Speech Processing for IP Networks: Media Resource Control Protocol (Hardcover)

David Burke

  • 出版商: Wiley
  • 出版日期: 2007-04-01
  • 定價: $3,980
  • 售價: 8.0$3,184 (限時優惠至 2024-10-31)
  • 語言: 英文
  • 頁數: 368
  • 裝訂: Hardcover
  • ISBN: 0470028343
  • ISBN-13: 9780470028346
  • 相關分類: HTTPIPV6XML
  • 立即出貨

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Description

Media Resource Control Protocol (MRCP) is a new IETF protocol, providing a key enabling technology that eases the integration of speech technologies into network equipment and accelerates their adoption resulting in exciting and compelling interactive services to be delivered over the telephone.  MRCP leverages IP telephony and Web technologies such as SIP, HTTP, and XML (Extensible Markup Language) to deliver an open standard, vendor-independent, and versatile interface to speech engines. 

Speech Processing for IP Networks brings these technologies together into a single volume, giving the reader a solid technical understanding of the principles of MRCP, how it leverages other protocols and specifications for its operation, and how it is applied in modern IP-based telecommunication networks.  Focusing on the MRCPv2 standard developed by the IETF SpeechSC Working Group, this book will also provide an overview of its precursor, MRCPv1.

Speech Processing for IP Networks:

  • Gives a complete background on the technologies required by MRCP to function, including SIP (Session Initiation Protocol), RTP (Real-time Transport Protocol), and HTTP (Hypertext Transfer Protocol).
  • Covers relevant W3C data representation formats including Speech Synthesis Markup Language (SSML), Speech Recognition Grammar Specification (SRGS), Semantic Interpretation for Speech Recognition (SISR), and Pronunciation Lexicon Specification (PLS).
  • Describes VoiceXML - the leading approach for programming cutting-edge speech applications and a key driver to the development of many of MRCP’s features.
  • Explains advanced topics such as VoiceXML and MRCP interworking.

This text will be an invaluable resource for technical managers, product managers, software developers, and technical marketing professionals working for network equipment manufacturers, speech engine vendors, and network operators. Advanced students on computer science and engineering courses will also find this to be a useful guide.

 

Table of Contents

 
 

PART I. BACKGROUND.

1. Introduction.

1.1 Introduction to Speech Applications.

1.2 The MRCP Value Proposition.

1.3 History of MRCP Standardisation.

1.3.1 Internet Engineering Task Force.

1.3.2 World Wide Web Consortium.

1.3.3 MRCP: From Humble Beginnings Toward IETF Standard.

1.4 Summary.

2. Basic Principles of Speech Processing.

2.1 Human Speech Production.

2.1.1 Speech Sounds: Phonemics and Phonetics.

2.2 Speech Recognition.

2.2.1 Endpoint Detection.

2.2.2 Mel-Cepstrum.

2.2.3 Hidden Markov Models.

2.2.4 Language Modelling.

2.3 Speaker Verification and Identification.

   2.3.1 Feature Extraction.

   2.3.2 Statistical Modelling.

2.4 Speech Synthesis.

2.4.1 Front-end Processing.

2.4.2 Back-end Synthesis.

2.5 Summary.

3. Overview of MRCP.

3.1 Architecture.

3.2 Media Resource Types.

3.3 Network Scenarios.

3.3.1 VoiceXML IVR Service Node.

3.3.2 IP PBX with Voicemail.

3.3.3 Advanced Media Gateway.

3.4 Protocol Operation.

3.4.1 Establishing Communication Channels.

3.4.2 Controlling a Media Resource.

3.4.3 Walkthrough Examples.

3.5 Security.

3.6 Summary.

PART II. MEDIA AND CONTROL SESSIONS.

