA provider, except that the scope of services is limited
   to DA services.

   Front End Automation - Front End Automation refers to automation of
   the initial customer contact, whereby a branded announcement may be
   played, a prompt is played to the user, and the user's spoken
   request is recorded. Speech recognition and querying for the
   listing information are performed as part of front end automation.

   Home Provider - The service provider who is responsible for
   providing voice services to the calling customer. This is the
   service provider that has the business relationship with the
   calling customer. The identity of the home provider influences call
   processing treatment, such as branding and operator queue
   selection.

   HSS - Home Subscriber Server. The Home Subscriber Server is an IMS
   network element similar to a Home Location Register. It is a
   database containing information about the subscriber, user
   equipment, filter criteria for call processing triggers, etc.

   Information Services (IS) Provider - The IS provider is the
   provider of Information Services to end users. The Information
   Services provider provides retail services directly to end users,
   and provides wholesale services to other service providers.

   ISC - IP multimedia Subsystem Service Control Interface. AThe IP
   multimedia Subsystem Service Control Interface is an interface
   point in the IMS architecture between an S-CSCF and a service
   platform such as a SIP AS or an OSA SCS. SIP is the protocol used
   over this interface.


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   Last Hop Provider - In this document, the term "last hop provider"
   refers to the provider which passes session requests directly to
   the OISP. It may be the same as the caller's Home Provider, or it
   may be some other provider. "Last hop" is with respect to the OISP
   - it is the last provider the request traverses before arriving at
   the OISP. This term implies only a topological relationship.

   Layer 3 connectivity - This refers to IP connectivity, for example
   as provided by an Internet Service Provider or Managed IP service
   provider. If one entity has Layer 3 connectivity to another entity,
   then it can route packets to that entity. This does not imply
   anything about any physical path between the entities. Nor does it
   imply any application layer connectivity between the entities.

   Media Server: A Media Server is a general-purpose platform for
   executing real-time media processing tasks. Examples of typical
   functions performed by media servers include playing announcements,
   collecting speech and/or DTMF digits, and performing conferencing
   functions.

   OISP - Operator and Information Services Provider (OISP) - In this
   document, this term refers to an Information Services Provider,
   Directory Assistance Provider, or Operator Services Provider,
   depending on the context. This term is used for brevity. We are
   also defining this to be an adjective, thus "OISP services" is a
   convenient, intuitive way to say "Operator and Information
   Services".

   Retail DA service - A retail DA service is a DA service that is
   provided to a user by the user's Home Provider.

   SIP Layer connectivity - When one entity has SIP level connectivity
   to another entity, this implies that the second entity will accept,
   process, and route SIP requests from the first entity. This would
   usually involve business agreements as well.

   Time Division Multiplexed (TDM) Local Exchange Carriers (LECs) -
   ATDM LEC provides local exchange service to end users utilizing
   TDM-based switching systems.

   Transit Provider - A provider acts as a Transit Provider when it
   facilitates session setup between other providers. It can perform
   this role irrespective of whether it hosts subscribers.

   Transport Services Provider (TSP) - The TSP provides access at
   Layer 3 or below to other providers. The most obvious case of TSP
   is that of an internet service provider or managed IP services


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   provider. VSPs, and IS or DA providers may or may not share the
   same TSP for access to each other. Further, each of these providers
   may have multiple TSPs. In this case, the decision as to which is
   used is determined by the policy of the entity sending the traffic;
   the Border Gateway Protocol (BGP) is often used. It is entirely
   possible that the traffic from each entity towards the other takes
   separate paths; i.e. it should not be assumed that the incoming and
   outgoing paths are symmetric.

   Though the TSP is transparent at the application layer, knowledge
   of its identity may play a role in influencing the service logic
   because in some cases the incoming facility can be used to identify
   the provider, for instance in cases where there is only one
   provider connected via that TSP.

   Voice Services Provider (VSP) - An entity that provides transport
   of SIP signaling to its customers. It may also provide media
   streams to its customers. Such a service provider may additionally
   be interconnected with other service providers; that is, it may
   "peer" with other service providers. A VSP may also interconnect
   with the PSTN.

   Whisper - During front end automation, the OIS-MS will record and
   may time compress the caller's perhaps meandering speech into what
   is known as the "whisper". This is intended to be played into a
   human operator's ear, should the call be referred to an operator,
   to avoid the operator from having to prompt the caller again. The
   whisper is obtained during the front end automation, and saved as
   an audio file.

   Wholesale DA service -A wholesale DA service is a DA service that
   is provided to a user by a Service Provider other than the user's
   Home Provider.





3. High Level Requirements



   In addition to all-IP scenarios, it must be possible to support
   interworking with existing PSTN and wireless based providers, via
   both SS7 and MF interconnections.




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   It must be possible to support accounting. This includes both
   online and offline accounting. It must be possible to perform
   accounting for all actions associated with a particular call, and
   further to be able to correlate actions across multiple provider
   elements and across providers.

   It must be possible to support multiple Operator and Information
   Services Providers (OISPs) per originating provider. The choice as
   to which OISP to be used could be on a per subscriber basis, or on
   other criteria.

   It must be possible to support multiple OISP providers per call.
   For example, one provider might be used for front end automation,
   and another used for operator assistance.

   It must be possible to provide an automated announcement to the
   user, and prompt the user for the type of query and query
   information.

   It must be possible to pass a "whisper" to the operator
   workstation.

   It must be possible to connect the user to a human operator.

   It must be possible to provide an automated announcement of the
   requested information.

   It must be possible to prompt the user for call completion.

   It must be possible to perform call completion.



   It must be possible to support the case where OIS services are
   provided by the caller's Home Provider. This scenario is known in
   the OIS industry as the Retail scenario. In this case, the caller's
   Home Provider is also an OISP, and provides OIS service to its own
   subscribers. This is illustrated in the following figure:

   +--------+    +--------------------+
   | Caller |----| Home      +------+ |
   |        |    | Provider  | OISP | |
   |        |    |           +------+ |
   +--------+    +--------------------+

               Figure 1 Services Provider by Home Provider



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   It must be possible to support the case where OIS services are
   provided by a direct third party provider. In this scenario, the
   OISP is a third party service provider, and there is direct SIP
   layer connectivity as well as business relationships between the
   calling user's provider and the OISP. This is illustrated in the
   following figure:

   +--------+    +----------+   +------+
   | Caller |----| Home     |---| OISP |
   |        |    | Provider |   |      |
   +--------+    +----------+   +------|


       Figure 2 Services Provider by a Direct Third Party Provider





   It must be possible to support the case where services are provided
   by an indirect third party provider. In this scenario, the OISP is
   a third party provider, but the caller's Home Provider does not
   have direct SIP connectivity to the OISP. Further, it's possible
   that it has no business relationship with the OISP. The caller's
   provider routes the call to a provider with whom it does have a
   relationship, and this provider in turn routes either to the OISP,
   with which it has a relationship, or there could be multiple
   intermediate networks. This is seen when providers have membership
   in multiple, potentially overlapping sets of "peering clubs" or
   "federations" - when a provider does not have a peering with the
   desired provider, some other provider with which it does have a
   peering might be able to get the call to the destination provider.
   Another example would be a VOIP aggregator, for example providing
   terminations for multiple providers, and might also provider
   services such as DA for those providers as well. This is
   illustrated in the following figure:









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   +--------+    +--------+   +---------+   +------+
   | Caller |    |Home    |   | Inter-  |   | OISP |
   |        |----|Provider|---| mediate |---|      |
   |        |    |        |   | Provider|   |      |
   |        |    |  (A)   |   |   (B)   |   |  (C) |
   +--------+    +--------+   +---------+   +------+

      Figure 3 Services Provided by an Indirect Third Party Provider



   The following are potential future requirements.

