Network Working Group J. Haluska Internet Draft R. Berkowitz Expires: January 2008 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 Volt Delta July 5, 2007 Considerations for Information Services and Operator Services Using SIP draft-haluska-sipping-directory-assistance-03.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 Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." Haluska, et al. Expires January 5, 2008 [Page 1] Internet-Draft Information Services Using SIP July 2007 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 January 5, 2008. 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. Operator Services are traditional PSTN services which often involve providing human or automated assistance to a caller, and often require the specialized capabilities traditionally provided by an operator services switch. Market and/or regulatory factors in some jurisdictions dictate that some subset of Operator Services continue to be provided going forward. This document aims to identify how Operator and Information 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.......................................8 4. Information Services.........................................12 5. Operator Services............................................15 5.1. Inter Provider Capabilities.............................18 5.2. Inter OISP Capabilities.................................18 5.3. Intra Provider Capabilities.............................19 5.4. Capabilities Required for Specific Services.............20 6. OISP Internal Architecture...................................21 7. General Approach.............................................23 8. Signaling Mechanisms.........................................25 Haluska, et al. Expires January 5, 2008 [Page 2] Internet-Draft Information Services Using SIP July 2007 8.1. Calling Party's Identity................................25 8.2. Provider Identification.................................26 8.2.1. Home Provider......................................27 8.2.2. Intermediate Provider..............................28 8.3. Originating Station Type................................30 8.4. Trunk Group Identifier..................................31 8.5. Dialed Digits...........................................32 8.6. Retargeting to Identify the Desired Service.............33 8.7. Charge Number...........................................33 8.8. Passing Whisper.........................................33 8.9. Calling Equipment Capabilities and Characteristics......37 8.10. Media Server Returning Data to the Application Server..38 8.11. Service Discovery......................................38 9. Example Call Flow - Directory Assistance.....................39 10. Example Call Flow - Operator Services.......................48 11. VoIP Information Services - Summary and Next Steps..........55 12. Security Considerations.....................................56 13. IANA Considerations.........................................56 14. References..................................................57 14.1. Normative References...................................57 14.2. Informative References.................................57 Author's Addresses..............................................59 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 Haluska, et al. Expires January 5, 2008 [Page 3] Internet-Draft Information Services Using SIP July 2007 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. Operator Services are traditional PSTN services which often involve providing human or automated assistance to a caller, and often require the specialized capabilities traditionally provided by an operator services switch. Market and/or regulatory factors in some jurisdictions dictate that some subset of Operator Services continue to be provided going forward. Some examples of such services include collect calls, third party billed calls, and busy line verification. Operator and 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 Operator and Information Services with SIP will require the exchange of certain information, and possibly the use of specialized capabilities which are not normally required 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 will be favored over new extensions. It is intended to provide a Best Current Practices document to facilitate interoperability. 2. Terminology This section defines terms that will be used to discuss Information and Operator Services. Haluska, et al. Expires January 5, 2008 [Page 4] Internet-Draft Information Services Using SIP July 2007 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 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. Haluska, et al. Expires January 5, 2008 [Page 5] Internet-Draft Information Services Using SIP July 2007 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. Intermediate Provider - In the context of this document, an Intermediate Provider is a provider which has agreements with home providers to handle OIS requests, and with OISPs which actually provide the requested services. Note that some home providers will have direct relationships with OISPs, rather than using an Intermediate Provider. Intermediate Providers are the targets of SIP requests from home providers since they are involved when a home provider does not have a direct relationship with an OISP. Intermediate Providers perform retargeting of received SIP requests toward the OISP. 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. 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".Operator Services - Traditional PSTN services which often involve providing human or automated assistance to a caller, and often require the specialized capabilities traditionally provided by an operator services switch. Some examples of such services include collect calls, third party billed calls, and busy line verification. Haluska, et al. Expires January 5, 2008 [Page 6] Internet-Draft Information Services Using SIP July 2007 Retail OIS service - A retail OIS service is one which 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 - In the context of this document, a transit provider simply "moves calls", and has no concept of OIS services. It may perform SIP rerouting of the request, but does not perform SIP retargeting. Such a provider is used when a provider cannot directly route calls to another provider. For example, an Intermediate Provider might use a Transit Provider if for some reason (e.g. error condition) it cannot route a call directly to an OISP. 