• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.655-657
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• 2008

We suggest the digital voice encryption module using the pseudo random number generator and design the sorority device of a telephone using the module. The proposed method provides encryption method of the telephone against the third party. This encryption method uses pseudo random number generator which computes the encryption key using the shared secret key and the current time value.

• MALSORI
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• no.49
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• pp.145-158
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• 2004

This paper is aiming at implementation of real-time speaker verification system using DSP board. Dialog/4, which is based on microprocessor and DSP processor, is selected to easily control telephone signals and to process audio/voice signals. Speaker verification system performs signal processing and feature extraction after receiving voice and its ID. Then through computing the likelihood ratio of claimed speaker model to the background model, it makes real-time decision on acceptance or rejection. For the verification experiments, total 15 speaker models and 6 background models are adopted. The experimental results show that verification accuracy rates are 99.5% for using telephone speech-based speaker models.

• MALSORI
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• no.48
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• pp.155-173
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• 2003

This paper proposes a noise reduction-based speech detection method under telephone channel environments. We adopt the AURORA front-end noise reduction algorithm based on the two-stage mel-warped Wiener filter approach as a preprocessor for the frequency domain speech detector. The speech detector utilizes mel filter-bank based useful band energies as its feature parameters. The preprocessor firstly removes the adverse noise components on the incoming noisy speech signals and the speech detector at the next stage detects proper speech regions for the noise-reduced speech signals. Experimental results show that the proposed noise reduction-based speech detection method is very effective in improving not only the performance of the speech detector but also that of the subsequent speech recognizer.

• Journal of the Korean Professional Engineers Association
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• v.16 no.1
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• pp.4-12
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• 1983

Data communication has historically evolved from leased lines, to use of the public telephone network, and eventually to dedicated(exclusive) data networks. It requires an enormous amount of money for establishing a separate and independent data network at the beginning stage. No country has ever adopted this method In the Republic of Korea too the age of leased circuits is passing and it is scheduled to open the public telephone network to the data transmission and to install packet mode processor in the last half of 1983. This paper presents a survey on characteristics of the public telephone network in Seoul and a future development of the data communication of Korea.

• Journal of Biomedical Engineering Research
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• v.13 no.1
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• pp.69-78
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• 1992

This Paper describes implementation of a remote ECG diagnostic system using telephone line. The overall system includes ECG data acquisition system, ECG terminal, system control software, automatic diagnosis system, and transmission system.'The proposed system provides various functions, which are ECG data acquisition, transmission, receiving, diagnosis and dialogue between patients and medical doctors. Thls system is very simple and convienient to use. We evaluate the performance of modem and the accuracy of automatic diagnosis algorithm. The obtained results suggest the Possibilities of a remote ECG diagnostic system using the only existed telephone line.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.91-96
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• 2009

In Public Switched Telephone Networks(PSTN) number portability is implemented by utilizing Intelligent Network(IN) functions for number mapping. And voice over IP(VoIP) and IP Multimedia Subsystem(IMS) networks can deploy number portability by using E.164 Number Mapping(ENUM). This paper discuss the possibility of using E.164 Number Mapping(ENUM) for number portability in voice over IP/IP Multimedia Subsystem and Public Switched Telephone Networks, eliminating the need for Number Portability Database(NPDB) for number portability routing data in Public Switched Telephone Networks.

• Electronics and Telecommunications Trends
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• v.9 no.4
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• pp.75-82
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• 1994

언제, 어디서나, 누구와도' 통신이 가능하게 한다는 궁극적인 목표를 가지고 통신의 개인화가 연구되고 있으며, 통신 선진국에서는 PCS를 실현하기 위한 PSTN(Public Switching Telephone Network) 접근 방법의 초기 단계로 CT2 (Cordless Telephone 2)의 개발과 시험망 운영을 하고 있거나 이미 서비스를 상용화하고 있다. CT2는 1989년 영국에서 제안되었고 1992년 유럽 표준 (I-ETS 300-131)으로 채택된 후 유럽 각국과 홍콩이나 싱가포르 같은 도시 국가에서 활성화 되어 많은 가입자들이 이용하고 있다. CT2 이용자는 도로, 지하철 혹은 공공 장소에서는 PSTN에 연결된 CT2 기지국에 접속되어 발신 위주의 휴대 전화 서비스를 받을 수 있으며, 가정에서는 우리가 현재 사용하고 있는 무선 전화기와 동일하게 사용할 수 있는 소출력 휴대 전화 서비스로 단말기의 이동성이 어느 정도 실현된 서비스이다. 국내 CT2 사업자는 1993년 12월 체신부로 부터 시범 사업 계획의 승인을 얻어 주파수 대역 910-914 MHz에서 1995년 서울 지역에 서비스 상용화를 목표로 여의도에 시험망 운용과 함께 연구되고 있다. 본 고에서는 CT2 서비스의 일반적인 개요와 특성 그리고 제외국의 CT2 동향을 고찰해봄으로써, CT2는 어떤 서비스이며 CT2의 기술 기준 마련과 무선 인터페이스 측정 및 분석 방법을 연구하는 데 기초로 삼고자 한다.

• Proceedings of the KSPS conference
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• 2003.10a
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• pp.145-148
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• 2003

To improve the recognition performance of Korean connected digit telephone speech, in this paper, both Aurora feature extraction method that employs noise reduction 2-state Wiener filter and DWFBA method are investigated and used. CMN and MRTCN are applied to static features for channel compensation. Telephone digit speech database released by SITEC is used for recognition experiments with HTK system. Experimental results has shown that Aurora feature is slightly better than MFCC and DWFBA without channel compensation. And when channel compensation is included, Aurora feature is slightly better than DWFBA with MRTCN.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.17-21
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• 1998

This paper presents an implementation of a multi-channel speech surveillance system over telephone lines using TMS320C31 DSP chips. The incoming speech into each telephone line are first compressed simultaneously in real-time by the popular vector-sum excited linear predictive (VSELP) speech coding algorithm at the rate of 8 Kbps. The compressed steech bit streams are then multiplexed with those of other users. The multiplexed speech bit streams are transferred to the system storage equipments with some other required information so that a system operator can later monitor the stored speech data whenever it is necessary. The host program runs under Microsoft Windows95 for an efficient man-machine interface and a future upgrade-ability. We have confirmed that the overall 64-channel system operates satisfactorily in realtime. We also have checked approximately up to 2,880 total hours of recording capability of the system on a playback module and two removable backup drives.

• Proceedings of the KIEE Conference
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• 2002.11c
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• pp.573-576
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• 2002

In this paper we propose multi-functional telephone rejecting a simultaneous call at home or small office with several telephones in one line. Also we describe the protocol to transfer a call to the other telephone in different place with ease. So the proposed scheme can protect a personal privacy and provide the conveniency. Since the proposed phone is implemented with Holtek HT95C20P microprocessor, the most of functions can be easily changed accordance with user's needs.