   Operation via the general internet, not specific to any particular
   SDO's architecture, and not depending on any protocol extensions
   specific to those architectures, should be supported.

   In addition to the basic DA functionality, the architecture will
   need to support additional technical capabilities. These
   capabilities are currently under investigation. The following are
   some business needs which drive these capabilities.

   It must be possible to support multiple Information Services
   providers per originating provider. For instance, a Home Provider
   must be able to select an appropriate Information Services provider
   from among several Information Services providers based on criteria
   including but not limited to the identity of the calling
   subscriber.  It must also be possible to support multiple
   Information Services providers per call. For example, once the
   initial request has been satisfied, the user may make another
   Information Services request without hanging up, and it must be
   possible in this case to select the appropriate Information
   Services provider for the next request. In such cases the
   Information Services provider may be involved in selecting a
   different Information Services provider.

   It must be possible to support non voice initiated Information
   Services requests. Possible examples include chat (IM), email, Web
   (HTTP) or SMS initiated requests. In the case that the subscriber
   makes a purchase via some online auction service, that subscriber
   can via IM or email request the assistance of an operator.

   It must be possible to support both Information Services as well as
   Operator Services. Examples of Operator services include Operator
   Intercept, Busy Line Verification, Call Restrictions, etc.




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   It must be possible to support Purchase services and Concierge
   services. Such services facilitate the Information Services
   operator providing assistance to the caller after the listing has
   been announced and perhaps call completion performed. The call is
   routed to an Information Services operator who interacts with the
   customer, offering to help make a purchase. Concierge service is
   similar; the Information Services operator offers to make e.g.
   restaurant reservations for the caller.

   It must be possible to provide an application interface to other
   types of systems. An example could be a web based API, so that once
   some online search engine has located some business listing, the
   services of the Information Services provider could be invoked by
   the user from the web page.

   It must be possible to support IPTV interactive services. As
   multiple services such as IM and telephony are integrated with
   IPTV, it must be possible to initiate Information Services requests
   in this context as well.



4. Service Description

   Information Services (IS) are services whereby information is
   provided in response to user requests. This may include involvement
   of a human or automated agent. Usually, the user accesses the
   Information Service by placing a voice call to the automated
   Information Service and verbally requests the information, such as
   phone number, movie listings, weather, etc. Frequently, a live
   operator is attached to recognize the user's verbal request.
   Sometimes, the user can utilize other access methods, such as SMS,
   IM, or HTTP-initiated requests. These additional methods are not
   currently covered in this document.

   Information Services are often provided on a wholesale basis to
   Home Providers, and include features such as branded announcements.

   Directory Assistance (DA) is a specific type of Information Service
   whereby end users request a telephone number for an entity.



   The following represents a list of representative steps taken
   during the course of a typical DA request.




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   1. Initial recognition of an OIS call. At some point, the call
   needs to be identified as an OIS call, and appropriate routing or
   other logic must be invoked in order to fulfill the request. This
   could be based on any number of criteria, including but not limited
   to analysis of the "address information" - e.g. the digits or
   Request-URI emitted by the caller's UA. This could occur at any
   number of places - e.g. in the caller's UA, in a proxy in the home
   provider, or in some downstream element.

   2. Identification of the requested service. There are many possible
   OIS services, and the number of these is only expected to increase
   as providers respond to customer needs. At some point during call
   processing it is necessary to identify exactly which service is
   desired. For example "directory assistance with call completion" is
   a service where after the listing information is provided to the
   caller, the option is provided for the call to be placed
   automatically, so the customer need not hang up, remember the
   digits, and dial the number. Another example is "directory
   assistance only", where call completion is not offered. There are
   multiple factors which could affect the service which is to be
   offered, and the logic deciding this could be located anywhere
   along the path to the OIS provider. Some of the information
   required to make such decisions could include:

     o The digits dialed by the caller.

     o The Request-URI emitted by the caller's UA.

     o The identity of the calling party, for instance the calling
        party number.

     o The charge number associated with the caller's account.

     o The identity of the calling party's home provider.

     o The identity of the provider which directly hands off the call
        to the OISP.

     o The identity of other provider which the request might traverse

     o The Originating Station Type, in case the call was originated
        in the PSTN.

     o Trunk group information, in case the call was originated in the
        PSTN.




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     o Capabilities and characteristics of the caller's user
        equipment.



   3. Routing of the OIS call. The call must be routed towards an
   entity which can provide the requested service. Each entity or
   network handling the call routes it based on the logic located in
   that provider, and the information currently available. For
   instance, the home provider may know very little about OIS
   services, having farmed that service out to another provider.
   Consequently it might simply route all such calls towards the OIS
   provider, which decides which service is to be offered.

   4. Authentication. When one provider passes a call to another
   provider, there is a need for the providers involved to be sure of
   each other's identity. This might be through explicit security
   mechanisms such as mutual TLS or security gateways using IPSec
   tunnel mode, it might be through reliance on a closed set of
   trusted interconnected providers, or some other policy set by the
   providers involved.

   5. Receipt of the OIS call. The OIS provider needs to be able to
   receive such calls.

   6. Querying upstream providers for information. The OISP, or an
   intermediate provider may require information from an upstream
   provider. For instance, the capabilities and characteristics of the
   caller's equipment may be needed in order to influence call
   processing.

   7. Connection of caller to automated voice platform. The OISP must
   be able to connect the caller to an appropriate automated voice
   platform.

   8. Provision of branded announcements. The OISP must be capable of
   providing custom announcements to the caller based on a number of
   criteria. For example, it might greet the caller, thanking them for
   using their Home Provider's service. Though the service is actually
   provided by the OISP, business arrangements would dictate such
   branded announcements.

   9. Query the caller. The OISP must be capable of playing a voice
   request to the customer asking them for the listing. For example
   "Name and city, please."




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   10. Recording a spoken request. The OISP must be capable of
   recording the caller's spoken request. This both for speech
   recognition, and also for playing back the "whisper" to a human
   operator should one be required, to prevent having to ask the
   customer to repeat the request.

   11. Speech recognition. The OISP must be able to pass the caller's
   spoken request to speech recognition system, suitable for querying
   a listing database.

   12. Listing database query. The OISP must be capable of querying
   one or more listings databases using the request.

   13. Decide to use human operator if listing query fails. If the
   listing query fails, or the speech recognition fails, the OISP must
   be able to decide to send the call to a human operator.

   14. Selection of appropriate operator. When it has been determined
   that the call must be routed to a human operator, there are a
   number of factors to be taken into account to determine the
   appropriate operator for the call. It must be possible to determine
   the appropriate human operator to which the call should be routed.

   15. Routing of call to operator workstation. Once the appropriate
   operator has been identified, the call must be routed to that
   operator's workstation.

   16. Whisper. Once the operator answers the call, the previously
   recorded request should be played to the operator as a "whisper",
   prior to connecting the caller to the operator.

   17. Connection of caller to operator. Once the operator has heard
   the whisper, the caller can be connected to the human operator. The
   operator queries the caller for the request, and initiates a query
   to the listing database.

   18. Playing listing information. Once the listing information is
   returned from the database, the caller must be connected to a media
   resource which speaks the listing information to the caller.

   19. Prompting for call completion. If the identified service
   includes call completion, the caller should be prompted for this
   service, for example by pressing some DTMF key.

   20. Call completion. If the caller requests call completion, the
   call should be automatically initiated for the caller.