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 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. 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 Haluska, et al. Expires January 5, 2008 [Page 7] Internet-Draft Information Services Using SIP July 2007 whisper is obtained during the front end automation, and saved as an audio file. Wholesale OIS service -A wholesale OIS service is one which is provided to a user by a Service Provider other than the user's Home Provider. Zero Minus ("0-") Dialing - Invocation of Operator Services by dialing "0" with no further digits. Zero Plus ("0+") Dialing - Invocation of Operator Services by dialing "0" followed by a phone number. 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. 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. Haluska, et al. Expires January 5, 2008 [Page 8] Internet-Draft Information Services Using SIP July 2007 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 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 Haluska, et al. Expires January 5, 2008 [Page 9] Internet-Draft Information Services Using SIP July 2007 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, referred to in this document as an "intermediate provider", and this intermediate provider in turn routes either to the OISP, with which it has a relationship, or there could be multiple intermediate providers. This is illustrated in the following figure: +--------+ +--------+ +---------+ +------+ | Caller | |Home | | Inter- | | OISP | | |----|Provider|---| mediate |---| | | | | | | Provider| | | | | | (A) | | (B) | | (C) | +--------+ +--------+ +---------+ +------+ Figure 3 Services Provided by an Indirect Third Party Provider It must be possible to support the case where transit providers are included between any other providers involved in the call. The transit provider only "moves calls" between other providers, and is involved in no other way with OIS services. This is illustrated in the following figure: +--------+ +--------+ +--------+ +---------+ +------+ | Caller | |Home | |Transit | | Inter- | | OISP | | |----|Provider|---|Provider|---| mediate |---| | | | | | | | | Provider| | | | | | (A) | | (B) | | (C) | | (D) | +--------+ +--------+ +--------+ +---------+ +------+ Figure 4 Involvement of a Transit Provider The following are potential future requirements. Haluska, et al. Expires January 5, 2008 [Page 10] Internet-Draft Information Services Using SIP July 2007 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. 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. Haluska, et al. Expires January 5, 2008 [Page 11] Internet-Draft Information Services Using SIP July 2007 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. Information Services 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 and identifies a set of required high level capabilities. 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 Haluska, et al. Expires January 5, 2008 [Page 12] Internet-Draft Information Services Using SIP July 2007 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. 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 Haluska, et al. Expires January 5, 2008 [Page 13] Internet-Draft Information Services Using SIP July 2007 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. Selection of automated voice platform. When it has been determined that the call must be routed to an automated voice platform, there are a number of factors to be taken into account to determine an appropriate, available platform for the call. It must be possible to determine an appropriate, available automated voice platform to which the call should be routed. 8. Connection of caller to automated voice platform. The OISP must be able to connect the caller to an appropriate automated voice platform. 9. 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. 10. 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." 11. 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. 12. Speech recognition. The OISP must be able to pass the caller's spoken request to speech recognition system, suitable for querying a listing database. Haluska, et al. Expires January 5, 2008 [Page 14] Internet-Draft Information Services Using SIP July 2007 13. Listing database query. The OISP must be capable of querying one or more listings databases using the request. 14. 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. 15. 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 an available, appropriate human operator to which the call should be routed. 16. Routing of call to operator workstation. Once the appropriate operator has been identified, the call must be routed to that operator's workstation. 17. 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. 18. 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. 19. 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. 20. 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. 21. Call completion. If the caller requests call completion, the call should be automatically initiated for the caller. 