4. Session Initiation Protocol.

4.1 Introduction.

4.2 Walkthrough Example.

4.3 SIP URIs.

4.4 Transport.

4.5 Media Negotiation.

4.5.1 Session Description Protocol.

4.5.2 Offer/Answer Model.

4.6 SIP Servers.

4.6.1 Registrars.

4.6.2 Proxy Servers.

4.6.3 Redirect Servers.

4.7 SIP Extensions.

4.7.1 Capability Discovery.

4.8 Security.

4.8.1 Transport and Network Layer Security.

4.8.2 Authentication.

4.8.3 S/MIME.

4.9 Summary.

5. Session Initiation in MRCP.

5.1 Introduction.

5.2 Initiating the Media Session.

5.3 Initiating the Control Session.

5.4 Session Initiation Examples.

5.4.1 Single Media Resource.

5.4.2 Adding and Removing Media Resources.

5.4.3 Distributed Media Source/Sink.

5.5 Locating Media Resource Servers.

5.5.1 Requesting Server Capabilities.

5.5.2 Media Resource Brokers.

5.6 Security.

5.7 Summary.

6. The Media Session.

6.1 Media Encoding.

6.1.1 Pulse Code Modulation (PCM).

6.1.2 Linear Predictive Coding (LPC).

6.2 Media Transport.

6.2.1 Real-Time Protocol (RTP).

6.2.2 DTMF.

6.3 Security.

6.4 Summary.

7. The Control Session.

7.1 Message Structure.

7.1.1 Request Message.

7.1.2 Response Message.

7.1.3 Event Message.

7.1.4 Message Bodies.

7.2 Generic Methods.

7.3 Generic Headers.

7.4 Security.

7.5 Summary.

PART III. DATA REPRESENTATION FORMATS.

8. Speech Synthesis Markup Language (SSML).

8.1 Introduction.

8.2 Document Structure.

8.3 Recorded Audio.

8.4 Pronunciation.

8.4.1 Phonemic/Phonetic Content.

8.4.2 Substitution.

8.4.3 Interpreting Text .

8.5 Prosody.

8.5.1 Prosodic Boundaries.

8.5.2 Emphasis.

8.5.3 Speaking Voice.

8.5.4 Prosodic Control.

8.6 Markers .

8.7 Metadata.

8.8 Summary.

9. Speech Recognition Grammar Specification (SRGS).

9.1 Introduction.

9.2 Document Structure.

9.3 Rules, Tokens, and Sequences.

9.4 Alternatives.

9.5 Rule References.

9.5.1 Special Rules.

9.6 Repeats.

9.7 DTMF Grammars.

9.8 Semantic Interpretation.

9.8.1 Semantic Literals.

9.8.2 Semantic Scripts.

9.9 Summary.

10. Natural Language Semantics Markup Language (NLSML).

10.1 Introduction.

10.2 Document Structure.

10.3 Speech Recognition Results.

10.3.1 Serialising Semantic Interpretation Results.

10.4 Voice Enrollment Results.

10.5 Speaker Verification Results.

10.6 Summary.

11. Pronunciation Lexicon Specification (PLS).

11.1 Introduction.

11.2 Document Structure.

11.3 Lexical Entries.

11.4 Abbreviations and Acronyms.

11.5 Multiple Orthographies.

11.6 Multiple Pronunciations.

11.7 Summary.

PART IV. MEDIA RESOURCES.