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5. OISP Internal Architecture

   This section describes an initial view of the elements internal to
   the OISP architecture.

   The following types of elements may be present within the OISP
   infrastructure:

   Automatic Call Distributor (ACD) server - The ACD provides queuing
   and call distribution functions for human operators. Specifically,
   it is the component that tracks the availability of the human
   operators and selects an available operator utilizing complex
   algorithms based on operator skills, type of call, type of request,
   calling party information, etc. The ACD server is modeled as an
   Application Server.

   Customer Profile Database - The Customer Profile Database is a per
   subscriber database maintained by an OISP about its customers. Some
   of this information might be statically provisioned, other
   information such as customer preferences or session information
   might be populated dynamically as a result of customer
   interactions.

   Messaging Gateways - Messaging Gateways provide access and data via
   E-mail, SMS, MMS, WAP.

   Operator and Information Services Application Server (OIS-AS) - The
   OIS-AS contains AS functions specifically for directory assistance
   and information services as well as other Operator Services. This
   may coordinate multiple call legs, connecting the caller in
   sequence to one or more OIS-MS and/or operator workstations
   according to the logic contained within. The OIS-AS may need to
   communicate with elements of other providers, for instance to query
   information about the capabilities and characteristics of the
   caller's equipment, or to access another provider's operator
   resources.

   Operator and Information Services Media Server (OIS-MS) - The OIS-
   MS provides the media resources for playing announcements,
   performing voice recognition, performing listing database queries,
   generating whisper from the user's verbal request, etc.

   Operator Workstations - Operator Workstations provide an interface
   to the human operator. It may receive the customer's recorded
   request (e.g. name and city of requested listing), information from


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   listing or other databases, and also terminate a multimedia session
   with the customer. These are modeled as SIP UAs.

   Service Databases - Service Databases store service specific
   information (e.g. listing information such as name, address, and
   phone number, etc.) These may be located in the OISP's network
   and/or in other networks, and more than one may be used.

   SIP Proxy - One or more SIP proxies may be present in the OISP
   network, to facilitate SIP communications with other providers as
   well as to perform call processing functions between OISP
   components.

   The following figure shows a simplified view of an OISP internal
   architecture. The lines show logical connectivity; elements
   communicate via the proxy. A single OIS-AS is shown, along with up
   to "k" OIS-MS and up to "m" Operator Work Stations, and an ACD
   server.




                +--------+   +---------+
             +--| OIS-AS |-+-| OIS-MS1 |
             |  +--+-----+ | +---------+
   +-------+ |     |       |
   | Proxy |-|     |       | +---------+
   +-------+ |     |       +-| OIS-MSk |
             |  +--+--+      +---------+
             +--| ACD |---------+---------+
                +-----+         |         |
                             +--+---+  +--+---+
                             | OWS1 |  | OWSm |
                             +------+  +------+

          Figure 4 Simplified view of OISP Internal Architecture





6. General Approach

   This section describes one possible way to implement DA using SIP.
   Other ways may be possible.




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   The general approach involves having the OIS-AS host most of the
   processing logic, and to control the call in general. The OIS-AS
   implements a third party call control (3PCC) functionality. It
   terminates the signaling dialog from the caller, and originates
   dialogs towards other components as necessary. There may be
   multiple sequential sessions set up during the course of a DA call.

   For example, the OIS-AS would initiate a new dialog towards a MS
   for front-end automation. When it gets the 200 OK from the MS with
   SDP, it passes that SDP back toward the caller. When the front end
   automation has completed, the OIS-MS sends a BYE containing message
   bodies conveying the success of the operation (i.e., was it able to
   obtain the listing) as well as any data related to the operation.
   In case of success, the body might carry the listing information,
   or a URI pointing to it. In case of failure, it might carry a URI
   pointing to the whisper file.

   In case of failure, the OIS-AS would determine that the call needs
   to be routed to a human operator. The OIS-AS first needs to
   identify a suitable operator to handle this request. The ACD server
   has this responsibility, and could be implemented as a redirect
   server facing the OIS-AS, redirecting towards the best suited
   available operator. Facing the operator workstations, the ACD
   server could be implemented as a presence server, maintaining
   availability of each operator, as well as the associated
   information for each (e.g. languages, skill level, cost, etc).

   The OIS-AS would then send an INVITE toward the identified operator
   workstation. This INVITE includes the caller's SDP as well as a URI
   pointing to the whisper file. The workstation could play the
   whisper to the operator as the call is answered. The operator
   workstation's SDP would be passed back to the caller via a re-
   INVITE or UPDATE request.

   If the operator is successful in locating the desired listing, the
   workstation would send a BYE containing message bodies with an
   indication of success, and either the listing information of a
   pointer to the same.

   The OIS-AS would then initiate a call leg towards an OIS-MS for
   back end automation. The INVITE would include the same body with
   the listing information that was sent by the operator workstation.
   The OIS-MS returns its SDP, which the OIS-AS would propagate back
   over the originating leg via a re-INVITE or UPDATE request. The
   back end automation process includes audibly playing out the
   listing information, and possibly offering call completion service.



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   The OIS-MS sends a BYE with a message body indicating whether call
   completion is desired.

   If call completion is desired, the OIS-AS sends a REFER back over
   the originating call leg to the caller, and clears the call.

   These examples describe simple voice scenarios. Other media types
   may be possible. For example, it may be desirable to send the
   listing information via text message to the caller's terminal, or
   to show a video clip. Such features require knowledge of the
   calling terminal's capabilities and characteristics. The mechanism
   described in RFC 3840 Indicating User Agent Capabilities in the
   Session Initiation Protocol (SIP)can be used for this. The
   capabilities might have been signaled in the initial INVITE
   request. Otherwise, the OIS-AS can query for capabilities using an
   OPTIONS request. Additionally, some non SIP mechanism might be
   used, such as querying a database (e.g. IMS HSS) in the caller's
   network.

   References to a whisper file can be passed using the mechanism
   described in RFC 4483 A Mechanism For Content Indirection in the
   Session Initiation Protocol (SIP).

   Other information signaled via message bodies includes the success
   or failure status of operations (such as identifying the requested
   listing), or other data (such as the listing information).

   Context information may be maintained on a per call basis. It could
   include such information as the caller's preferred language, etc. A
   URI pointing to the context information could be passed between
   elements in the OISP infrastructure.





7. Signaling Mechanisms

   This section discusses the signaling mechanisms required to provide
   OIS services.

7.1. Calling Party's Identity

In many cases, downstream providers may need to know the calling
party's identity. This might be needed to influence call processing,
or for accounting purposes. Also, the calling party's identity in the



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form of a SIP URI might be needed so that the identity of the home
network can be determined.

The calling party's equipment populates the From header in SIP
messages. This is not trusted. There are several methods for providing
"network-asserted identities", which under the appropriate conditions
can be trusted.

The SIP Identity mechanism defined in [SIP-IDENT] provides a
standardized, architecture agnostic SIP mechanism for
cryptographically assuring the user's identity.

The P-Asserted-Identity header [PAI] is a private extension which can
be used to carry a network asserted identity of the caller between
trusted providers.

Note that some networks may allow their users to hide their identity.
In the current North American PSTN, for such cases the caller id
information is often transported through the network, marked with a
privacy indication such that it will not be presented to the called
party.

Bilateral agreements between VOIP providers determine whether
providers are within the same "trust domain" as defined in [RFC3324],
and what information, including network-asserted identities, may be
exchanged between providers. Depending on such agreements, it is
possible that the caller identity information is obscured or
completely absent. As indicated in [PAI], "Masking identity
information at the originating user agent will prevent certain
services, e.g., call trace, from working in the Public Switched
Telephone Network (PSTN) or being performed at intermediaries not
privy to the authenticated identity of the user."