5. Operator Services Operator Services are traditional PSTN services which often involve providing human or automated assistance to a caller, and often require the specialized capabilities traditionally provided by an operator services switch. Market and/or regulatory factors in some Haluska, et al. Expires January 5, 2008 [Page 15] Internet-Draft Information Services Using SIP July 2007 jurisdictions dictate that some subset of Operator Services continue to be provided going forward. This document assumes an architecture with SIP based OISPs, SIP based home providers, and SIP based end users. Since it is necessary to maintain backward compatibility with traditional TDM based providers and end users, these are also considered. It may not be necessary, desirable, or technically feasible to support every existing Operator Service using SIP, or to support both SIP and TDM based end users for all Operator Services. This is the subject of ongoing investigation, and the current iteration of this document assumes that both SIP and TDM based home providers and end users are in scope for these services, unless specifically indicated to the contrary. A future revision may update this assumption based on the findings of the investigation. With respect to Operator Services, this iteration of this document intends to provide an introduction to and descriptions of some of these services, as well as provide some high level requirements. It is intended that the subsequent iteration will build upon this, providing more detailed requirements, suggested SIP mechanisms, and more call flows. Operator Services are typically provided by the requesting party's OISP. In some cases, such as Busy Line Verification, the target or called party's OISP may be involved as well. Next, several traditional Operator Services will be described. As indicated above, the current iteration of this document is silent regarding which of these may or may not be candidates for implementation with SIP, or towards SIP end users. Note that unless specifically indicated, most of these services are traditionally provided by the caller's OISP. Operator Assistance. This allows the caller to perform either "zero minus" or "zero plus" dialing to be connected to a human or automated system for assistance with the call. Collect calls. This allows the caller to request that the called party accept the charges for the call. Typically an OISP utilizes a human operator or automated system to provide this service. Rate Quotes. This allows the caller to request a quote for the cost or rate for specific calls. Haluska, et al. Expires January 5, 2008 [Page 16] Internet-Draft Information Services Using SIP July 2007 Third party billed calls. This allows the caller to request that a third party (different than the calling or called party) be contacted and requested to accept charges for the call (although in some limited cases, contacting the third party is not necessary). Busy Line Verification and Interrupt. This allows a caller to have the OISP determine whether a target line is in use, and if so, to "barge in" to the conversation and request whether the target party is willing to accept a call from the caller. This service is initially handled by the caller's OISP, which then contacts the target party's OISP, which is able to perform the verification and interrupt on the target party. Coin Calls. Operator services systems must be able to control TDM- based network controlled coin stations (payphones). This includes monitoring of coin deposit tones (to verify payment) sent from the coin station, as well as sending supervision (control) signals to the coin station. Network controlled coin stations are connected to TDM based end offices via specialized phone lines which support the required signaling. These end offices, in turn, connect to TDM based OISPs using specialized trunks capable of conveying the coin signaling. The OISP monitors and controls the coin station via these trunks. "Smart" coin stations perform coin detection locally and do not require network control, and are not discussed here. This service is provided by the OISP associated with the coin station. Emergency Calls. Sometimes a caller dials "0" instead of the standard emergency dialstring, resulting in placement of an emergency call to the OISP. The OISP must properly route such a call toward the PSAP. This service is provided by the caller's OISP. Calling Card Billing Service. This enables a calling party to bill a call to a calling card number. Commercial Credit Card Billing Service. This enables a calling party to bill a call to a commercial credit card. Directory Assistance (DA). In some contexts, DA is considered as an Operator Service. Within the context of this document, we consider DA as an Information Service, which is related to but distinct from Operator Services. Haluska, et al. Expires January 5, 2008 [Page 17] Internet-Draft Information Services Using SIP July 2007 The following sections describe an initial set of basic high level capabilities required for supporting Operator Services. The capabilities for Information Services generally apply for Operator Services as well. This work is currently under study, and a complete set of required capabilities is expected to be identified in the near future. Similarly to the required capabilities for Information Services, the use of existing SIP mechanisms will be investigated for providing these capabilities. 5.1. Inter Provider Capabilities Ability to accept requests from other providers. This is the ability to accept incoming OIS requests from other providers, including home providers, intermediate providers, and transit providers. Ability to terminate calls to other providers. This applies to call completion services, as well as other services such as third party billing. 5.2. Inter OISP Capabilities These are capabilities between OISPs. Inward connection. This is a call from one OISP to another, e.g. so that the originating OISP may request services from the terminating OISP. One example of this is Busy Line Verification, where the caller calls their own OISP, and this OISP places an "inward" call to the target party's OISP, which would have the capability to perform the verification of the target party. Transfer between OISPs. In this case, one OISP transfers the call to another OISP, to be handled by that OISP. Moving connection from one OISP to another. An example of this case is where one OISP farms out a specific service to another OISP. The first OISP performs initial handling of the call, to determine the desired service. The call is sent to a different OISP with which the first has a relationship. The first OISP remains in the signaling path, and when the provided service is complete, the first OISP determines what if any additional processing may be necessary. This is similar to a third party call control type arrangement. Haluska, et al. Expires January 5, 2008 [Page 18] Internet-Draft Information Services