• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.483-486
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• 1988

Link-by-link and end-to-end signaling affect the probability of inefficient trunk seizure, the usage of backward signal features, post dialing delay, total signal information processing load, each differently. Considering the circumtances of the domestic public switched telephone network, we conclude that end-to-end signaling is better suited to the domestic network.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.16 no.28
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• pp.181-186
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• 1993

The objective of this paper is to present a method for synthetic performance and quality evaluation of dial up network modem in users place. In the past, universally recognized criteria have not been available to model switched network modem performance. Reasonable assurance of modem performance on the public switched telephone network by standardized test channel which is suggested by ELA(Electronic Industries Association). These tests of modem from a variety manufacturers more reflective of actual performance on the network.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.37-40
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• 2000

전화망 음성 인식 시스템에서 사용할 수 있는 데이터베이스 구축의 질과 양은 인식 시스템의 성능에 중대한 영향을 미친다. 따라서, 전화망 음성 데이터 베이스 구축에 관한 효과적인 방법들이 연구되고 있다. 본 논문은 공용으로 사용할 수 있는 음성 데이터 베이스의 전화망 변환 방법 및 활용 방안에 대하여 소개한다.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2002.05a
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• pp.432-437
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• 2002

DSPA311(Analog Subscriber Line Board Assembly) is offer the interface of between analog subscriber and TDX-100 exchange system. DSPA311 is belong ASI block, accommodate dial and MFC telephone subscriber of 32 channel, and voice signal designed for interface with TSW, and 2 and 4 wire loop impedance is 600 (ohm). DSPA311 is consist 4 channel daughter beard QSLM-10(Quad Subscriber Line Module-10) and perform BORSCHT and be possible A/U-law select and GAIN value control by data control of DSPA171(Device controller I). In this Paper, We described the function test program for the DSPA311 Board by using the HP3070CT combinational test system, and an unmanned automatic test system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10B
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• pp.1793-1802
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• 2000

In this paper we propose an efficient implementation technique for cyclic extension in VDSL(Very High bit-rate Digital Subscriber Line) systems using Zipper duplexing and analyze its performances under typical telephone channel environments. In Zipper-based VDSL systems each DTM(discrete-multitone) block is appended by both cyclic prefix(CP) and cyclic suffix(CS). The CP is inserte to prevent both intersymbol interference (ISI) and iterchannel interference (ICI) while the CS is appended to ensure orthogonality between the upstream and downstream carriers thus preventing near-end crosstalk (NEXT). However in order to implement the CP in the transmitter side of the VDSL system an additional hardware is required to append the latter part of each DMT symbol to the beginning of the DMT symbol. In this paper we propose a VDSL system with Zipper duplexing using only CS to reduce hardware complexity (memory and processing delay) required for implementation of CP. It is shown by computer simulation that the proposed approach has the same capacity under typical channel environments as the previous Zipper-based VDSL system using both CP and CS. even with a significantly lower hardware complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.12 no.1
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• pp.60-70
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• 1987

Recently the tendency of the network system development rapidly progresses toward ISDN that intergrates all presetn-day communication networks and services into a single, universal network set up on the basis of the exisiting telephone network. For this reason many countries pay worldwide attention to the researches about ISDN, especially to the researches about the exchange and the ISDN user-network interface which is one of the most important parts of network system. This paper is the first part of the two-par papers describing the ISDN user-network interface. in this paper, after surveying the architecture of ISDN exchange recommended by CCITT, the general architecture of the ISDN exchage is proposed, Based on this architecture, the switching emulator is implemented, and the necessary conditions of the ISDN exchage(LAPO, CCP etc) are also studied.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.45-50
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• 2000

This paper proposes a delta energy method based on Parameter Filtering(PF), which is a speech segmentation algorithm for text dependent speaker verification system over telephone line. Our parametric filter bank adopts a variable bandwidth along with a fixed center frequency. Comparing with other methods, the proposed method turns out very robust to channel noise and background noise. Using this method, we segment an utterance into consecutive subword units, and make models using each subword nit. In terms of EER, the speaker verification system based on whole word model represents 6.1%, whereas the speaker verification system based on subword model represents 4.0%, improving about 2% in EER.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.182-187
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• 2003

In this thesis, we propose the algorithm to improve the performance of speaker verification that is mapping feature parameters by using RBF neural network. There is a big difference between wire vector region and wireless one which comes from the same speaker. For wire/wireless speakers model production, speaker verification system should distinguish the wire/wireless channel that based on speech recognition system. And the feature vector of untrained channel models is mapped to the feature vector(LPC Cepstrum) of trained channel model by using RBF neural network. As a simulation result, the proposed algorithm makes 0.6%∼10.5% performance improvement compared to conventional method such as cepstral mean subtraction.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.361-373
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• 2002

In this paper, we propose a channel normalization method to improve the performance of CMS (cepstral mean subtraction) which is widely adopted to normalize a channel variation for speech and speaker recognition. CMS which estimates the channel effects by averaging long-term cepstrum has a weak point that the estimated channel is biased by the formants of voiced speech which include a useful speech information. The proposed Formant-broadened Cepstral Mean Subtraction (FBCMS) is based on the facts that the formants can be found easily in log spectrum which is transformed from the cepstrum by fourier transform and the formants correspond to the dominant poles of all-pole model which is usually modeled vocal tract. The FBCMS evaluates only poles to be broadened from the log spectrum without polynomial factorization and makes a formant-broadened cepstrum by broadening the bandwidths of formant poles. We can estimate the channel cepstrum effectively by averaging formant-broadened cepstral coefficients. We performed the experiments to compare FBCMS with CMS, PFCMS using 4 simulated telephone channels. In the experiment of channel estimation, we evaluated the distance cepstrum of real channel from the cepstrum of estimated channel and found that we were able to get the mean cepstrum closer to the channel cepstrum due to an softening the bias of mean cepstrum to speech. In the experiment of text-independent speaker identification, we showed the result that the proposed method was superior than the conventional CMS and comparable to the pole-filtered CMS. Consequently, we showed the proposed method was efficiently able to normalize the channel variation based on the conventional CMS.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.1
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• pp.35-47
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• 2002

This paper proposes a self organizing RBF neural network equalizer for the equalization of digital communications. It is the most important for the equalizer using the RBF neural network to estimate the RBF centers correctly and quickly, which are the desired channel states. However, the previous RBF equalizers are not used in the actual communication system because of some drawbacks that the number of channel states has to be known in advance and many centers are necessary. Self organizing neural network equalizer proposed in this paper can implement the equalization without prior information regarding the number of channel states because it selects RBF centers among the signals that are transmitted to the equalizer by the new addition and removal criteria. Furthermore, the proposed equalizer has a merit that is able to make a equalization with fewer centers than those of prior one by the course of the training using LMS and clustering algorithm. In the linear, nonlinear and standard telephone channel, the proposed equalizer is compared with the optimal Bayesian equalizer for the BER performance, the symbol decision boundary and the number of centers. As a result of the comparison, we can confirm that the proposed equalizer has almost similar performance with the Bavesian enualizer.