When an OISP is outside any trust domain with the caller's home
network, or is not otherwise privy based on bilateral agreements to
network asserted identity information from the calling network when
the caller has requested privacy, it is unable to implement any call
processing logic based on such information.

If the OISP desires to reject anonymous calls, a mechanism is proposed
in "Rejecting Anonymous Requests in the Session Initiation Protocol
(SIP) - draft-ietf-sip-acr-code-03", which defines a specific response
code for this.

The following shows an example of an INVITE message contain a P-
Asserted-Identity header.



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   INVITE sip:da@provider-c.com SIP/2.0
   Via: SIP/2.0/UDP proxy-b.provider-b.com:5060 ;branch=y9hG4bK74bf9
   Via: SIP/2.0/UDP proxy-a.provider-a.com:5060 ;branch=x9hG4bK74bf9
   Via: SIP/2.0/UDP client.provider-a.com:5060 ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: <sip:7327581234@provider-a.com>
   P-Asserted-Identity: "732758123" <sip:73237581234@provider-a.com>
   Content-Type: application/sdp
   Content-Length: ...
   [SDP not shown]




7.2. Provider Identification

   As discussed, in some deployment scenarios, the OISP makes use of
   the identities of other providers involved in the call. This
   section discusses how those identities can be conveyed using SIP.



7.2.1. Home Provider

In many cases, the OISP needs to identify the caller's Home Provider.
This may be needed for branding purposes as well as to potentially
influence treatment in other ways.

The basic mechanism for determining the home network is to derive it
from the right hand side (RHS) of the network asserted identity.

In SIP, identities are expressed as URIs. These can be SIP (or SIPS)
URIs, or "tel" URIs.

[1] defines the SIP URI, which is used for identifying SIP resources.
The RHS can be expressed as a DNS domain name or as an IPv4 or IPv6
address. The hostname format is most suitable for providing an
identity to reach the calling party. For instance the mechanisms
defined in [RFC3263] for locating SIP servers depends on the use of
domain names for the various types of DNS lookups such as NAPTR, SRV,
and A.

If a provider decides to provide network asserted identities expressed
as SIP URIs using IP addresses instead of hostnames, it forfeits the


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use of such standardized mechanisms for reaching its users. It also
becomes difficult to derive the home network identity from the network
asserted identity.

RFC3966 defines the "tel" URI, which is used for describing resources
identified by phone numbers. The "tel" URI format does not include a
hostname. Thus, if the network asserted identity includes only a "tel"
URI, no direct information about the home network is provided.

The SIP Identity mechanism is intended for use with SIP URIs. The PAI
mechanism can use either a SIP URI, a "tel" URI, or both.

This document depends on the home network providing a network asserted
identity containing a hostname. This includes the SIP identity where
the SIP URI contains a hostname, or a PAI header containing at least a
SIP URI with a hostname.

Very simply, the RHS of the hostname in the SIP URI is extracted and
used as the basis to influence call processing. In cases where the
caller's identity is not available, as discussed in the  "Calling
Party's Identity" section, then the home network's identity is
consequently also not available, and call processing logic based on
such information (such as branding) cannot take place.



7.2.2. Last Hop Provider

In many cases, the OISP needs to identify the last hop provider; that
is, the provider which sent the call to the OISP. This may be needed
for accounting purposes, and also to potentially influence treatment
in other ways.

Mutual TLS authentication is often used by SIP peers to authenticate
each other. Authentication by definition means that the identity of
the other party is unambiguously verified. Using mutual TLS, the right
hand side of the SubjectAltName field in the X.509 certificate would
identify the previous provider.

Other methods of identifying the previous network's identity include
the use of HTTP challenge authentication, where a cryptographic
challenge verifies the asserted identity. The transport and/or network
layer address of the peer could also be used, though this presents
significant security risks.





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In the absence of mutual TLS, the "host" field of the "sent-by" field
of the topmost mandatory Via header can be used to identify the last
hop network.

The Via header could be populated with a DNS hostname or an IP
address. If populated with a hostname, it is possible to derive the
identity of the last hop network directly from the domain portion of
the hostname. If it is populated with an IP address, this step may not
be possible. Configuration data may need to include both domain names
and lists of IP addresses associated with last hop networks.





7.2.3. Arbitrary Traversed Provider

In some cases, the OISP may need to know the identity of some provider
involved in the call which is neither the Home Provider nor the last-
hop provider. This may be needed to influence treatment.

The use of the SIP History-Info mechanism defined in RFC 4244, An
Extension to SIP for Request History Information, can be used for
this. As the call moves from one provider to the next and is
retargeted, corresponding entries are added to the SIP History-Info
header. If the domain name format is used for the retargeted entities,
then the History-Info header now includes a list of traversed SIP
domains or providers, which can be consulted by the OISP.

According to RFC 4244, entries should be added to the History-Info
header whenever the Request-URI is modified. Cases may exist where the
call is sent to another provider but the URI is not modified. In such
cases, the provider is not captured by the History-Info header.

The following figure illustrates the use of the History-Info header
for this purpose.













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    Caller        Provider-A     Provider-B     Provider-C
      |              |              |              |
      |              |              |              |
      |              |              |              |
      |(1) INVITE tel:+411          |              |
      |------------->|              |              |
      |              |              |              |
      |              |              |              |
      |              |(2) INVITE sip:da@prov-b.net |
      |              |------------->|              |
      |              |              |              |
      |              |              |              |
      |              |              |(3) INVITE sip:da@prov-c.net
      |              |              |------------->|
      |              |              |              |
      |              |              |              |

   Figure 5 Use of History-Info header to identity traversed providers




1. The user dials "411", and the resulting INVITE to its home proxy is
for "tel: +411". No History-Info header is included yet.

   INVITE tel:+411 SIP/2.0
   (other message content omitted)




2. The home proxy retargets this to "sip:da@prov-b.net", and adds a
History-Info header which includes the targeted-from URI:

   INVITE sip:DA@prov-b.net SIP/2.0
   History-Info: <tel: +411>; index=1
   (other message content omitted)


3. Proxy-B retargets this to "SIP: da@prov-c.net", and appends another
entry to the History-Info header:

   INVITE sip:DA@prov-b.net SIP/2.0
   History-Info: <tel: +411>; index=1, <sip:da@prov-b.net>; index=2
   (other message content omitted)




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When this request arrives a Proxy-C in Provider C (OISP), it conveys
the following:

     oThe Request-URI (SIP: da@prov-c.net) indicates this as a DA call

     oThe History-Info header conveys the history of the request:

     oIt started as a tel URI for digits "411"

     oIt was then targeted to provider B

     oIt is now targeted to provider C



7.3. Originating Station Type

In the current PSTN in North America, OIS providers have the ability
to tailor treatment based on the type of originating station. For
instance, calls from prison phones are restricted from accessing DA
services. Example values include POTS, coin, hospital, prison/inmate,
cellular, etc. In the PSTN in North America, this information is
signaled for SS7 calls using the Originating Line Information (OLI)
information element, and in MF calls using the ANI II digits. To
support interworking with the PSTN, it must be possible to convey the
Originating Station Type value.