Using SIP July 2007 5.3. Intra OISP Capabilities Note that some of the following capabilities may be required for inter OISP scenarios as well; this is the subject of ongoing analysis and is not covered in the current iteration of this document. Placing a caller on hold, possibly with announcements. This is used in many services, including Information Services. Exchanging information between Application Server and Operator Workstations/Automated Platforms. This capability is required whenever an operator workstation or automated platform is used. Because an Operator Workstation interacts with a human user, it is expected that additional information, beyond that which an automated system would exchange with an application server, will be required. Further, several modes of application server control are currently under investigation. The first is where the workstation or automated platform is more or less autonomous, and is capable of initiating calls and directly impacting call processing. The other is more of a master-slave relationship, where the AS is in complete control. The master-slave model requires that more information be exchanged with the AS than does the autonomous model. Queuing and call distribution. Resources including human operators and automated platforms need to be tracked and managed, and the appropriate resource of the appropriate type needs to be selected on a per invocation basis. What is needed is that for a particular call, that a set of criteria be provided and the best match resource be selected. This is the job of the ACD server. Some means is needed to communicate the selection criteria for human operators and automated platforms to the ACD server. Operator Registration and Location. Human operators may not be interchangeable, and have specific attributes such as skillsets which can be used to identify the best human operator to service a particular call. Operators log in at workstations at the beginning of a shift, and log out during breaks and at the end of a shift. It is important to associate each available operator with the workstation at which they are logged in, so that requests can be sent to the appropriate human operator. This is needed because the selection process described above identifies a particular human operator; it is then necessary to identify the workstation at which that operator can be reached. Haluska, et al. Expires January 5, 2008 [Page 19] Internet-Draft Information Services Using SIP July 2007 Bridging and removal of operator or automated system. Many operator services require that either a human operator or automated system be "bridged" onto a call, and to be removed at some point. 5.4. Capabilities Required for Specific Services. Connection Hold and Ringback. This capability involves having the OISP "hold" the connection, such that the originating caller remains connected, even if they attempt to hang up. This is mainly used in relation to emergency services. Ringback is the ability for the OISP to call back the calling party after they have hung up. This too is often used in conjunction with emergency calls. Coin Station Control. This is the ability of the OISP to determine the coinage deposited into a TDM based network controlled coin station (as opposed to an "intelligent" coin station which performs such functions locally). This involves interpretation of the coin control signals sent via specialized trunks from the end office to which the TDM based coin station is connected via a specialized phone line. Additionally, the need to signal toward the coin station needs to be supported as well, for example to instruct the station to return coins to the caller. This capability is intended to interact with the specialized coin trunk. Network Service Recall. After a call resulting from Operator Services is completed, the caller may signal the desire to return back to the OISP, without placing another call. In the traditional PSTN, this is typically accomplished by the user signaling a hook flash or other distinctive stimulus. Verification and Interrupt. This is used in the Busy Line Verification and Interrupt service, and allows the OISP to determine if the target number is in use, to listen to a scrambled representation of the conversation, and to interrupt the target party's conversation to ask if they would accept a call from the caller. Transfer of emergency services call to selective router. Sometimes a caller places an emergency call using a dial string which invokes operator assistance (such as "0" in North America), rather than an emergency call dial string. In such cases, the OISP must be able to pass the emergency call to the appropriate PSAP. Haluska, et al. Expires January 5, 2008 [Page 20] Internet-Draft Information Services Using SIP July 2007 6. 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. Similar functionality is required with respect to automated platforms. The ACD server is modeled as an Application Server. There is currently work in the MEDIACTL working group regarding Media Resource Brokers (MRBs) which may be relevant to this. 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. Haluska, et al. Expires January 5, 2008 [Page 21] Internet-Draft Information Services Using SIP July 2007 Operator Workstations - Operator Workstations provide an interface to the human operator. They may receive the customer's recorded request (e.g. name and city of requested listing), information from listing or other databases, and also terminate a multimedia session with the customer. Operator workstations are specialized SIP endpoints, and may be modeled in various ways, such as UAs or media servers. PSTN Gateways - OISPsneed to interface with the PSTN. Thus, gateways are needed to interface between the OISP and both signaling and bearer. The bearer is handled by trunk gateways, which interface RTP streams on the OISP side to TDM trunks on the PSTN side. The signaling may be handled by signaling gateways which interface SS7 on the PSTN side and SIP on the OISP side. Currently in North America, inband signaling on MF trunks is common for interfacing to OISPs, and trunk gateways need to be able to interpret and report as well as generate the appropriate MF signaling. 