• 방송과미디어
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• 제19권1호
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• pp.48-55
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• 2014

오디오 신호 압축 기술 관점에서 최근 모바일 통신 시장의 가장 큰 변화 중에 하나는 광대역 음성 코덱인 AMR-WB가 HD voice 또는 VoLTE(Voice over Long Term Evolution) 서비스에 사용된다는 것이다. 모바일 방송 및 음악 재생뿐만 아니라, 통화에서도 다양한 신호를 고음질로 압축해야 하는 필요성이 대두되어 3GPP에서 EVS(Enhanced Voice Service) Codec이 표준화 진행 중이다. 본 논문에서는 실감 통화를 위해 초광대역 신호까지 압축하는 음성 통화 코덱 기술 및 3GPP표준화 현황을 소개한다. 3GPP 표준 기수에서 정의한 디자인 및 성능 요구사항과 더불어 이전 음성 압축기술과의 차이점 등을 논의한다. 또한, 향후 표준화 계획 및 시장 전망에 대해서 논의한다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -1
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• pp.208-211
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• 2002

At present there is no hardware design system by using the voice. We try to use the voice communication to the hardware design and to communication with the computer. This time, we make au application that introduces voice communication and we execute small-scale circuit description.

• Journal of information and communication convergence engineering
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• 제1권4호
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• pp.205-208
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• 2003

Voice activity detection is a important problem in the speech recognition and speech communication. This paper introduces new feature parameter which are reconstructed by spectral entropy of information theory for robust voice activity detection in the noise environment, then analyzes and compares it with energy method of voice activity detection and performance. In experiments, we confirmed that spectral entropy and its reconstructed parameter are superior than the energy method for robust voice activity detection in the various noise environment.

• International Journal of Internet, Broadcasting and Communication
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• 제11권3호
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.

• 한국인터넷방송통신학회논문지
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• 제13권6호
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• pp.29-35
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• 2013

우리나라 재난관리 기관에서는 재난발생시 대응활동을 위해 이종의 재난통신 무전기를 사용하고 있다. 이러한 무전기들은 통신 방식에 따라 각기 다른 통신접속지연을 보이므로 이들 간에 원활한 음성통신을 위해서는 시간차에 따른 보정기술이 필요하다. 이에 본 논문에서는 아날로그와 디지털 그리고 직접 통신과 기지국 통신에 따른 각각의 통신지연에 따른 무전기간 음성 전송권 획득 및 음성데이터의 손실 문제를 해결하기 위한 방안을 제시하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.229-232
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• 2000

AAL2 has been adopted for bandwidth-efficient trans-mission of low bit tate traffic over ATM networks in ITUT and ATM Forum. Since ATM/AAL2 is expected to be used as a switching technology in third-generation mobile access networks and mobile data traffic is expected to increase rapidly in near future, there must be a need for efficient scheduling scheme satisfying the QoS requirement of ow bit rate voice as well as the one of high bit rate data. In this paper, we propose a new class-scheduling scheme to improve data packet loss probability, while Qos of voice traffic is guaranteed, when data traffic is multiplexed together with mobile voice traffic into a single ATM VCC. The proposed scheme can efficiently support data traffic by assigning a time threshold value to voice traffic. Through simulation study, we show that the proposed scheme does not only achieve better efficiency for providing both mobile voice and data services than HOL class-scheduling scheme and normal FIFO scheme, but also guarantees mean voice packet delay under a certain criteria.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2000년도 추계학술대회 학술발표 논문집
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• pp.471-475
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• 2000

Voice and data are transmitted over a single packetized data communications network which is designed for data communications. The public switched telephone network for voice and the packet data network for data are merging into a single data network to get efficiency and to reduce operational cost. However, integrating voice and data transmission over a single data network is not easy because voice should be transmitted without delay but data should be transmitted without error. Advances in technology begin to overcome basic differences. Several integration methods in voice and data will be examined and reviewed here. Moreover, trends and problems on integration will be also discussed.

• 한국철도학회:학술대회논문집
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• 한국철도학회 2008년도 춘계학술대회 논문집
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• pp.1962-1967
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• 2008

A event recorder system stores a train speed and the related information for train operation in real time. Using those information, we can analysis the train operation and the reason of train accident. Currently the event recorder only manipulate the data related the train operation mechanically and electrically. In this paper we propose the event recorder to record the voice communication between the manager in the control center and train operator. By recording the voice communication in the high speed train, the correctness of analysis of train accident can be increased. The system architecture of the event recorder with voice recording is studied and interface between other equipment is proposed. And the software architecture of new event recorder is developed. We study the method of converting analog voice signal into digital data and compressing method. Also the architecture of memory to store the compressed voice data and regeneration of original analog voice are studied.

• 한국전자통신학회논문지
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• 제10권3호
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• pp.319-325
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• 2015

무선 통신 기술에 있어서 눈부신 발전은 서로 멀리에 있는 사람들 간에 통신을 가능케 했다. 최근에는, 저비용과 단순한 하드웨어를 기반으로 하는 무선 시스템을 이용한 근거리 음성커뮤니케이션의 필요성이 급속히 대두되고 있다. 그러나, 이러한 애플리케이션은 작은 지역에서 같은 무선 채널로 다중 이용자(multi-users)들이 통신하는 것을 요구하기 때문에, 현존하는 음성 기술들은 이러한 애플리케이션들에 직접적으로 적용하는 것은 적절치가 않다. 본 논문에서는 다중 이용자 음성통신을 가능케 하는 참신한 아이디어를 제안하고자 한다 특히, 단거리 무선 해결책으로서, 저전력, 저비용을 바탕으로 한 IEEE 802. 15. 4를 이용한다. 그러나, 원래 그 표준은 음성 통신을 위해 개발된 것은 아니기 때문에, 음성 통신에 적당하도록 GTS의 확장 개념을 이용함으로써 원래의 운용계획을 확장한다. 제안된 운용계획의 용량과 타당성은 다양한 음성 압축비에서 양적 분석을 통해 평가된다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 ICEIC The International Conference on Electronics Informations and Communications
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• pp.387-390
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• 2004

In this paper, we design and implement practical application service using VoiceXML. And we suggest new solutions of problems can be occurred when implementing a new systems using VoiceXML, based on the fact. Up to now, speech related services were developed using API (Application Program Interface) and programming languages, which methods depend on system architectures. It thus appears that reuse of contents and resource was very difficult. To solve these problems, nowadays, companies develop their applications using VoiceXML. Advantages of using VoiceXML when developing services are as follows. First, we can use web developing technologies and technologies for transmitting web contents. And, we can save labors for low level programming like C language or Assembler language. And we can save labors for managing resources, too. As the result of these advantages, we can reduce developing hours of applications services and we can solve problem of compatibility between systems. But, there's poor grip of actual problems can be occurred when implementing their own services using VoiceXML. To overcome these problems, we implemented interactive stock trading system using VoiceXML and concentrated our effort to find out problems when using VoiceXML. And then, we proposed solutions to these problems and analyzed strong points and weak points of suggested system.