Ways to represent this information in SIP need to be explored. There
are currently two proposals being considered in the IETF which might
potentially satisfy this requirement.

     oThe Calling Party's Category tel URI Parameter - draft-mahy-
        iptel-cpc-04.txt (work in progress) [IPTEL-CPC]

This defines a new parameter "cpc" which is applied to the SIP or tel
URI of the calling user. It allows for values such as "ordinary",
"prison", "police", "test", "operator", "payphone", "unknown",
"hospital", "cellular", "cellular-roaming". An example from the
internet draft:

   INVITE sip:bob@biloxi.example.com SIP/2.0
   To: "Bob" <sip:bob@biloxi.example.com>
   From: <tel:+17005554141;cpc=payphone>;tag=1928301774






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     oConveying Calling Party Category (CPC) and Originating Line
        Information (OLI) using the Security Assertion Markup Language
        (SAML) - draft-schubert-sipping-saml-cphc-02.txt [CPC-SAML]

While [IPTEL-CPC] is simple to implement, [CPC-SAML] provides a
cryptographically verifiable assertion. Both are currently works in
progress, and any document with normative dependencies to such works
cannot be published until the works in progress are published.
Further, there is an open question as to whether [IPTEL-CPC] can carry
OLI information as well as CPC or ANI II information.



7.4. Trunk Group Identifier

The incoming trunk group number provides information which could be
used to influence call processing, thus this information is needed.
Trunks are point to point circuits and as such, their remote
termination point is unambiguously known. As such, knowledge of the
incoming trunk group conveys the identity of the provider offering the
call.

For PSTN interworking, the incoming trunk group identifier is a key
piece of information and must be known. Thus, at a PSTN to IP
interworking point, the trunk group information must be kept and
signaled forward. This holds for OISP's accepting incoming calls from
the PSTN as well as upstream providers accepting calls from the PSTN.

"Representing trunk groups in tel/sip Uniform Resource Identifiers
(URIs)" - draft-ietf-iptel-trunk-group-10.txt defines a way to signal
incoming and/or outgoing trunk group information as a parameter in SIP
URIs and tel URIs.

To represent incoming trunk groups, the trunk group parameter is
included in the Contact header of the SIP message. The "trunk-context"
parameter should also be included, to ensure that the trunk group is
unambiguously identified, since trunk group numbers are not globally
unique.

The following example shows an INVITE containing a trunk group
identification in the Contact header:








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   INVITE sip:da@provider-c.com SIP/2.0
   Via: SIP/2.0/UDP proxy-b.provider-b.com:5060 ;branch=y9hG4bK74bf9
   Via: SIP/2.0/UDP proxy-a.provider-a.com:5060 ;branch=x9hG4bK74bf9
   Via: SIP/2.0/UDP client.provider-a.com:5060 ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: < sip:7327581234@provider-b.com;tgrp=101; trunk-
   context=provider-b.com@proxy-b.provider-b.com;user=phone>
   P-Asserted-Identity: "7327581234" <sip:73237581234@provider-a.com>
   Content-Type: application/sdp
   Content-Length: ...




7.5. Dialed Digits

Currently in the North American PSTN, the OIS provider may take into
account the digits dialed by the user. In that scenario the dialed
digits are frequently forwarded to the OIS provider.

Using SIP, the dialed digits would typically be sent by the user's
equipment in the form of a tel URI or SIP URI in the Request-URI of a
SIP INVITE. It is possible that retargeting could take place, in which
case the dialed digits would be lost.

The SIP History-Info mechanism defined in RFC 4244 provides a
mechanism for solving exactly this type of problem. It defines a new
optional SIP header, History-Info, to provide a standard mechanism for
capturing the request history information. Whenever a node which
supports this mechanism modifies the Request-URI of a request, it
captures this in the History-Info header.

The following example shows an INVITE containing a History-Info
header, which conveys the original dialed digits, after having been
retargeted.

   INVITE sip:DA@prov-b.net SIP/2.0
   (other message content omitted)
   History-Info: <tel: +411>; index=1, <sip:da@prov-b.net>; index=2


Please see the section titled "Arbitrary Involved Provider" for an
example of a flow where the History-Info mechanism delivers the dialed
digits to the OISP when retargeting has occurred.




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7.6. Retargeting to Identify the Desired Service

It is necessary to identify the service being requested. Such services
might include directory assistance with or without call completion.
The logic to determine this might reside in one or more points in the
network. Additionally, the identification of the service might be
refined as the request traverses potentially multiple networks,
depending on the availability of additional information.

It is proposed here to retarget the Request-URI of the SIP request to
specify the desired service. While the initial Request-URI might
specify "SIP:411@provider-a.net", a downstream element might invoke
service logic and determine that this call should be sent to OISP C's
network for directory assistance with call completion, and change the
Request-URI to "SIP:da-with-call-completion@oisp-c.net".

A similar approach is taken for identifying resources in [NETANN].

[CSI], a work in progress, discusses explicit service identifiers for
using in IMS based networks.



7.7. Charge Number

   There is a need to convey a charge number, which may differ from
   the calling party's identity. The charge number usually identifies
   the customer or account with which the caller is associated, e.g.
   the main number for a business which has many individual calling
   numbers. This might be needed for accounting, but it also could
   influence call processing, especially when a particular type of
   service applies for any caller associated with a particular charge
   number.

   The ability to convey charge number information is currently
   lacking in SIP. It has been suggested in Analysis of TISPAN
   requirements for Connected Identity in the Session Initiation
   Protocol (SIP) - draft-elwell-sip-tispan-connected-identity-01.txt
   that the P-Asserted-Identity header can be used to convey this
   information, with the caller's identity in the From header.
   However, using the P-Asserted-Identity header and From header to
   convey separate information is seen as controversial and has not
   been accepted by the IETF.






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7.8. Passing Whisper

   During front end automation, the OIS-MS will record and may time
   compress the caller's perhaps meandering speech into what is known
   as the "whisper". This is intended to be played into a human
   operator's ear, should the call be referred to an operator, to
   avoid the operator from having to prompt the caller again. The
   whisper is obtained during the front end automation, and saved to
   an audio file.

   If the call needs to be transferred to a human operator, the
   whisper will need to be played to the operator at the same time or
   slightly prior to connecting the caller to the operator. Thus, the
   operator workstation needs to be able to access the whisper file.

   When the OIS-MS performs front end automation, it generates the
   whisper and saves it as an audio file. The location, storage type,
   and format are out of the scope of this document. What is needed is
   a way for the OIS-MS to convey the whisper information to the OIS-
   AS, so it could potentially be used for later processing, such as
   passing to a human operator.

   Due to size constraints, it may not be feasible or desirable to
   pass the actual audio file containing the whisper. This document
   will discuss the most general case of passing a pointer, in the
   form of a URI, to the audio content.

   Since the whisper is an output of the front end automation process,
   it makes sense to return this upon completion of that process. The
   most reasonable time to do this is when the OIS-MS sends the BYE.

   Any SIP request, including BYE, can contain a message body. RFC
   4483 A Mechanism for Content Indirection in Session Initiation
   Protocol (SIP) Messages defines an extension to the URL MIME
   External-Body Access-Type to satisfy the content indirection
   requirements for SIP. These extensions are aimed at allowing any
   MIME part in a SIP message to be referred to indirectly via a URI.

   This is illustrated in the following figure. Note that the proxy
   has been omitted for clarity, as have some messages not crucial to
   illustrating the use of this mechanism. All SIP signaling traverses
   the proxy.