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. Haluska, et al. Expires January 5, 2008 [Page 22] Internet-Draft Information Services Using SIP July 2007 +-------+ +--------+ +---------+ +---------+ | Proxy |----| OIS-AS |-+-| OIS-MS1 |...| OIS-MSk | +-------+ +--+-----+ | +---------+ +---------+ | | | +---------+ +---------+ | +-| OWS1 |...| OWSk | +--+--+ | +---------+ +---------+ | ACD | | +-----+ | +--+---+ +-|PSTNGW| +------+ Figure 5 Simplified view of OISP Internal Architecture 7. General Approach This section describes one possible way to implement DA using SIP. Other ways may be possible. 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 Haluska, et al. Expires January 5, 2008 [Page 23] Internet-Draft Information Services Using SIP July 2007 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. 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). Haluska, et al. Expires January 5, 2008 [Page 24] Internet-Draft Information Services Using SIP July 2007 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. 8. Signaling Mechanisms This section discusses the signaling mechanisms required to provide OIS services. 8.1. 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 form of a SIP URI might be needed so that the identity of the home provider 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], Haluska, et al. Expires January 5, 2008 [Page 25] Internet-Draft Information Services Using SIP July 2007 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 provider, 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. 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: ;tag=1234567 To: 411 Contact: P-Asserted-Identity: "732758123" Content-Type: application/sdp Content-Length: ... [SDP not shown] 8.2. 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. Haluska, et al. Expires January 5, 2008 [Page 26] Internet-Draft Information Services Using SIP July 2007 8.2.1. 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 provider 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 use of such standardized mechanisms for reaching its users. It also becomes difficult to derive the home provider 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 provider 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 provider 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 provider's identity is consequently also not available, and call processing logic based on such information (such as branding) cannot take place. Haluska, et al. Expires January 5, 2008 [Page 27] Internet-Draft Information Services Using SIP July 2007 8.2.2. Intermediate Provider In some cases, the OISP may need to know the identity of an intermediate provider which the call has traversed. Recall that for our purposes, we define "intermediate provider" as having a business relationship with both the home provider (to handle OIS requests) and with an OISP (which will actually provide the requested service.) 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. Haluska, et al. Expires January 5, 2008 [Page 28] Internet-Draft Information Services Using SIP July 2007 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 6 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: ; 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: ; index=1, ; index=2 (other message content omitted) Haluska, et al. Expires January 5, 2008 [Page 29] Internet-Draft Information Services Using SIP July 2007 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 Please note that if a transit provider were encountered, the transit provider would simply route the request toward Provider C, and would not perform retargeting. It would not modify the Request-URI nor the SIP History-Info header contents. 8.3. 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. It is also desirable to characterize certain types of originating SIP based callers using these same values, e.g. prison, police, etc. 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-06.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: Haluska, et al. Expires January 5, 2008 [Page 30] Internet-Draft Information Services Using SIP July 2007 INVITE sip:bob@biloxi.example.com SIP/2.0 To: "Bob" From: ;tag=1928301774 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. 8.4. 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. RFC 4904, "Representing trunk groups in tel/sip Uniform Resource Identifiers (URIs)" 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: Haluska, et al. Expires January 5, 2008 [Page 31] Internet-Draft Information Services Using SIP July 2007 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: ;tag=1234567 To: 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" Content-Type: application/sdp Content-Length: ... 8.5. 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: ; index=1, ; 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. Haluska, et al. Expires January 5, 2008 [Page 32] Internet-Draft Information Services Using SIP July 2007 8.6. 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. 8.7. 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. There is currently no IETF standardized mechanism to convey the Charge Number in SIP. This is currently under investigation. The need to convey equivalent information for SIP based callers is also under investigation. 8.8. 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 Haluska, et al. Expires January 5, 2008 [Page 33] Internet-Draft Information Services Using SIP July 2007 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. What follows is a description of one possible way to implement this. The work of the recently formed IETF MEDIACTRL working group may provide alternatives. 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. Haluska, et al. Expires January 5, 2008 [Page 34] Internet-Draft Information Services Using SIP July 2007 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 7 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] Haluska, et al. Expires January 5, 2008 [Page 35] Internet-Draft Information Services Using SIP July 2007 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"; Content-Type: audio/x-wav Content-Disposition: render 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"; Content-Type: audio/x-wav Content-Disposition: render 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. Haluska, et al. Expires January 5, 2008 [Page 36] Internet-Draft Information Services Using SIP July 2007 8.9. 