• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2224-2227
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• 2003

본 논문에서는 비동기식 3 세대 이동통신망인 WCDMA의 표준 음성 부호화기인 AMR(Adaptive Multi-Rate)[1]과 VoIP(Voice over Internet Protocol) 응용분야에 최근 널리 활용되고 있는 ITU-T 8kbit/s 0.729A[2]의 효율적인 연동을 위한 상호부호화(transcoding) 알고리듬을 제안한다. AMR은 통신 채널 환경에 따라 4.75kbit/s부터 12.2kbit/s까지 가변 하여 통화품질을 보장한다. 따라서, 제안된 상호부호화 알고리듬은 순방향 8 모드, 역방향 8모드를 합하여 총 16모드를 지원한다. 제안된 알고리듬의 성능 평가를 위해 지연 추정, 연산량 측정과 주/객관적 음질평가를 수행한 결과, 제안한 알고리듬은 기존의 tandem보다 최소 5㎳의 짧은 지연, 평균 50.2％의 적은 연산량으로 우수한 음질의 복호화 음성 신호를 제공함을 확인하였다.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.11 no.2
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• pp.67-72
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• 2001

An efficient encryption mechanism for transmitting voice packets on the Internet was proposed in this study. The VPN approach of encrypting all the packets through a gateway increases delay and delay jitter that may degrade the quality of service (QoS) in real-time communications. A user-controlled secure Internet phone, therefore. was designed and implemented. The secure phone enables the user to apply encryption to his own call when necessary, and reduces security overheads on the gateway.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.107-110
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• 2000

최근 이동전화의 사용이 급격히 확산됨에 따라 편이성과 안정성의 문제로 핸즈프리 전화통신의 필요성이 대두되고 있다. 핸즈프리 통신상황의 경우 근거리에 위치한 스피커와 마이크로폰의 커플링에 의해 발생하는 음향반향과 차량내에 존재하는 배경잡음에 의하여 통화 품질이 크게 저하되는 문제가 발생한다. 본 논문에서는 차량내에서 핸즈프리 전화통신에 적합한 음향반향과 잡음제거 시스템을 제안하였다. 특히, 안정적인 음향반향제거 성능을 얻기 위하여 두 개의 평균 상호상관도를 이용한 동시통화검출 알고리즘을 제안하였다. 음향반향제거를 위해서는 NLMS 알고리즘에 의해 구동되는 제한된 차수의 적응반향제거기법을 사용하였으며, 잔여 반향 및 배경 잡음제거를 위해 IS-127 EVRC음성 부호화기의 잡음제거 방식을 사용하였다. 제안된 시스템은 16비트 고정소수점 DSP인 OAK DSP를 이용하여 약23.17MIPS의 연산량으로 실시간 구현되었다.

• Journal of the Korea Convergence Society
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• v.8 no.12
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• pp.93-99
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• 2017

The elevator system combines various convergence technologies with the development of ICT technology. Emergency call devices which are safety related devices is applied as an obligation of the elevator and those scope also varies. In this paper, we propose an integrated model that overcomes the limitations of existing voice emergency call devices and efficiently manages and manages video call based service structures in VoIP based on wired and wireless environments. This method effectively manages and operates various lift data and video records in the elevator between the manager, the server and the user. And also It is possible to secure the quality of video call in VoIP and cloud service environment and increase the reliability of safety management and enhance various service environment by creating an integrated structure utilizing various data and additional services in the elevator.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.2
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• pp.326-332
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• 1996

In this paper, a method to design the sending loudness rating(SLR) is proposed and the desirable transmission characteristics are considered in order to specify the transmission quality, based on the loudness ratings, for the digital telephone system that is a terminal for digital speech communication. To specify the desirable SLR for digital telephone system, the subjective test defining the preferred range of inout level for CODEC was performed. From the test results, it was identified that the optimal input level for CODEC is -15dB and the range not to cause the quantization noise and the distortion of CODEC fall within -12dB and -18dB.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.04a
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• pp.205-206
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• 2015

스마트폰이 유선전화의 단말기가 되면 스마트폰의 전화번호부를 유선통화에도 사용할 수 있게 되어 유선전화의 사용량이 늘어나고, 이를 통해 사용자는 통신요금이 낮아지는 혜택을 누리며 유선 사업자는 수익 증대를 기대할 수 있다. 그러나 이동통신용 단말기로 제작된 스마트폰을 유선전화의 단말기로 이용하는 데는 i) 사용자 편의성 증대를 위한 스마트폰 등록 절차의 간소화, ii) 저전력 상태와 WiFi 비활성화 상태에서 전화수신 안정화, iii) 무선데이터 전송시간 변이에 의한 음성품질 저하 회피 등 다양한 기술적 문제들이 존재한다. 본 논문에서는 이와 같은 문제들에 대한 해결 방안을 제시하고 이를 반영한 실험용 시스템의 구현 결과를 보여준다. 실험용 시스템은 사설교환기 기능을 내장한 WiFi-AP, 안정적인 전화수신 알림을 위한 PUSH broker, VoIP 어플리케이션이 설치된 스마트폰으로 구성된다. 실험용 시스템은 PSTN 망과 VoIP 망에 연동하여 스마트폰으로 안정적인 통화품질을 제공할 수 있음을 보였다.

• Journal of Korea Society of Industrial Information Systems
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• v.24 no.5
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• pp.105-113
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• 2019

Korean military uses 'Tactical information communication network' to exchange information between units. In this study, we developed a media processing board for multi-party voice and video telephony based on open source software. On the other hand, in order to apply open source software for weapon systems and parts that are mounted on weapon systems, appropriate review is required according to the weapon system software development and management manual of the Defense Acquisition Program Administration (DAPA). In this study, the analysis of the requirement items was performed and the appropriate countermeasures were proposed for the open software applied to the media processing board with respect to 'the guidelines for the application of weapon systems to open source software', an appendix to the DAPA's manual.

• 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.