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            AS             MS          Operator
             |              |              |
             |              |              |
             |              |              |
             |(1) INVITE    |              |
             |------------->|              |
             |              |              |
             |              |              |
             |(2) 200 OK    |              |
             |<-------------|              |
             |              |              |
             |              |              |
             |MS prompts user, collects whisper
             |              |              |
             |              |              |
             |              |              |
             |(3) BYE, body w. status, whisper URI
             |<-------------|              |
             |              |              |
             |              |              |
             |(4) 200 OK    |              |
             |------------->|              |
             |              |              |
             |              |              |
             |(5) INVITE w. whisper URI    |
             |---------------------------->|
             |              |              |
             |              |              |
             |(6) 200 OK SDP|              |
             |<----------------------------|
             |              |              |
             |              |              |
             |              |              |
             |              |              |

            Figure 6 Call flow illustrating passing of whisper





   1. INVITE AS->MS
   INVITE sip:da@ms-1.oisp-c.net SIP/2.0
   [remainder of message omitted]




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   2. 200 OK MS->AS
   SIP/2.0 200 OK
   [remainder of message omitted]


   3. BYE MS->AS
   BYE sip:as-1@as-1.oisp-c.net SIP/2.0
   [non relevant portions of message omitted]
   Content-Type: message/external-body; access-type="URL";
       URL="http://ms1.oisp-c.net/whisper/20070206092700-0001.wav"
       expiration="Tues, 06 Feb 2007 09:30:00 GMT";
   <CRLF>
   Content-Type: audio/x-wav
   Content-Disposition: render
   <CRLF>



   4. 200 OK AS->MS
   SIP/2.0 200 OK
   [remainder of message omitted]


   5. INVITE AS->Operator Workstation
   INVITE sip:operator@operator-123.oisp-c.net SIP/2.0
   [non relevant portions of message omitted]
   Content-Type: message/external-body; access-type="URL";
       URL="http://ms1.oisp-c.net/whisper/20070206092700-0001.wav"
       expiration="Tues, 06 Feb 2007 09:30:00 GMT";
   <CRLF>
   Content-Type: audio/x-wav
   Content-Disposition: render
   <CRLF>


   6. 200 OK Operator->AS
   SIP/2.0 200 OK
   [remainder of message omitted]


   Note that this same mechanism also supports the case where front
   end automation is performed by one provider, and another provider
   provides the operator assistance. In this type of scenario,
   provisions need to made such that the second provider can access
   the resources referenced by the URI.




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7.9. Calling Equipment Capabilities and Characteristics



   It may be necessary for the OIS provider to learn the capabilities
   and characteristics of the caller's equipment. This would be useful
   when the OIS provider wishes to provide content to the caller other
   than that which was used on the call to the OISP. For example, the
   OIS provider might wish to send listing information via text
   message, or play a video clip about a particular venue about which
   he has requested information.

   RFC 3840 Indicating User Agent Capabilities in the Session
   Initiation Protocol (SIP), defines mechanisms by which a UA can
   convey its capabilities and characteristics to other user agents
   and to the registrar for its domain. This information is conveyed
   as parameters of the Contact header field.

   This information might be included in the incoming INVITE to the
   OISP, if the caller's UA supports this mechanism and is configured
   to do so. Otherwise, the OISP could query the caller's UA by
   sending a SIP OPTIONS request, and the UA, if it supports this
   mechanism, would include its capability feature tags in the
   response to the OISP.

   The following is an example of an INVITE containing capability
   feature tags, as it arrives at the OISP. In this case, the UA
   supports audio, video, and text. Other included tags provide
   additional information.



   INVITE sip:da@provider-c.com SIP/2.0
   Via: SIP/2.0/UDP proxy-b.provider-b.com:5060 ;branch=y9hG4bK74bf9
   Via: SIP/2.0/UDP proxy-a.provider-a.com:5060 ;branch=x9hG4bK74bf9
   Via: SIP/2.0/UDP client.provider-a.com:5060 ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: <sip:7327581234@provider-a.com>;audio;video;text
        ;actor="principle";automata;mobility="fixed"
        ;methods="INVITE,BYE,OPTIONS,ACK,CANCEL"
   P-Asserted-Identity: "7327581234" <sip:73237581234@provider-a.com>
   P-Asserted-Identity: tel:+7327581234
   Content-Type: application/sdp
   Content-Length: ...
   [SDP not shown]



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   If the OISP wishes to query the UA, it can send an OPTIONS request
   to the UA, and the UA, if it supports this mechanism, would include
   the feature capability tags in the Contact header, as show above,
   in the 200 OK response.





7.10. Media Server Returning Data to the Application Server

   The OIS-AS needs to know the outcome of the operations performed by
   the OIS-MS, e.g. success/failure of front end automation, etc. Some
   mechanism is needed to convey this information. This could be
   conveyed using non SIP mechanisms.

   Any SIP message, including BYE, can carry message bodies. The
   simplest way for a OIS-MS to return data to an OIS-AS is to
   encapsulate the data in a MIME body. This requires agreement
   between both sides on the format and semantics of these bodies.

   Another approach is to use the content indirection mechanism to
   point to the data, however this may be rather cumbersome if only a
   small amount of data is to be returned.

   Some OIS service may make use of VoiceXML, whereby the OIS-AS
   invokes VoiceXML scripts on the OIS-MS, and the OIS-MS returns data
   to the OIS-AS. SIP Interface to VoiceXML Media Services - draft-
   burke-vxml-02.txt (work in progress) describes a SIP interface to
   VoiceXML media services, which is commonly employed between
   application servers and media servers offering VoiceXML processing
   capabilities. This may be found useful for OIS services.

   This information can also be conveyed using non SIP mechanisms.
   Describing such mechanisms is out of the scope of this document.



7.11. Service Discovery

   An OISP might desire that its service be discoverable on the
   internet, instead of or in addition to static provisioning into
   provider networks. The Service URN concept discussed in the work in
   progress "A Uniform Resource Name (URN) for Services - draft-ietf-
   ecrit-service-urn-05" can be used to facilitate this. This allows
   for discovery of the service in a context dependent manner, where
   context could include for example the user's location. Thus a user


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   agent could send a SIP request to "urn: service: info", and this
   very generic URI could be resolved to a point to a specific network
   element belonging to a specific provider. If some other context
   information such as the user's location is available, this could be
   used to refine the resolution to e.g. an element best suited for
   that particular location.



8. Call Flow

   The following call flow provides examples of how a DA service could
   be implemented using the mechanisms described in this document. It
   is intended to illustrate the intended use of the proposed
   signaling mechanism. Some messages not crucial to this may be
   omitted for clarity.

































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   Caller    Proxy A   Proxy B   Proxy C   OIS-AS    OIS-MS1
      |         |         |         |         |         |
      |         |         |         |         |         |
      |         |         |         |         |         |
      |(1) INVITE tel:411 |         |         |         |
      |-------->|         |         |         |         |
      |         |         |         |         |         |
      |         |(2) INVITE sip:da@prov-b.com |         |
      |         |-------->|         |         |         |
      |         |         |         |         |         |
      |         |         |(3) INVITE sip:da@prov-c.com |
      |         |         |-------->|         |         |
      |         |         |         |         |         |
      |         |         |         |(4) INVITE sip:da-cc@prov-c.com
      |         |         |         |-------->|         |
      |         |         |         |         |         |
      |         |         |         |         |(5) INVITE prompt &
   collect
      |         |         |         |         |-------->|
      |         |         |         |         |         |
      |         |         |         |         |(6) 200 OK w.SDP
      |         |         |         |         |<--------|
      |         |         |         |         |         |
      |         |         |         |(7) 200 OK w.SDP   |
      |         |         |         |<--------|         |
      |         |         |         |         |         |
      |         |         |(8) 200 OK w.sdp   |         |
      |         |         |<--------|         |         |
      |         |         |         |         |         |
      |         |(9) 200 OK w.sdp   |         |         |
      |         |<--------|         |         |         |
      |         |         |         |         |         |
      |(10) 200 OK w.sdp  |         |         |         |
      |<--------|         |         |         |         |
      |         |         |         |         |         |
      |         |         |         |         |         |
      |         |         |         |         |         |


                        Figure 7 Call flow, part 1





   For brevity, only relevant SIP headers will be shown. The following
   test refers to Figure 7.