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: ;tag=1234567 To: 411 Contact: ;audio;video;text ;actor="principle";automata;mobility="fixed" ;methods="INVITE,BYE,OPTIONS,ACK,CANCEL" P-Asserted-Identity: "7327581234" P-Asserted-Identity: tel:+7327581234 Content-Type: application/sdp Content-Length: ... [SDP not shown] Haluska, et al. Expires January 5, 2008 [Page 37] Internet-Draft Information Services Using SIP July 2007 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. 8.10. 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. The topic of application server control of media services is currently under study, and is the subject of the IETF MEDIACTRL working group's efforts. This information can also be conveyed using non SIP mechanisms. Describing such mechanisms is out of the scope of this document. 8.11. 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 Haluska, et al. Expires January 5, 2008 [Page 38] Internet-Draft Information Services Using SIP July 2007 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 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. 9. Example Call Flow - Directory Assistance 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. Haluska, et al. Expires January 5, 2008 [Page 39] Internet-Draft Information Services Using SIP July 2007 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 8 DA Call flow, part 1 For brevity, only relevant SIP headers will be shown. The following test refers to Figure 8. Haluska, et al. Expires January 5, 2008 [Page 40] Internet-Draft Information Services Using SIP July 2007 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: ;tag=1234567 To: 411 Contact: Content-Type: application/sdp Content-Length: ... The home provider 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. Haluska, et al. Expires January 5, 2008 [Page 41] Internet-Draft Information Services Using SIP July 2007 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: ;tag=1234567 To: 411 Contact: P-Asserted-Identity: "732758123" P-Asserted-Identity: tel:+7327581234 History-Info: ; 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. Haluska, et al. Expires January 5, 2008 [Page 42] Internet-Draft Information Services Using SIP July 2007 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: ;tag=1234567 To: 411 Contact: P-Asserted-Identity: "732758123" P-Asserted-Identity: tel:+7327581234 History-Info: ; index=1, ; 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 provider, 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. Haluska, et al. Expires January 5, 2008 [Page 43] Internet-Draft Information Services Using SIP July 2007 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: ;tag=1234567 To: sip:dialog@ois-as.prov-c.com; \ voicexml=http://vxmlserver.example.net/cgi-bin/script.vxml Contact: 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. Haluska, et al. Expires January 5, 2008 [Page 44] Internet-Draft Information Services Using SIP July 2007 Caller OIS-AS OIS-MS1 ACD OWS | | | | | | | | | | | | | | | |(11) RTP session | | | |...................| | | | | | | | | |(12) INFO w.URI, body | | |<--------| | | | | | | | | |(13) INVITE | | | |------------------>| | | | | | | | |(14) 3xx | | | | |<------------------| | | | | | | | |(15) INVITE | | | |---------------------------->| | | | | | | |(16) 200 OK | | | |<----------------------------| | | | | | | |(17) BYE | | | | |-------->| | | | | | | | | |(18) 200 OK | | | |<--------| | | | | | | | |(19) re INVITE | | | |<--------| | | | | | | | | |(20) 200 OK | | | |-------->| | | | | | | | | |(21) RTP session | | | |.......................................| | | | | | | |(22) BYE | | | | |<----------------------------| | | | | | | |(23) 200 OK | | | |---------------------------->| | | | | | | | | | | | | | | | Haluska, et al. Expires January 5, 2008 [Page 45] Internet-Draft Information Services Using SIP July 2007 Figure 9 DA Call flow, part 2 The following text refers to Figure 9. 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 "whisper" file, or condensed version of the request. In this example, the MS is unable to successfully perform the query, so it sends an indication of this to the AS. In this example, the indication is sent using an as yet unspecified protocol message carried in a message body in a SIP INFO message, which also carries a URI which points to the whisper file. Other mechanisms, including non SIP mechanisms, could also be used to this end (this is the subject of further study). The AS allows the caller to remain connected to the MS while it sets up a call to an operator workstation (OWS), allowing for the possibility to play custom announcements to the caller. 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. This INVITE carries information about the required characteristics, such as language and skill set, of the operator which should be selected for this call. The means by which this information is carried has yet to be defined. 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. The Media Resource Broker being discussed in the IETF MEDIACTRL working group also represents an approach to ACD. 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 its SDP, which the OIS-AS sends toward the caller in a re-INVITE (19). Only when the workstation has sent a final response to the INVITE, the AS sends a BYE (17) to the MS. The caller is now connected to the operator (21), and the operator helps the caller with the listing. The operator workstation launches a query, and a response is received. The operator signals Haluska, et al. Expires January 5, 2008 [Page 46] Internet-Draft Information Services Using SIP July 2007 a BYE (22) 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. Caller OIS-AS OIS-MS2 | | | | | | | | | | |(24) INVITE | |-------->| | | | | |(25) 200 OK | |<--------| | | | |(26) re INVITE | |<--------| | | | | |(27) 200 OK | |-------->| | | | | |(28) RTP session | |...................