12. Speech Synthesiser Resource.

12.1 Overview.

12.2 Methods.

12.2.1 SPEAK.

12.2.2 PAUSE.

12.2.3 RESUME.

12.2.4 STOP.

12.2.5 BARGE-IN-OCCURRED.

12.2.6 CONTROL.

12.2.7 DEFINE-LEXICON.

12.3 Events.

12.3.1 SPEECH-MARKER.

12.3.2 SPEAK-COMPLETE.

12.4 Headers.

12.5 Summary.

13. Speech Recogniser Resource.

13.1 Overview.

13.2 Recognition Methods.

13.2.1 RECOGNIZE.

13.2.2 DEFINE-GRAMMAR.

13.2.3 START-INPUT-TIMERS.

13.2.4 GET-RESULT.

13.2.5 STOP.

13.2.6 INTERPRET.

13.3 Enrollment Methods.

13.3.1 START-PHRASE-ENROLLMENT.

13.3.2 ENROLLMENT-ROLLBACK.

13.3.3 END-PHRASE-ENROLLMENT.

13.3.4 MODIFY-PHRASE.

13.3.5 DELETE-PHRASE.

13.4 Events.

13.4.1 START-OF-INPUT.

13.4.2 RECOGNITION-COMPLETE.

13.4.3 INTERPRETATION-COMPLETE.

13.5 Recognition Headers.

13.6 Enrollment Headers.

13.7 Summary.

14. Recorder Resource.

14.1 Overview.

14.2 Methods.

14.2.1 RECORD.

14.2.2 START-INPUT-TIMERS.

14.2.3 STOP.

14.3 Events.

14.3.1 START-OF-INPUT.

14.3.2 RECORD-COMPLETE.

14.4 Headers.

14.5 Summary.

15. Speaker Verification Resource.

15.1 Overview.

15.2 Methods.

15.2.1 START-SESSION.

15.2.2 END-SESSION.

15.2.3 VERIFY.

15.2.4 VERIFY-FROM-BUFFER.

15.2.5 VERIFY-ROLLBACK.

15.2.6 START-INPUT-TIMERS.

15.2.7 GET-INTERMEDIATE-RESULT.

15.2.8 STOP.

15.2.9 CLEAR-BUFFER.

15.2.10 QUERY-VOICEPRINT.

15.2.11 DELETE-VOICEPRINT.

15.3 Events.

15.3.1 START-OF-INPUT.

15.3.2 VERIFICATION-COMPLETE.

15.4 Headers.

15.5 Summary.

PART V. PROGRAMMING SPEECH APPLICATIONS.

16. Voice eXtensible Markup Language (VoiceXML).

16.1 Introduction.

16.2 Document Structure.

16.2.1 Applications and Dialogs.

16.3 Dialogs.

16.3.1 Forms.

16.3.2 Menus.

16.3.3 Mixed Initiative Dialogs.

16.4 Media Playback.

16.5 Media Recording.

16.6 Speech and DTMF Recognition.

16.6.1 Specifying Grammars.

16.6.2 Grammar Scope and Activation.

16.6.3 Configuring Recognition Settings.

16.6.4 Processing Recognition Results.

16.7 Flow Control.

16.7.1 Executable Content.

16.7.2 Variables, Scopes, and Expressions.

16.7.3 Document and Dialog Transitions .

16.7.4 Event Handling.

16.8 Resource Fetching.

16.9 Call Transfer.

16.10 Summary.

17. VoiceXML and MRCP Interworking.

17.1 Introduction.

17.2 Interworking Fundamentals.

17.2.1 Play Prompts.

17.2.2 Play and Recognise.

17.2.3 Record.

17.3 Application Example.

17.3.1 VoiceXML Scripts.

17.3.2 MRCP Flows.

17.4 Summary.

Appendix A. MRCP Version 1.

A.1 Overview.

A.2 Session Management and Message Transport.

A.3 General Protocol Details.

A.4 Speech Synthesiser Resource.

A.5 Speech Recogniser Resource.

Appendix B. XML Primer.

B.1 Background.

B.2 Basic Concepts.

B.3 Namespaces.

B.4 Document Schemas.

Appendix C. HTTP Primer.

C.1 Background.

C.2 Basic Concepts.

C.2.1 GET Method.

C.2.2 POST Method.

C.3 Caching.

C.4 Cookies.

C.5 Security.

References.

Index.

Acronyms.

商品描述(中文翻譯)

描述

媒體資源控制協議(MRCP)是一種新的IETF協議,提供了一個關鍵的技術,使語音技術能夠輕鬆集成到網絡設備中,並加速其采用,從而提供令人興奮和引人入勝的互動服務。MRCP利用IP電話和Web技術(如SIP、HTTP和XML)提供了一個開放標準、供應商獨立和多功能的語音引擎接口。

《IP網絡語音處理》將這些技術結合到一本書中,讓讀者對MRCP的原理有著扎實的技術理解,了解它如何利用其他協議和規範進行操作,以及它如何應用於現代基於IP的電信網絡。本書重點介紹了IETF SpeechSC工作組開發的MRCPv2標準,並概述了其前身MRCPv1。

《IP網絡語音處理》包括以下內容:
- 提供MRCP運行所需的技術背景,包括SIP(會話初始化協議)、RTP(實時傳輸協議)和HTTP(超文本傳輸協議)。
- 詳細介紹W3C的數據表示格式,包括語音合成標記語言(SSML)、語音識別語法規範(SRGS)、語音識別的語義解釋(SISR)和發音詞典規範(PLS)。
- 描述了VoiceXML,這是一種領先的用於編程尖端語音應用的方法,也是MRCP許多功能發展的關鍵驅動因素。
- 解釋了高級主題,如VoiceXML和MRCP的互操作性。

這本書對於網絡設備製造商、語音引擎供應商和網絡運營商的技術經理、產品經理、軟件開發人員和技術營銷專業人員來說是一個寶貴的資源。計算機科學和工程課程的高級學生也會發現這是一本有用的指南。

目錄

第一部分 背景
1. 簡介

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