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   The user, homed in provider A, initiates a request for an OIS
   service, for instance by dialing "411". The user's UA sends a SIP
   INVITE. It might contain a "tel" URI.

   1. INVITE UE -> Home Proxy

   INVITE tel:+411 SIP/2.0
   Via: SIP/2.0/UDP client.provider-a.com:5060
   ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: <sip:7327581234@provider-a.com>
   Content-Type: application/sdp
   Content-Length: ...




   The home network knows nothing of OISP services, for instance it
   might be a rather small scale provider. It is essentially set up to
   forward all calls of this type to Provider B. It translates the
   Request-URI to a SIP URI and sends the call on to provider B.
   Because of this retargeting, it adds a History-Info header to
   capture the dialed digits.



   The caller's identity is verified in a manner consistent with this
   provider's policies, and the proxy adds two P-Asserted-Identity
   headers: one with a SIP URI, and another with a "tel" URI.



















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   2. INVITE proxy-a -> proxy-b

   INVITE sip:411@provider-b.com SIP/2.0
   Via: SIP/2.0/UDP proxy-a.provider-a.com:5060 ;branch=x9hG4bK74bf9
   Via: SIP/2.0/UDP client.provider-a.com:5060
   ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: <sip:7327581234@provider-a.com>
   P-Asserted-Identity: "732758123" <sip:73237581234@provider-a.com>
   P-Asserted-Identity: tel:+7327581234
   History-Info: <tel: +411>; index=1
   Content-Type: application/sdp
   Content-Length: ...



   Proxy-b in provider-b's network receives the request. This is a
   larger network, and it has business relationships with several OIS
   providers, as well as with several providers which serve
   subscribers. This provider has logic which requires it to query the
   Home Provider's network to find some information related to the
   caller. This is not likely to be a SIP related function, and is
   thus out of scope for this document. The logic executes, taking the
   result of this query into account. It is determined that the call
   is for directory assistance, and that the call should be routed to
   provider C for handling.



   So, proxy-b retargets the Request-URI to reflect this, and routes
   the call to provider C (the OISP). It adds another entry to the
   History-Info header to capture this retargeting.














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   3. INVITE proxy-b -> proxy-c

   INVITE sip:da@provider-c.com SIP/2.0
   Via: SIP/2.0/UDP proxy-b.provider-b.com:5060 ;branch=y9hG4bK74bf9
   Via: SIP/2.0/UDP proxy-a.provider-a.com:5060 ;branch=x9hG4bK74bf9
   Via: SIP/2.0/UDP client.provider-a.com:5060
   ;branch=z9hG4bK74bf9
   From: <sip:7327581234@provider-a.com>;tag=1234567
   To: 411 <tel:+411>
   Contact: <sip:7327581234@provider-a.com>
   P-Asserted-Identity: "732758123" <sip:73237581234@provider-a.com>
   P-Asserted-Identity: tel:+7327581234
   History-Info: <tel: +411>; index=1, <sip:da@provider-a.com>;
   index=2
   Content-Type: application/sdp
   Content-Length: ...



   Proxy-c in provider C's network receives the request. The source of
   the request is authenticated via mechanisms not described here. It
   needs to know how to bill this call, and thus needs to know which
   provider it came from. It looks at the topmost Via header, and sees
   that the call came from provider B.

   It examines the History-Info header, and is able to identity the
   dialed digits. It can also determine from the SIP URI which domains
   have been traversed, as long as each has retargeted and appended an
   entry in the header.

   The proxy determines that the call needs to go the OIS-AS for
   handling, so it retargets if necessary and forwards the INVITE.

   The OIS-AS performs 3PCC. It determines that the call needs a
   branded announcement based on the identity of the home network,
   which it derives from the P-Asserted-Identity header. It initiates
   a new call leg toward OIS-MS1 for front end automation. Per RFC
   4240, the "dialog" portion of the Request-URI indicates the "prompt
   & collect" service. The URI identifies the VoiceXML script to be
   executed. The SDP is the caller's SDP.








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   5. INVITE OIS-AS -> MS1

   INVITE sip:dialog@ois-as.prov-c.com; \
          voicexml=http://vxmlserver.example.net/cgi-bin/script.vxml \
   SIP/2.0
   Via: SIP/2.0/UDP ois-as.prov-c.com:5060
   ;branch=z9hG4bK74bf9
   From: <sip:ois-as@ois-as.prov-c.com>;tag=1234567
   To: sip:dialog@ois-as.prov-c.com; \
          voicexml=http://vxmlserver.example.net/cgi-bin/script.vxml
   Contact: <sip:ois-as@ois-as.prov-c.com>
   Content-Type: application/sdp
   Content-Length: ...



   The OIS-MS responds with a 200 OK, with its own SDP. The OIS-AS now
   sends a 200 OK response back toward the caller, with the MS's SDP.
   Note that the OIS-AS could first have sent non final response back
   toward the user.




























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   Caller    OIS-AS    OIS-MS1     ACD    Operator
       |         |         |         |         |
       |         |         |         |         |
       |         |         |         |         |
       |(11) RTP session   |         |         |
       |...................|         |         |
       |         |         |         |         |
       |         |(12)BYE w.URI, body|         |
       |         |<--------|         |         |
       |         |         |         |         |
       |         |(13)INVITE         |         |
       |         |------------------>|         |
       |         |         |         |         |
       |         |(14)3xx  |         |         |
       |         |<------------------|         |
       |         |         |         |         |
       |         |(15)INVITE w.URI   |         |
       |         |---------------------------->|
       |         |         |         |         |
       |         |(16)200 OK         |         |
       |         |<----------------------------|
       |         |         |         |         |
       |(17) re INVITE     |         |         |
       |<--------|         |         |         |
       |         |         |         |         |
       |(18) 200 OK        |         |         |
       |-------->|         |         |         |
       |         |         |         |         |
       |(19) RTP session   |         |         |
       |.......................................|
       |         |         |         |         |
       |         |(20) BYE |         |         |
       |         |<----------------------------|
       |         |         |         |         |
       |         |         |         |         |
       |         |         |         |         |
                        Figure 8 Call flow, part 2



   The following text refers to Figure 8.

   The user is now connected (11) to the MS, which plays a branded
   announcement, and prompts for the listing information. When the
   user speaks his request, the MS processes the audio to obtain a


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   "whisper" file, or condensed version of the request. In this
   example, the MS is unable to successfully perform the query, so it
   terminates the call be sending a BYE (12) to the OIS-AS. This BYE
   also contains a URI which points to the whisper file, and also
   contains a message body (not shown) containing the output of the
   VoiceXML script.

   The OIS-AS decides based on the failure indication that it needs to
   route the call to a human operator. It sends an INVITE (13) to the
   ACD server. One possible way an ACD could be implemented is as a
   presence server, such that it keeps track of the availability of
   all the operators.

   In this example, the ACD server is implemented as a redirect server
   - it sends a 3XX response (14) which identifies the operator the
   OIS-AS should contact. Alternately, the ACD server could have
   proxied the request to the operator.

   The OIS-AS now sends an INVITE (15) containing the URI to the
   whisper, as well as the caller's SDP, to the indicated operator
   workstation. The operator workstation sends a 200 OK (16) with the
   operator's SDP, which the OIS-AS sends toward the caller in a re-
   INVITE (17).

