• Proceedings of the KSR Conference
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• 2009.05a
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• pp.1993-1998
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• 2009

Recently, with expansion and advent of the wire&wireless networks has been set up by various heterogeneous networks in urban transit networks. It's focussing on an interworking technology. As same as way, In a subway & railway monitoring system networks has been set up by various heterogeneous technologies such as IEEE802.11 WLAN, IEEE802.15 WPAN, DVBT, UWB, BinaryCDMA, ZigBee. These heterogeneous technologies is deeply investigating for the purpose of interworking and easy control as a research area. In this paper, we propose using VoiceXML based of XML for ensure interworking within heterogeneous subway & railway networks. Therefore It will be provide service for staff who has moving and anywhere, anytime in outside. Eventually, we expect that confirm efficiency of effective system through this study.

• Journal of Scientific & Technological Knowledge Infrastructure
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• s.6
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• pp.68-81
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• 2001

현재 진행되고 있는 XML 응용분야의 표준은 각 분야별로 구체적으로 진행되고 있으며 이러한 시점에서 AT&T, 루슨트 테크놀로지스, 모토롤러 등 3사는 전화와 인터넷 서버와의 연동을 음성 처리 기술을 바탕으로 하여 기존 인터넷의 다양한 정보를 검색 처리할 수 있는 VXML(Voice Extensible Markup Language)이라는 인터넷 음성처리 표준안을 마련하고 있다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.5
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• pp.949-954
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• 2003

In this study, Automatic Speech Recognition Application System is designed and implemented to realize transformation from present GUI-centered web services to VUI-centered web service. Due to ASP's restriction with web in reusability and portability, in this study, Automatic Speech Recognition Application System with Javabeans Component Architecture is devised and studied. Also the voice web browsing which is able to transfer voice and graphic information simultaneously is studied using Remote AWT(Abstract Windows Toolkit).

• MALSORI
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• no.53
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• pp.171-183
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• 2005

Call Control eXtensible Markup (CCXML) is a standard language that supports a call control of voice dialog systems such as VoiceXML based systems. CCXML allows developers to handle telephony calls in an easy way without deep knowledge about telephony networks and their switching systems.We design and implement a call control markup interpreter. At the implementation, we use a Dialogic JCT-LS board, but, by designing a wrapping class for CTI (computer telephony board) features, the interpreter can easily adopt other CTI boards. We also design and implement event-based interaction scheme between the interpreter and voice dialog systems. For verifying the interaction scheme, we implement a simple voice dialog system.

• Annual Conference on Human and Language Technology
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• 2002.10e
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• pp.69-76
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• 2002

최근 음성 합성 과정에서 화자의 의도를 가장 많이 반영하는 언어 정보인 문맥 정보를 사용하려는 시도가 이루어지고 있으나 문맥 정보를 적은 비중으로 사용하기 때문에 자연성 향상에 큰 도움을 주지 못하고 있다. 본 연구에서는 구문 정보, 의미 정보를 억양 생성 과정에 이용함과 동시에 문맥 정보와 음성 정보와의 관계를 음성 데이터를 바탕으로 분석하여 다양한 문맥 정보를 음성 합성 과정에 반영하는 방법을 제안한다. 또한 한국어에서 나타나는 다양한 억양 곡선 유형을 형태소를 이용하여 의다 효율적으로 처리할 수 있는 방법을 제안하여 자연스러운 억양 생성 시스템을 구현하고 시스템의 결과를 음소 단위 억양 생성기와 VoiceXML을 이용하여 적용시켜보고 결과를 논의한다.

• Proceedings of the KIEE Conference
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• 2007.04a
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• pp.275-277
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• 2007

Recent activation of DMB has enabled various high quality video, audio and data services. And there are various user facilities functions using digital data transmission. One of the various user facilities functions is EPG(Electronic Program Guide). EPG supports schedule of programs on screen for. audiences. EPG is composed to time, title, channel, genre etc. Users can select a program what they want to browsing. Currently EPG services are displaying program schedule on screen visually and make users to input ke:ywords with keypads, remote control devices or touch screen etc. However, this approach could cause a serious restriction to some users like to drivers or visually handicapped persons. A standard for a voice EPG to T-DMB is proposed. This method must be transferred VoiceXML based EPG files from the transmitter to receivers. This approach has a problem to process a standardization because the transmitter and receivers should be modified. We proposed and implemented a voice EPG platform that generates the voice EPG files from T-DMB SI without transferring voice EPG file from the transmitter.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.21-30
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• 2005

Development of both multimedia technology and communication network technology has accomplished many changes through the field of learning system. For the construction of a more efficient and clever learning system there is a research being done by the use of the Web and the telephone network. But until now, the case of current implemented teaming system is single system and so it has each merits and demerits. That is to say, when we use the learning system through the Web, the demerit is only possible by the static states using computer. For those who do not use the computer, the demerit is that the user must learn the use of the new system. Also, the case of using telephone network has merits that one can use the system anyplace, anytime by the telephone. But it has the problem of not being able to transmit information very efficiently. From these, this paper proposes the learning system that can be used efficiently and conveniently anyplace, anytime by connecting both telephone network and web. Also, we propose a new algorithm of user ID, password and name registration function using teaming system using VoiceXML and individual learning progress save function using VoiceXML and web.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.273-276
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• 2003

In this study, Automatic Speech Recognition Application System is designed and implemented to realize transformation from present GUI-centered web services to VUI-centered web service. Due to ASP's restriction with web in speed and implantation, in this study, Automatic Speech Recognition Application System with Java beans Component Architecture is devised and studied. Also the voice web browsing which is able to transfer voice and graphic information simultaneously is studied using Remote AWT(Abstract Windows Toolkit).

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.119-122
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• 2002

This paper focuses on building up a database of commercial stocks using XML syntax and looks into a way of building up a system with the combination of XML and XSLT that provides connectivity to client-server databases through vocal means. The use of XSLT has several advantages. Most importantly, it can transform a type of data into different formats. A vocal interface minimizes some space and time limits imposed on users outside premises when they need an instant connection to their database. In this fashion, the users can check information on stock lists without being pressurized by certain limits. PC, PDAs and cellular phones are some examples of mobile connection. The use of VoiceXML creates vocal applications. In VoiceXML servies, users can gain immediate access to data upon the input of their voices and the DTMF signals of the telephone.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.745-748
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• 2003

본 논문에서는 SIP(Session Initiation Protocol) 호의 처리 모듈을 가지고 있지 않은 웹 상의 클라이언트들에게 음성서비스 및 멀티미디어 서비스를 제공할 수 있는 SIP 호 처리 서버를 설계 및 구현 하였다. 서버상에서 SIP 호 설정을 위해 사용되는 메시지를 VoiceXML의 태그형태로 변환하는 기능을 제공함으로써 비 SIP 사용자들에게도 SIP 서비스를 할 수 있도록 하였다.