| | | | | |(29) BYE w.body | |<--------| | | | |(30) REFER | |<--------| | | | | | | | | | | Figure 10DA Call flow, part 3 Haluska, et al. Expires January 5, 2008 [Page 47] Internet-Draft Information Services Using SIP July 2007 The following text refers to Figure 10. The OIS-AS sends an INVITE (24) to another OIS-MS, MS2, for back end automation. When it receives MS2's SDP in the 200 OK (25), it sends a re INVITE (26) 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 (28). The MS plays the listing information, and offers call completion service. The caller accepts, so OIS-MS2 sends a BYE (29) with a message body containing the result of the call completion offer. Since call completion was requested, the OIS-AS sends a REFER (30) to the caller, to cause it to place a call to the listed party. The OIS-AS may or may not care about subsequent NOTIFYs from the caller, and drops out of the call. 10. Example Call Flow - Operator Services The following call flow provides examples of how a specific operator service, Operator Assisted Collect Call, could be implemented using the mechanisms described in this document. The purpose is to illustrate one way to implement this service using the proposed signaling mechanisms. In practice, this particular service could be implemented in an automated fashion without human intervention. Haluska, et al. Expires January 5, 2008 [Page 48] Internet-Draft Information Services Using SIP July 2007 Caller Proxy AS WS Called MS ACD | | | | | | | | | | | | | | | | | | | | | |[ Caller dials 0+ or 0- ] | | | | | | | | | | | | | | | | | | |(1) INVITE | | | | | |------------------>| | | | | | | | | | | | |(2) 100 Trying | | | | | |<------------------| | | | | | | | | | | | |[ AS determines operator is needed, performs 3PCC ] | | | | | | | | | | | | | | | | | |(3) INVITE | | | | | |-------------------------------------->| | | | | | | | | | |(4) 3xx redirect to WS where selected operator registered | | |<--------------------------------------| | | | | | | | | | |(5) INVITE | | | | | |-------->| | | | | | | | | | | | | |(6) 200 OK | | | | | |<--------| | | | | | | | | | | | | |(7) ACK | | | | | | |-------->| | | | | | | | | | | |(8) 200 OK | | | | | |<------------------| | | | | | | | | | | | |(9) ACK | | | | | | |------------------>| | | | | | | | | | | | |[ Caller is now connected to operator ]| | | | | | | | | | | | | | | | | |(10) RTP | | | | | | |.............................| | | | | | | | | | | |[ Operator determines calling's request and places on hold]| | | | | | | | Haluska, et al. Expires January 5, 2008 [Page 49] Internet-Draft Information Services Using SIP July 2007 | | | | | | | | | |(11) re-INVITE (HOLD) | | | | |<--------| | | | | | | | | | | | | |(12) 200 OK | | | | | |-------->| | | | | | | | | | | | | |(13) ACK | | | | | | |<--------| | | | | | | | | | | |[ AS logic determines announcements should be played instead ] | | | | | | | | | | | | | | | | |(14) INVITE (play announcements) | | | |---------------------------->| | | | | | | | | | | |(15) 200 OK (SDP-MS) | | | | |<----------------------------| | | | | | | | | | | |(16) ACK | | | | | | |---------------------------->| | | | | | | | | |(17) reINVITE (SDP-MS) | | | | |<------------------| | | | | | | | | | | | |(18) 200 OK (SDP-calling) | | | | |------------------>| | | | | | | | | | | | |(19) ACK | | | | | | |<------------------| | | | | | | | | | | | |(20) RTP (MS plays announcements to calling) | | |.................................................| | | | | | | | | |[ Operator calls called party ] | | | | | | | | | | | | | | | | | | | | |(21) INVITE | | | | | |-------->| | | | | | | | | | | | | |(22) 200 OK | | | | | |<--------| | | | | | | | | | | | | |(23) ACK | | | | | | |-------->| | | | | | | | | | | | | |(24) RTP | | | Haluska, et al. Expires January 5, 2008 [Page 50] Internet-Draft Information Services Using SIP July 2007 | | | |.........| | | | | | | | | | |[ Called agrees to accept charges ] | | | | | | | | | | | | | | | | | |[ Operator takes Calling off hold ] | | | | | | | | | | | | | | | | | | | |(25) re-INVITE (un-HOLD) | | | | |<--------| | | | | | | | | | | |[ AS logic directs Calling from MS back to WS ] | | | | | | | | | | | | | | | | |(26) re-INVITE (SDP-ws) | | | | |<------------------| | | | | | | | | | | | |(27) 200 OK | | | | | |------------------>| | | | | | | | | | | | | | |(28) 200 OK | | | | | |-------->| | | | | | | | | | | | | |(29) ACK | | | | | | |<--------| | | | | | | | | | | |(30) ACK | | | | | | |<------------------| | | | | | | | | | | | |(31) RTP | | | | | | |.............................| | | | | | | | | | | |[ AS releases MS ] | | | | | | | | | | | | | | | | | | | | | |(32) BYE | | | | | | |---------------------------->| | | | | | | | | | | |(33) 200 OK | | | | | |<----------------------------| | | | | | | | | |[ Calling and Called both have RTP session with WS ] | | | | | | | | | | | | | | | |[ WS bridges conversations together internally ] | | | | | | | | | | | | | | | | Haluska, et al. Expires January 5, 2008 [Page 51] Internet-Draft Information Services Using SIP July 2007 |[ After brief interlude WS transfers Calling directly to Called ] | | | | | | | | | | | | | | |[ then drops out ] | | | | | | | | | | | | | | | | | | | | | |(34) REFER (to called) | | | | |<--------| | | | | | | | | | | | | |(35) 202 Accepted | | | | | |-------->| | | | | | | | | | | | | |(36) NOTIFY (trying) | | | | |-------->| | | | | | | | | | | | | |(37) 200 OK | | | | | |<--------| | | | | | | | | | | |[ Replaces header causes Called to replace old call with new ] | | | | | | | | | | | | | | | | |(38) INVITE (replaces: WS ) | | | | |------------------>| | | | | | | | | | | | |(39) 200 OK (SDP-called) | | | | |<------------------| | | | | | | | | | | | |(40) ACK | | | | | | |------------------>| | | | | | | | | | | | | |(41) BYE | | | | | | |<--------| | | | | | | | | | | | | |(42) 200 OK | | | | | |-------->| | | | | | | | | | |[ The following interactions synch up SDP - optimization possible ] | | | | | | | | | | | | | | |(43) re-INVITE (SDP-called) | | | | |<------------------| | | | | | | | | | | | |(44) 200 OK (SDP-calling) | | | | |------------------>| | | | | | | | | | | | Haluska, et al. Expires January 5, 2008 [Page 52] Internet-Draft Information Services Using SIP July 2007 |(45) ACK | | | | | | |<------------------| | | | | | | | | | | | | | |(46) re-INVITE (SDP-calling) | | | | |------------------>| | | | | | | | | | | | |(47) 200 OK | | | | | |<------------------| | | | | | | | | | | | |(48) ACK | | | | | | |------------------>| | | | | | | | | | |(49) RTP | | | | | | |.......................................