Forecasts show that speech-driven services will play an important role on the 3G market. People want the ability to access information while on the move and the small portable mobile devices that will be used to access this information need improved user interfaces using speech input. At present, however, the complexity of medium and large vocabulary speech recognition systems is beyond the memory and computational resources of such devices. Also associated delay to download speech data files (e.g. grammars, acoustic models, language models, vocabularies…) may be prohibitive. Eventually, it may not always be acceptable for the speech service providers to allow download of these speech data files if they contained confidential information (password (security issue), customer names and address (privacy issue)) or intellectual properties; for example a well crafted speech grammar is often considered by speech service providers as a trade secret. Server-side processing of the combined speech and DTMF input and speech output can overcome these constraints by taking full advantage of memory and processing power as well as specialized speech engines and data files. However, the distortions introduced by the encoding used to send the audio between the client and the server as well as additional network errors can degrade the performance of the speech engines; therefore also limiting the achievable speech functionalities. A server-side speech service is generally equivalent to a phone call to an automatic service. As for any other telephony service, DTMF is a feature that should always be considered as needed. This document describes a generic speech recognition framework to distribute the audio sub-system and the speech services by sending encoded speech and meta-information between the client and the server. Instead of using a voice channel as in today's server-based speech services, an error-protected data channel will be used to transport encoded speech from the client audio sub-system (terminal client) to remote speech engines (on server) for processing (e.g. speech recognition, speaker recognition,). The speech recognition framework will also enable downlink data streaming of voice and recorded audio prompt generated by server to the terminal client audio subsystem. The speech recognition framework may use conventional codecs like AMR or Distributed Speech Recognition (DSR) optimized codecs. The speech recognition framework will provide users with a high performance distributed speech interface to server-based automatic speech services with communication, information access or transactional purposes. The types of supported user interfaces include those that are voice only, for example, automatic speech access to information, such as a voice portal described in this section. These typically support combined speech or DTMF input. In the future, a new range of multi-modal applications is also envisaged incorporating different modes of input (e.g. speech, keyboard, pen) and speech and visual output.

The present document defines the stage one description of the Speech Recognition Framework for Automated Voice Services. Stage One is the set of requirements for data seen primarily from the user's and service providers' points of view. This Technical Specification includes information applicable to network operators, service providers, terminal and network manufacturers. This Technical Specification contains the core requirements for the Speech Recognition Framework for automated voice services. The scope of this Stage 1 is to identify the requirements for 3G networks to support the deployments of a speech recognition framework - based automated voice services and therefore to introduce a 3GPP speech recognition framework as part of speech-enabled services. The Speech Recognition Framework for automated voice services is an optional feature in a 3GPP system. Figure 1 positions the Speech recognition Framework (SRF) with respect to other speech-enabled services as discussed in [6]. As illustrated, SRF is designed to support server-side speech recognition over packet switched network (e.g. IMS). As such SRF also enable configurations of multimodal and multi-device services that include distribute the speech engines. Note that it is possible to design speech-enabled services that alternate or combine the use of client-side only engines and SRF.

The following documents contain provisions which, through reference in this text, constitute provisions of the present document.

• References are either specific (identified by date of publication, edition number, version number, etc.) or non specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same Release as the present document.

A 3GPP speech recognition framework enables the use of conventional codecs (e.g. AMR) or DSR optimized codecs to distribute in the network the speech engines that process speech input or generate speech output. It includes:

• Default uplink and downlink codec specifications.
• A stack of speech recognition protocols to support:
• Establishment of uplink and downlink sessions, along with codec negotiation
• Transport of speech recognition payload (uplink) with conversational QoS
• Support of transport (also at conversational QoS) of meta-information required for the deployment of speech recognition applications between the terminal and speech engines (meta-information may include terminal events and settings, audio sub-system events, parameters and settings, etc.).

IMS provides a protocol stack (e.g. SIP/SDP, RTP and QoS), that may advantageously be used to implement such capabilities. It shall be possible to recommend a codec to be supported by default to deploy services that rely on the 3GPP speech recognition framework. To that effect, the specifications will consider either conventional speech codecs (e.g. AMR) or DSR optimized codecs. ETSI has published DSR optimized codecs specifications (ETSI ES 201 108 [7] and ETSI ES 202 050 [10]) and a payload format for transport of DSR data over RTP (IETF AVT DSR). The following list gives the high level requirements for the SRF-based automated voice services: .

• Users of the SRF-based automated voice service shall be able to initiate voice communication, access information or conduct transactions by voice commands using speech recognition. Examples of SRF-based automated voice services are provided in Appendix A.

The speech recognition framework for automated voice services will be offered by the network operators and will bring value to the network operator by the ability to charge for the SRF-based automated voice services. This service may be offered over a packet switched network; however in general this requires specification of a complete protocol stack. When this service is offered over the IMS, the protocols used for the meta information and front-end parameters (from terminal to server) and associated control and application specific information can and shall be based on those in IMS.

