• Proceedings of the KSLP Conference
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• 1998.11a
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• pp.218-219
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• 1998

The term "Professional voice user" refers to not only a singer or actor. but also anyone who uses the voice as a primary means of occupational communication Examination begins with observation of a patient and careful listening of the patient's voice during history taking. Examination of the vocal fold includes observation of the vocal fold, its mobility and vibration. (omitted)

• Journal of Korea Society of Industrial Information Systems
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• v.23 no.6
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• pp.59-66
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• 2018

In this paper, we experimented how long distance voice communication is possible After implemented PTT(Push to talk) Bi-directional radio using open source project Codec 2, which is a low speed voice codec for digital amateur radio and 900MHz FSK transceiver. In case of a general digital radio, the AMBE+2 codec, which is regarded as an industry standard in terms of performance, is expensive and has the monopoly of technology. Using the 400MHz band in terms of frequency, narrow bandwidth of DMR(12.5kHz) and DPMR(6.25kHz) is used, so the data rate is low. In the 900MHz bandwidth can be extended, which is advantageous in terms of data transmission. As a result of the voice quality and distance field test, we could find that the communication takes place within about 500m. In this paper, only voice communication is reviewed. if a review of data transmission such as a simple image is added, this solution can be used in various fields as a low cost IOT radio.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.111-116
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• 2016

According to the Ministry of Science, ICT and Future Planning's survey of information gap, smartphone retention rate of disabled people stayed in one-third of non-disabled people, the situation is significantly less access to information for people with disabilities than non-disabled people. In this paper, we develop an application, CallHelper, that helps to be more convenient to use mobile voice calls to the auditory disabled people. CallHelper runs automatically when a call comes in, translates caller's voice to text output on the mobile screen, and displays the emotion reasoning from the caller's voice to visualize emoticons. It also saves voice, translated text, and emotion data that can be played back.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.92-96
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• 2000

In this paper, we propose a call admission control(CAC) scheme fer the mixed voice/data DS-CDMA systems and analyze the Er-lang capacity under the proposed CAC scheme. Voice and data traffics require different system resources based oil their Quality of Service(QoS) requirements. In the proposed CAC scheme, some system resources are reserved exclusively for handoff calls to have high priority Over new calls. Additionally the queueing of both new and handoff data traffics that are not sensitive to delay is allowed. Ar a performance measure for the suggested CAC scheme. Erlang capacity is utilized. For the performance analysis, a four-dimensional Markov chain model is developed. Erlang capacity of a practical IS-95B type system depicts, and optimum values of system parameters such as the number of reservation channels and queue lengths are found with respect to Erlang capacity. Finally, it is observed that Erlang capacity is improved more than two times by properly selecting the system parameters with the proposed CAC scheme.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.10
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• pp.2462-2472
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• 2014

Recently, the computing paradigm has been changing and VoIP technology is being revisited to support various services. In this paper, we have designed and implemented the system of software PBX open source Asterisk, hardware platform, and mobile devices to support secured voice service based on VoIP. Specially, we designed the various platform from single board to servers based on XenServer in hardware platform. And we verified the delay test of network traffics and the secured voice communication test based on this platform.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.1
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• pp.165-175
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• 2023

As the reduction of information exposure threats by organization insiders contributes to achieving information security(IS) goals, organizations are establishing strict IS policies applicable to insiders and increasing investment in IS systems. However, since IS incidents cause damage to an organization even by malicious information exposure by one person, psychological support for strengthening IS compliance behavior by insiders. This study aims to confirm how the uncertain organizational environment related to IS affects individual IS-related behavior. We surveyed insiders of organizations operating IS policies and tested the hypothesis using 440 samples. As a result, IS technology and communication uncertainty reduced IS voice behavior through IS prospective anxiety, and individuals' susceptibility to information influence moderated the relationship between IS technology, communication, and prospective anxiety and IS voice behavior. Our results suggest the necessity and direction of supplementing the uncertain IS environment in practice.

• 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).

• ETRI Journal
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• v.38 no.3
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• pp.444-454
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• 2016

Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN.

• Journal of Advanced Navigation Technology
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• v.21 no.1
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• pp.113-118
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• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.77-83
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• 2013

In this paper, we suggested a method about the improvement of the voice recognition rate and carried out a study on it. In general, voices were detected by applying the most widely-used method, HMM (Hidden Markov Model) algorithm. Regarding the method of detecting voices, the zero crossing ratio was calculated based on the units of voices before the existence of data was identified. Regarding the method of recognizing voices, the patterns shown by the forms of voices were analyzed before they were compared to the patterns which had already been learned. According to the results of the experiment, in comparison with the recognition rate of 80% shown by the existing HMM algorithm, the suggested algorithm based on the recognition of the patterns shown by the forms of voices showed the recognition rate of 92%, reflecting the recognition rate improved by about 12% compared to the existing one.