| | | | | | | | | | |[ Calling and Called are now talking directly ] | | | | | | | | | | | | | | | | | | |(50) NOTIFY (Call completed) | | | | |-------->| | | | | | | | | | | | | |(51) 200 OK | | | | | |<--------| | | | | | | | | | | |[ Operator now drops out ] | | | | | | | | | | | | | | | | | | | | | |(52) BYE | | | | | | |-------->| | | | | | | | | | | | | |(53) 200 OK | | | | | |<--------| | | | | | | | | | |[ AS remains in signaling path until call ends ] | | | | | | | | | | | | | | | | |(54) BYE | | | | | | |------------------>| | | | | | | | | | | | | | |(55) BYE | | | | | | |------------------>| | | | | | | | | | | | |(56) 200 OK | | | | | |<------------------| | | | | | | | | | |(57) 200 OK | | | | | |<------------------| | | | | Haluska, et al. Expires January 5, 2008 [Page 53] Internet-Draft Information Services Using SIP July 2007 | | | | | | | | | | | | | | | | | | | | | Figure 11 Operator Service Call flow (Operator Assisted Collect Call) The caller initiates the call by dialing 0+ or 0-. The call is routed to the AS. The AS examines the calling party number and calling party's home provider, which are derived from the P-Asserted-Identity header. The charge number is also needed, in case the caller's service is determined by agreements with another party, such as the caller's employer. The employer may have a large number of calling identities representing its employees, which are covered under its agreement with the OISP. Rather than provision every possible calling number/identity with the OISP (and this may be constantly changing), the ability to pass a charge number would allow the OISP to determine whether this charge number has any associated treatment on a per charge number basis. In any case, in our example, the AS examines the request and determines that the call is for an operator assisted collect call. Typically a MS could be initially connected to prompt the user for the type of call. This step is omitted in this example. The AS performs third party call control (3PCC). It sends a 18x response towards the caller. It needs to initiate a call leg to an operator workstation. It populates the selection criteria in an INVITE message (the exact mechanism for this is under study) which it sends to the ACD server in (3). The ACD server identifies the best match available operator and returns the contact information for the workstation where that operator is currently registered in a 3xx redirection response. In (5) the AS sets up a call to the workstation identified by the ACD server, and using 3PCC connects the caller to the WS, resulting in an RTP session in (10). The operator determines the caller's requested number, and sends an INVITE toward the AS to put the caller on hold. The logic in the AS determines that instead the caller should be connected to custom announcements, and in (14) through (20) creates a session between the caller and a MS. Haluska, et al. Expires January 5, 2008 [Page 54] Internet-Draft Information Services Using SIP July 2007 Meanwhile, in (21) the WS places a call to the called party, and asks whether the called party would accept the charges for a collect call. In this example, the called party agrees to the request. In (25), the operator takes the caller off hold (recall that it believes it has placed the caller on hold). The AS, in (26) through (33), performs 3PCC, and removes the caller from the MS which is playing custom announcements, and reconnects the caller back to the WS. The WS uses its own internal bridging functionality to conference the operator, calling, and called parties. After a brief interlude, the operator initiates a transfer of the calling and called parties directly together using a REFER in (34) through (37). The AS, performing 3PCC, utilizes the SIP Replaces mechanism beginning in (38) to complete the transfer. When the transfer is complete, the operator drops out completely. Note that the AS, performing 3PCC, remains in the signaling path until the call is torn down in (54) through (57). 11. 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. An initial set of required capabilities for Operator Services has been identified. The next revision of this document is expected to contain a complete set of required capabilities. Once such a set is identified, existing SIP mechanisms will be examined for supporting these capabilities. 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. Haluska, et al. Expires January 5, 2008 [Page 55] Internet-Draft Information Services Using SIP July 2007 12. 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. Security of signaling and bearer. Implications of inter provider signaling. 13. 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. Haluska, et al. Expires January 5, 2008 [Page 56] Internet-Draft Information Services Using SIP July 2007 14. References 14.1. 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) 14.2. 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) Haluska, et al. Expires January 5, 2008 [Page 57] Internet-Draft Information Services Using SIP July 2007 [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) Haluska, et al. Expires January 5, 2008 [Page 58] Internet-Draft Information Services Using SIP July 2007 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 Haluska, et al. Expires January 5, 2008 [Page 59] Internet-Draft Information Services Using SIP July 2007 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 Haluska, et al. 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Acknowledgment Funding for the RFC Editor function is currently provided by the Internet Society. Haluska, et al. Expires January 5, 2008 [Page 61]