   The caller is now connected to the operator (19), and the operator
   helps the caller with the listing. The operator workstation
   launches a query, and a response is received. The operator signals
   a BYE (20) toward the OIS-AS, which may contain the listing
   information in a message body, a pointer (URI) to the listing
   information, or it may pass this information to the OIS-AS using
   some other, non SIP mechanism.


















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          Caller    OIS-AS    OIS-MS2
             |         |         |
             |         |         |
             |         |         |
             |         |(21) INVITE
             |         |-------->|
             |         |         |
             |         |(22) 200 OK
             |         |<--------|
             |         |         |
             |(23) re INVITE     |
             |<--------|         |
             |         |         |
             |(24) 200 OK        |
             |-------->|         |
             |         |         |
             |(25) RTP session   |
             |...................|
             |         |         |
             |         |(26) BYE w.body
             |         |<--------|
             |         |         |
             |(27) REFER         |
             |<--------|         |
             |         |         |
             |         |         |
             |         |         |
                        Figure 9 Call flow, part 3



















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   The following text refers to Figure 9.

   The OIS-AS sends an INVITE (21) to another OIS-MS, MS2, for back
   end automation. When it receives MS2's SDP in the 200 OK (22), it
   sends a re INVITE (23) toward the user to update the SDP. At this
   point an audio session is in place between the caller and the back
   end automation MS (25). The MS plays the listing information, and
   offers call completion service. The caller accepts, so OIS-MS2
   sends a BYE (26) with a message body containing the result of the
   call completion offer. Since call completion was requested, the
   OIS-AS sends a REFER (27) to the caller, to cause it to place a
   call to the listed party. The OIS-AS may or may not care about
   subsequent NOTIFY from the caller, and drops out of the call.





9. VoIP Information Services - Summary and Next Steps

A list of information which needs to be conveyed has been developed,
and candidate proposals identified for each of these.

The desired next steps include soliciting feedback from the IETF
community on the choices and intended usages of the proposed
mechanisms.

Future revisions of this document will need to include security
considerations as well as IANA considerations. Example messages and
message flows will be more complete. The References section will also
need to be complete.



10. Security Considerations

This revision of this document does not yet address security
considerations. A subsequent revision will do so, and will likely
include the following among items it considers:

Usage of headers such as P-Asserted-Identity which are intended to use
between trusted providers.

Potentially revealing information about subscribers or service
provider infrastructure via signaling messages.


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Security of signaling and bearer.

Implications of inter provider signaling.



11. IANA Considerations

This revision of this document does not yet address IANA
considerations. It is not anticipated that this document will define
any new protocol extensions or other values requiring action of IANA.






































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12. References



12.1. Normative References

   [1]   Rosenberg, et al, J., "SIP: Session Initiation Protocol", RFC
         3261, June 2002.

   [TRKGRP] Gurbani, Jennings, "Representing trunk groups in tel/sip
             Uniform Resource Identifiers (URIs)", draft-ietf-iptel-
             trunk-group-08.txt, October 2006. (work in progress)

   [SIP-IDENT] Peterson, Jennings, "Enhancements for Authenticated
             Identity Management in the Session Initiation Protocol
             (SIP)", RFC 4474, August 2006.

   [PAI]    Jennings, et al, "Private Extensions to the Session
             Initiation Protocol (SIP) for Asserted Identity within
             Trusted Networks", RFC 3325, November 2002.

   [IPTEL-CPC]  Mahy, "The Calling Party's Category tel URI
             Parameter", draft-mahy-iptel-cpc-04.txt, October 2006.
             (work in progress)

   [CPC-SAML]  Schubert, et al, "Conveying Calling Party Category
             (CPC) and Originating Line Information (OLI) using the
             Security Assertion Markup Language (SAML)", draft-
             schubert-sipping-saml-cpc-02.txt, July 2006. (work in
             progress)

    [CONNECTED-ID] Elwell, et al, "Analysis of TISPAN requirements for
             Connected Identity in the Session Initiation Protocol
             SIP)", draft-elwell-sip-tispan-connected-identity-01.txt,
             June 2006. (work in progress)

12.2. Informative References



   [CSI]    Loreto, Terril, "Input 3rd-Generation Partnership Project
             (3GPP) Communications Service Identifiers Requirements on
             the Session Initiation Protocol (SIP)", draft-loreto-
             sipping-3gpp-ics-requirements-00.txt, June 2006. (work in
             progress)




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   [RFC3324] Watson, "Short Term Requirements for Network Asserted
             Identity", RFC 3324, November 2004.

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

   [NETANN] Burger, et al, "Basic Network Media Services with SIP",
             RFC 4240, December 2005.

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

   [3PCC]    Rosenberg, et al, "Best Current Practices for Third Party
             Call Control (3pcc) in the Session Initiation Protocol
             (SIP)", RFC 3725, April 2004.

   [IMS] 3GPP TS 23.228 V7.4.0 (2006-06) - 3rd Generation Partnership
             Project; Technical Specification Group Services and
             System Aspects; IP Multimedia Subsystem (IMS); Stage 2
             (Release 7)





























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Author's Addresses

      John Haluska
      Telcordia Technologies, Inc.
      331 Newman Springs Road
      Room 2Z-323
      Red Bank, NJ  07701-5699
      USA

      Phone: +1 732 758 5735
      Email: jhaluska@telcordia.com


      Renee Berkowitz
      Telcordia Technologies, Inc.
      One Telcordia Drive
      Piscataway, NJ  08854-4157
      USA

      Phone: +1 732 699 4784
      Email: rberkowi@telcordia.com


      Paul Roder
      Telcordia Technologies, Inc.
      One Telcordia Drive
      Room RRC-4A619
      Piscataway, NJ  08854-4157
      USA

      Phone: +1 732 699 6191
      Email: proder2@telcordia.com

      Wesley Downum
      Telcordia Technologies, Inc.
      One Telcordia Drive
      Piscataway, NJ  08854-4157
      USA

      Phone: +1 732 699 6201
      Email: wdownum@telcordia.com


      Richard Ahern
      AT&T Customer Information Services
      1876 Data Drive
      Room 314


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      Hoover, AL  35244
      USA

      Email: Richard.Ahern@bellsouth.com


      Paul Lum Lung
      Qwest Communications International
      1801 California Street
      Suite 1210
      Denver, CO  80202
      USA

      Email: paul.lumlung@qwest.com


     Nicholas S. Costantino
     Soleo Communications, Inc.
     300 Willowbrook Drive
     Fairport, NY 14450

     Email: ncostantino@soleocommunications.com


     Chris Blackwell
     Verizon
     1000 Century Tel Dr
     Room 115
     Wentzville, MO 63385

     Email: chris.blackwell@verizon.com

     Jim Mellinger
     Verizon
     7979 N Beltline Rd
     Irving, TX 75063

     Email: james.j.mellinger@verizon.com

     D. E. Scott
     VoltDelta
     2401 N. Glassell St.
     Orange, CA  92865-2705

     Email: dscott@voltdelta.com


   Intellectual Property Statement


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   The IETF takes no position regarding the validity or scope of any
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   to pertain to the implementation or use of the technology described
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   Information on the procedures with respect to rights in RFC
   documents can be found in BCP 78 and BCP 79.

   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
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   specification can be obtained from the IETF on-line IPR repository
   at http://www.ietf.org/ipr.

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

   Disclaimer of Validity

   This document and the information contained herein are provided on
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   WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY
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   Copyright Statement

   Copyright (C) The IETF Trust (2007).

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

   Funding for the RFC Editor function is currently provided by the
   Internet Society.




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