In addition to the capabilities required for IMS Basic Voice session (such as the default voice codec that will be used for the downlink audio prompt stream), the following SRF-based automated voice service-specific capabilities shall be required in the UE and network:

• A default uplink codec (conventional codec or DSR optimized codec).
• A downlink conventional codec and downlink streaming capabilities (simultaneous with uplink).
• The capability to transmit keypad information from the client to the server (e.g., either DTMF or the keypad string).
• The capability to reconstruct encoded speech. The reconstruction requirement does not apply to the UE.

It shall be possible to enable application specific information exchanges between the client and the server (e.g. client events (e.g. barge-in events), display information, etc…), in the form of speech meta-information. It shall be possible to enable these exchanges with conversational QoS. SRF shall be supported by an uplink channel available in GERAN and UTRAN networks for the transport of the codec payload and with QoS (Quality of Service) for conversational class, streaming and interactive QoS services as specified in TS 22.105. It shall be possible for the network to distinguish a SRF session from a basic voice session (e.g. for charging purposes).

SRF-based automated voice services may be provided by the network operator (home or visited) or by third parties. See appendix A for examples of such services. The administration of the SRF-based automated voice services will be under the control of the network operator. But when decided to do so by the network operator, it should be possible to the third party providers to administer the SRF-based automated voice services them selves through the gateway that they would connect to IMS. In such case, the third party provider performs all the administrative steps and no registration would be required with the network operator.

The SRF-based automated voice services shall be able to be provisioned by either the network operator (roaming or home) or by a 3rd party service provider. It shall be possible for network operators and 3rd party service providers to offer SRF-based automatic voice services by providing identity of service, such as a phone number, an IP address or a URI that the user can enter or select on the terminal.

The "Security Threats and Requirements" specified in TS 21.133 shall not be compromised. It shall be possible to deny unauthorized access to 3GPP SRF-based automated voice services. An authorization may be based on the following,

• identity of the accessing user agent, server or device
• the destination user, device or user agent

Third parties shall have authorization from the User and PLMN Operators in order to access 3GPP SRF-based automated voice services. It shall be possible to reconstitute PCM samples from DSR packets so that the user's spoken command can be transcribed at a later time, if required.

For SRF-based automated voice services, privacy requirements shall be at least as good as for IMS voice or data sessions [5]:

• It shall be possible to encrypt speech and speech meta-data exchanges;
• It shall be possible to prevent exchange of the user's true identity, location and other terminal or user related information when required.

SRF-based automated voice services, may imply that the service provider collects information about the user or usage. This information should be treated according to the policies in place for data and voice (e.g. human to operator or human to automated service) services. The SRF-based automated voice services shall not add additional privacy risks.

The user can be charged for sessions with SRF-based automated voice services in a variety of ways. The following shall be possible:

1. By duration of session (including "one-off" charge/flat rate)
2. By data volume transferred (number of packets) or other similar criteria.
3. By subscription fees for the service (unlimited usage or unlimited usage up to a point and then per-use fees)
4. Free (e.g. with the service being subsidised by advertising revenue from advertisement spots). The advertisement spots may be inserted either at session start-up or close, or designed in such that system delay time is masked (e.g., while the user is waiting for the flight schedules to be returned, or a purchase transaction to be completed). The network operator will receive revenue from users directly as well as from the content and service providers who want their sites to be accessible via the automated voice service, and from advertisers. Advertising spots can be inserted at appropriate points during the session (e.g., at the beginning of the session, while the user is waiting for a system response, or at the end of a session).

SRF-based automated voice services shall be available to pre-paid and post-paid subscribers.

The user shall be able to utilize SRF-based automated voice services when roaming in any IMS compatible mobile network. The capabilities of the SRF-based automated voice service shall be available in the roamed-to network in the same manner as in the home network, within the limitation of the capabilities of the serving network.

No interaction with other services identified. When connected to the IMS, other IMS services are available to the user through the terminal. Other services (non IMS voice etc..) may be available with or without disconnecting from the IMS.

Examples of Automated Voice Services include:

• Communication assistance (Name dialling, Service Portal, Directory assistance)
• Information retrieval (e.g., obtaining stock-quotes, checking local weather reports, flight schedules, movie/concert show times and locations)
• M-Commerce and other transactions (e.g., buying movie/concert tickets, stock trades, banking transactions)
• Personal Information Manager (PIM) functions (e.g., making/checking appointments, managing contacts list, address book, etc.)
• Messaging (IM, unified messaging, etc…)
• Information capture (e.g. dictation of short memos)
• A usage scenario for multimodal applications with a GUI user agent on the terminal synchronized with an SRF automated voice service.