ETSI TS 122 243 V17.0.0 (2022-04)

Digital cellular telecommunications system (Phase 2+) (GSM); Universal Mobile Telecommunications System (UMTS); LTE; Speech recognition framework for automated voice services; Stage 1 (3GPP TS 22.243 version 17.0.0 Release 17)

RTS/TSGS-0122243vh00

General Information

• Status: Not Published
• Technical Committee: 3GPP SA 1 - Services
• Current Stage: 12 - Completion
• Completion Date: 21-Apr-2022

Standards Content (Sample)

TECHNICAL SPECIFICATION
Digital cellular telecommunications system (Phase 2+) (GSM);
Universal Mobile Telecommunications System (UMTS);
LTE;
Speech recognition framework for automated voice services;
Stage 1
(3GPP TS 22.243 version 17.0.0 Release 17)

3GPP TS 22.243 version 17.0.0 Release 17 1 ETSI TS 122 243 V17.0.0 (2022-04)

Reference
RTS/TSGS-0122243vh00
Keywords
GSM,LTE,UMTS
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ETSI
3GPP TS 22.243 version 17.0.0 Release 17 2 ETSI TS 122 243 V17.0.0 (2022-04)
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ETSI
3GPP TS 22.243 version 17.0.0 Release 17 3 ETSI TS 122 243 V17.0.0 (2022-04)
Contents
Intellectual Property Rights . 2
Legal Notice . 2
Modal verbs terminology . 2
Foreword . 4
Introduction . 4
1 Scope . 5
2 References . 6
2.1 Normative References . 6
2.2 Informative References . 6
3. Definitions and abbreviations . 6
3.1 Definitions . 6
3.2 Abbreviations . 7
4 Requirements . 8
4.1 Initiation . 8
4.2 Information during the speech recognition session . 9
4.3 Control . 9
4.4 User Perspective (User Interface) . 9
5 UE and network capabilities. 9
6 Administration . 10
6.1 Authorization . 10
6.2 Deauthorization . 10
6.3 Registration . 10
6.4 Deregistration . 10
6.5 Activation . 10
6.6 Deactivation . 11
7 Service Provisioning. 11
8 Security. 11
9 Privacy . 11
10 Charging . 11
11 Roaming . 12
12 Interaction with other services . 12
Annex A (informative):  Speech recognition Framework-based automated voice service
examples . 13
Annex B (informative):  Change History . 14
History . 15

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3GPP TS 22.243 version 17.0.0 Release 17 4 ETSI TS 122 243 V17.0.0 (2022-04)
Foreword
rd
This Technical Specification has been produced by the 3 Generation Partnership Project (3GPP).
The contents of the present document are subject to continuing work within the TSG and may change following formal
TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an
identifying change of release date and an increase in version number as follows:
Version x.y.z
where:
x the first digit:
1 presented to TSG for information;
2 presented to TSG for approval;
3 or greater indicates TSG approved document under change control.
y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections,
updates, etc.
z the third digit is incremented when editorial only changes have been incorporated in the document.
Introduction
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 etc. .) 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.
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1 Scope
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.
Speech-enabled Services
Multimodal Services Speech-only Services
Multi-device Services
Client-side only Server-side
Speech engines Speech engines
Speech Recognition
Circuit Switched
Framework
Speech Recognition
(Packet Switched)
DSR optimized Conventional
Other
Codec Codec
Figure 1 - Positions the scope of the speech recognition framework as part of general speech
enabled services.
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2 References
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.
2.1 Normative References
[1] 3GPP TS 21.133: "3G security; Security threats and requirements".
[2] 3GPP TR 21.905: "Vocabulary for 3GPP Specifications".
[3] 3GPP TR 22.941: "IP based multimedia framework; Stage 0".
[4] 3GPP TS 22.105: "Services and service capabilities".
[5] 3GPP TS 22.228: "Service requirements for the Internet Protocol (IP) multimedia core network
subsystem; Stage 1".
[6] 3GPP TR 22.977: "Feasibility study for speech-enabled services".
2.2 Informative References
[7] ETSI ES 201 108 v1.1.2: "Distributed Speech Recognition: Front-end Feature Extraction
Algorithm; Compression Algorithm", April 2000.
[8] Void
[9] Void
[10] ETSI ES 202 050 v0.0.0 "Speech Processing, Transmission and Quality aspects (STQ);
Distributed speech recognition; Advanced front-end feature extraction algorithm; Compression
algorithms; DSR advanced front end", standard selected; document in preparation.
3. Definitions and abbreviations
Definitions and abbreviations used in the present document are listed in TR 21.905 [2]. For the purposes of this document
the following definitions and abbreviations apply:
3.1 Definitions
Automated Voice Services: Voice applications that provide a voice interface driven by a voice dialog manager to drive
the conversation with the user in order to complete a transaction and possibly execute requested actions. It relies on speech
recognition engines to map user voice input into textual or semantic inputs to the dialog manager and mechanisms to
generate voice or recorded audio prompts (text-to-speech synthesis, audio playback,). It is possible that it relies on
additional speech processing (e.g. speaker verification). Typically telephony-based automated voice services also provide
call processing and DTMF recognition capabilities. Examples of traditional automated voice services are traditional IVR
(Interactive Voice Response Systems) and VoiceXML Browsers.
Barge-in event: Event that takes place when the user starts to speak while audio output is generated.
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Conventional Codec: The module in UE that encodes the speech i
...