• Journal of Korea Multimedia Society
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• v.9 no.5
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• pp.681-690
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• 2006

With the rapid development of network and multimedia technologies, computer mediated communication has been realized and there has been a great potential to use and research on asynchronous voice communication systems. This paper reports a case study where 6 groups(3 for documentation work, 3 for software development) of 24 people who used voice mail when carrying out their projects. The purpose of this study is to obtain an overall understanding of usability of voice mail which is a typical example of asynchronous voice communication systems, under a particular situation where project works are performed. Through the study, we came to understand general purposes of usage of voice mail, patterns of using it revealed during the project process, different ways of using it according to different types of projects, and reasons why people are currently not likely to use voice mail. The results hopefully provide systems developers with a guideline to understand the nature of voice mail from users' perspectives.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.34-38
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• 2012

The recent increase in use of the IEEE 802.15.4 standard for wireless connectivity in personal area networks makes of it an important technology for low-cost low-power wireless personal area networks. Studies showed that voice communications over IEEE 802.15.4 networks is feasible by Guaranteed Time Slot (GTS) allocation; but there are some constraints to accommodate voice transmission beyond two hops due to the excessive transmission delay. In this paper, we propose a GTS allocation scheme for bidirectional voice traffic in IEEE 802.15.4 multihop networks with the goal of achieving fairness and optimization of resource allocation. The proposed scheme uses a greedy algorithm to allocate GTSs to devices for successful completion of voice transmission with efficient use of bandwidth while considering closest devices with another factor for starvation avoidance. We analyze and validate the proposed scheme in terms of fairness and resource optimization through numeral analysis.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.2
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• pp.171-178
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• 2016

An aircraft in flight and ATC(: Air Traffic Controllers) working in the Ground Control Center carry out a voice communication using the radio. Voice signal to be transmitted from the aircraft is received to a plurality of terrestrial sites around the country at the same time. The ATC receives the various quality of voice signal from the aircraft depending on the distance, speed, weather conditions and adjusted condition of the antenna and the radio. The ATC carries out a voice communication with aircraft in the optimal conditions finding the best voice signal. However, the present system chooses the values of the CD(: Carrier Dectect) which is determined to be superior to, based on the input voice level, as optimal channel. Thus this system can not be seen to select the optimal channel because it doesn't consider the effect of the noise which influences on the communication quality. In this paper, after removing the noise in the voice signal, we could give the digitized information and an improved voice signal quality, so that users can select an optimal channel. By using it, when operating the training eavesdropping system or the aircraft control, we can expect prevention accident and improvement of training performance by selecting the improved quality channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1841-1854
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• 1993

In this paper, the packet voice protocol for the transmission of voice signal onto ethernet is implemented in a personal computer (PC). The packet voice protocol used is a modified one from CCITT G.764 packetized voice protocol. The hardware system to facilitate the voice communication onto ethernet is divided into telephone interface, speech processing, PC interface and controllers. The software structure of the protocol is designed according to the OSI seven layer architecture and is divided into three routines : ethernet device driver, telephone interface, and processing routine of the packet voice protocol. Experiments through ethernet with telephone interface show that this packet voice communication achieves satisfactory quality when the network traffic is light.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.199-204
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• 2013

Google's android-based voice interface is limited to the web application and the users are rare. In this paper, We suggest the method that can be done using existing android-based voice engine and develope voice application. We also study the environments of android-based voice interface and present the appropriate voice interface in mobile environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.17-22
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• 2013

Heterogeneous radios are used in disaster management agencies for a variety of reasons though the radio must have the same radio frequency and protocol for voice communication. For this reason, the variety of heterogeneous radio voice connection methods have been studied but these are simple analog voice line cross connection or partial networked based on digitalization. In this paper, we suggest the method of voice packet transmission method based on All-IP per radio through IP network using SIP/RTP for scalability and openness and developed a prototype of the proposed method was verified.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.12
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• pp.1321-1328
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• 2015

A radio is a half-duplex voice communication method using the PTT(: Push To Talk), occupy a single line calls during transmission. As an interface between the telephone and the radio, UHF and VHF, for voice communication between the different heterogeneous devices, A device automatically switches between the two devices is required. Therefore, in accordance with the performance of the voice switching apparatus for detecting a voice to be transmitted from an input signal, loss of the audio signal to be transmitted is subjected to Significant influence. Conventional method has the problem responding to noise by setting the level through simple means of amplitude of input signal, in other words, the energy level of the input signal. This paper, by using the audio signal processing techniques, this discriminated what the voice is among the input signal and substantiated a device for the automatic voice transmission between heterogeneous devices. With this proposal, I was confirmed of improvement of performance in the automatic voice switching device, could perform loss-less transmission of voice between heterogeneous devices.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.1
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• pp.107-114
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• 2021

By analyzing the voice communication performance of the existing tactical FM radios, the performance target of the newly developing TICN combat network radio system VHF band tactical voice communication waveform was derived. In addition, a vocoder and modulation method that can satisfy the performance target and additional requirements are presented, and the expected voice communication quality is analyzed.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.73-76
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• 2006

유비쿼터스 컴퓨팅에서 중요한 기술 중 하나로 평가되는 음성인식 및 합성기술은 인간과 컴퓨터의 상호 작용에 있어 가장 편리하고 보편적인 방법이다. 음성인식 및 합성기술을 이용한 인간과 컴퓨터 상호작용 기반의 애플리케이션의 개발을 위해 음성 확장성 생성 언어(VoiceXML)을 이용하면 음성 인식 및 합성에 관한 전문 지식이 없어도 애플리케이션 제작을 쉽게 할 수 있다는 장점이 있어서 음성인식 및 합성기술의 인프라 구축과 저변 확대를 목적으로 일부 국내 업체들은 VoiceXML을 이용한 음성 애플리케이션을 제작하고 실험할 수 있도록 VoiceXML 실험 환경을 제공한다. 본 논문에서는 기존에 공개된 실험 환경을 소개하고, 다양한 실험 환경 제공을 위해 기존에 있던 Linux기반의 실험 환경과는 다른 Windows NT기반의 IIS(Internet Information Service)를 이용한 VoiceXML실험 환경을 제안하고 구현하였다. 그 결과 ASP(Active Server Page)와 ADO(ActiveX Data Object)를 이용한 VoiceXML음성 애플리케이션 실험이 가능한 환경을 구축하였고, 사용자 평가 결과 제안한 방법이 유효하다는 것을 확인하였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04b
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• pp.1177-1180
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• 2001

전화를 통해 인터넷 정보를 들을 수 있는 보이스 포탈(Voice Portal) 서비스가 인기를 얻고 있다. Voice Portal 서비스란 알고자 하는 정보를 Speech Recognition System에 음성으로 명령하면 전화를 통해 음성으로 원하는 정보를 듣는 서비스이다. Authentication의 절차를 수행하는 SMS (Short Message Service) 서버 Module, PSTN과 Database 서버사이의 Interface를 제공하는 CTI (Computer Telephony Integration) Module, CTI 서버와 WWW (World Wide Web) 사이의 Voice XML Module, 정보를 검색하기 위한 Searching Module들이 필요하다. 본 논문은 Speech Recognition technology를 기반으로 한 CTI Module 설계를 구현하였다. 또한 인정 방식으로 Random한 일회용 password를 기반으로 한 SMS Authentication을 택하므로 더욱 더 안정된 서비스 제공을 목적으로 하였다.