• The Journal of the Acoustical Society of Korea
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• v.26 no.7
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• pp.366-374
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• 2007

In underwater wireless communication, needs for long distance communication using the high frequency are surpassing ones of short range communication by ultrasonic wave, and demands for transmitting and receiving various data such as voice or high resolution image data are increasing as well. In this work, we studied the effects on the real underwater communication depending on the difference of digital modulation methods. Simulation shows that only the performance of GMSK among many other PSK based modulation schemes(BPSK, QPSK, MSK, GMSK) is significant. Test condition simulates the oceanographic conditions along the 207-survey line, 15Km south of Busan and SNR is maintained 35dB or below. Simulated tests are composed of both transmitting image data($3{\times}10^5$ pixel, 4 bit per pixel) and voice communication($10^{-2}$BER, channel capacity of 1Kbps). Test results show that there are gain of about 7 seconds in transmission time in image transmission case, where channel capacity for BPSK, QPSK, and MSK and for GMSK were 65 Kbps and 45 Kbps, respectively and gain of about 8Km in distances in voice communication case.

• Journal of Digital Convergence
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• v.12 no.8
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• pp.235-242
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• 2014

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares.

• Journal of Digital Convergence
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• v.10 no.5
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• pp.233-238
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• 2012

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Aerospace Engineering and Technology
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• v.7 no.2
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• pp.11-20
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• 2008

The current air-to-ground communication between aircraft pilots and ground controllers is done by voice communication and text-based data communication. International Civil Aviation Organization (ICAO) suggested the digital data communication techniques to improve accuracy and effectiveness of the current air-to-ground communication. As one of them, VDL M2, a VHF band digital data communication link, is expected to substitute the voice communication and text-based ACARS data communication. In this paper, the software design and implementation of the VDL M2 system developed by Korea Aerospace Research Institute.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.2C
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• pp.167-172
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• 2008

Since a 2-dimensional (2D) bar code can retrieve data and information quickly, it is widely used and recognized as a useful tool for many industrial applications. However, the information capacity of the 2D bar code is still limited. Recently the analog-digital code (AD code), which has the largest storage capacity yet contained in a code, has been developed, thereby expanding the bar code's application range because it overcomes the limitation of data capacity. In this paper, we present the AD code and implement an effective embedded system which can transform text information into voice using the 2D AD code and Text To Speech (TTS). This voice information can also be transmitted to blind people as well as the old by capturing the AD code on paper or in books.

• Journal of Digital Convergence
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• v.11 no.10
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• pp.453-458
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• 2013

HMM Speech recognition systems have a few weaknesses, including failure to recognize speech due to the mixing of environment noise other voices. In this paper, we propose a speech feature extraction methode using CHMM for extracting selected target voice from mixture of voices and noises. we make use of channel similarity and correlate relation for the selective speech extraction composes. This proposed method was validated by showing that the average distortion of separation of the technique decreased by 0.430 dB. It was shown that the performance of the selective feature extraction is better than another system.

• Journal of the Korean BIBLIA Society for library and Information Science
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• v.22 no.3
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• pp.173-188
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• 2011

This research intended to identify empirical recommendations for designing the DAISY service interface and used a case-based method. The case of this research was LG Digital Talking Book Library(http://voice.lg.or.kr) of LG Sangnam Library. A group of experts evaluated the current DAISY interface serviced by the web. After evaluation, major recommendations were suggested and these recommendations were used to develop the new DAISY Interface. Major recommendations were consideration of the reading flow of the screen-reader program, prevention of perception as an error about the time delay, development of web-based software, support for convenient functions and prevention of shortcut key overlap and so on.

• Journal of Korean Institute of Industrial Engineers
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• v.13 no.2
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• pp.17-34
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• 1987

Modern telecommunication switching systems are SPC (Stored Program Control) machines handling voice, data and other kinds of traffic, in an environment which tends to be fully digital switching and transmission. The throughput of such systems is determined by the real time capacity of its centralized or distributed control processors and by the traffic capacity of the switching network. Designers must verify the traffic and call processing capacity of the switching system and check its performance under traffic load before it is put into service. Verification of traffic and call processing capacity of switching systems is one of the problems treated by teletraffic studies; teletraffic studies are based on stochastic process, queueing theory, simulations and other quantitative methods of decision making. This paper reviews the general methodologies to evaluate the throughput and performance of the digital switching system. TDX-10, which is a fully digital switching system under development in ETRI, is employed to give illustrative examples of the methodologies discussed in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.842-856
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• 1994

The various services that a broadband integrated services digital network (B-ISDN) carries, have a wide range of delay, delay jitter and cell loss probability requirements. Design of appropriate control schemes for B-ISDN is an extremely important and challenging problem. In this paper, we proposed multiplexing algorithm with both delay and loss priorities in order to satisfy the diverse requirements. For the implementation of cell lose priority, we assumed that voice cells are generated as non-discardable(i.e., high priority) and discardable (i.e., low priotity)cells. The low priority voice cell may be discarded inside the network if congestion occurs. The cell dropping scheme is shown to reduce cell losses as well as delays for both voice and data. Such a load shedding scheme is expected to improve significantly utilization of B-ISDN.

• International Journal of Computer Science & Network Security
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• v.22 no.2
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• pp.175-184
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• 2022

Voice over Internet Protocol (VoIP) has grown in popularity as a low-cost, flexible alternative to the classic public switched telephone network (PSTN) that offers advanced digital features. However, additional security vulnerabilities are introduced by the VoIP system's flexibility and the convergence of voice and data networks. These additional challenges add to the normal security challenges that a VoIP system's underlying IP data network infrastructure confront. As a result, the VoIP network adds to the complexity of the security assurance task faced by businesses that use this technology. It's time to start documenting the many security risks that a VoIP infrastructure can face, as well as analyzing the difficulties and solutions that could help guide future efforts in research & development. We discuss and investigate the challenges and requirements of VoIP security in this research. Following a thorough examination of security challenges, we concentrate on VoIP system threats, which are critical for present and future VoIP deployments. Then, towards the end of this paper, some future study directions are suggested. This article intends to guide future scholars and provide them with useful guidance.