• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1992.06a
• /
• pp.98-102
• /
• 1992

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1995.06a
• /
• pp.188-191
• /
• 1995

• Speech Sciences
• /
• v.9 no.3
• /
• pp.47-59
• /
• 2002

A new brilliant service using human-voice and DTMF (Dual Tone Multi Frequency) technique is expected nowadays in order to obtain valuable information on the internet more easily. VoiceXML (Voice eXtensible Markup Language) is the right choice that makes the new service possible. In this paper, the design and implementation of IVR (Interactive Voice Response) server using VoiceXML is described, where it connects with internet and IVR server efficiently. IVR server using VoiceXML is composed of two groups: VoiceXML document handling and VoiceXML execution. Scenario part of IVR server corresponds to VoiceXML document, the execution is performed by VoiceXML execution.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.6 no.4
• /
• pp.513-518
• /
• 2002

This paper presents the new method for removing impulse noise using wavelet. Time and frequency-localization capabilities in denoising can provide excellent specialties comparing with the existing methods, because of including detail information of signal. The method in this paper, using denoising by threshold and slope of signal by wavelet, has superior denoising effect and can recognize edge of original signal. For objective judgement, the test signals are used HeaviSine and DTMF and this paper simulated by test signals which have added to impulse noise of different site individually.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.6 no.8
• /
• pp.1139-1145
• /
• 2002

Wavelet transformed data can filter signal with each frequency band, because it includes detail information about original signal. Therefore, in this paper, important two noises were removed by wavelet. About AWGN environment UDWT(undecimated discrete wavelet transform), applying hard-threshold, was used and about impulse noise environment, it can be possible to recognize edge of original signal as well as superior denoising effect by using two methods, denoising by threshold and slope of signal by wavelet. SNR was used as a judgemental criterion of a denoising effect and Blocks and DTMF(dual tone multi frequency) were used as a test signal.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.21 no.4
• /
• pp.1-6
• /
• 1984

An algorithm for the implementation of R2 MFC digital receiver is evaluated using simulated data. The algorithms satisfy the CCITT recommendations. In this paper, our aim is to provide more efficient way of decoding an input in PCM format and detecting the frequency components. The digital signal processor chip (S2811) is suggested to meet these aims Through the software simulation using an Altair 8800b microcomputer, the proposed receiver is shown to have tolerable performance in spite of a given noise, frequency drift, and twist.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2014.05a
• /
• pp.774-776
• /
• 2014

노령층의 증가로 인하여 독거 노인층이 점점 증가하고 있다. 독거 노인은 외부의 침입이나 갑작스런 건강상의 문제로 위험에 처해질 가능성이 매우 높다. 따라서 독거 노인의 의료복지에 대한 관심이 날로 증가하고 있다. 본 연구에서는 독거 노인의 의료 복지에 적용할 수 있는 활동량 모니터링 시스템을 개발하였다. 이 시스템은 저 전력으로 동작하는 지그비 임베드 네트워크와 DTMF(Dual Tone Multi Frequency) 전화 시스템으로 구성되어, U-Health 케어 개념으로 활동량과 혈압 등의 인체 정보를 일상 생활 속에서 병원 및 보호자에게 전송할 수 있다. 또한 본 연구 결과는 미래의 독거 노인 원격진료 시스템에 적용될 수 있다.

• Proceedings of the Korean Information Science Society Conference
• /
• 2002.10e
• /
• pp.412-414
• /
• 2002

본 논문에서는 PSTN망을 사용하는 일반전화기 사용자와 인터넷 전화 등이 접근할 수 있는 다이얼로그서버를 구현한다. 개발되는 다이얼로그서버는 VoiceXML로 작성된 서비스 컨텐츠를 사용하고, 이를 HTTP를 통하여 가져오며, DTMF 등의 사용자 입력을 받아서 미리 저장된 음성 데이터나 TTS 컴포넌트 등을 통한 음성정보 서비스를 제공한다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.19 no.7
• /
• pp.1346-1363
• /
• 1994

Packet voice protocols have been realized in many research works. But few studies for the interconnection of LAN and PABX to facilitate the voice communication have been done. In this paper, the gateway to interconnect the Ethernet LAN with the existing PABX telephone network for voice communication has been designed and implemented. The implemented gateway protocol is a modified protocol based on CCITT`s G.764 packetized voice protocol. To accomplish this goal the hardware system has been realized, which is divided into five parts: interface part with the telephone line, voice-processing part, PC interface part, controller part, and finally DTMF part. And the gateway software is divided into three parts: interface to make use of the packet driver which drives the network card, driver to drive the PABX gateway, and the protocol handling part.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06e
• /
• pp.219-222
• /
• 1998

음성인식시험은 다양한 사용자의 음성을 입력으로 음성인식을 수행하고 그 결과를 이용하여 시스팀의 성능을 평가하거나, 음성의 특징을 파악하기 위한 중요한 기능으로 음성인식 서비스의 질을 향상시키기 위한 필수적인 요소이다. 본 논문에서 제시하는 음성인식 자동시험장치는 음성인식의 결과를 DTMF 신호로 처리하도록 하여 사람의 개입 없이 빠르고 정확한 결과를 통해 인식율, 인식속도 등 인식기술과 관련된 중요한 정보를 얻을 수 있도록 하였다. 본 논문에서는 한국통신의 기업체 음성다이얼서비스의 음성인식시험을 중심으로 음성인식 자동시험장치의 구성 및 기능에 대해서 설명한다.