• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• The Journal of the Acoustical Society of Korea
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• v.17 no.1
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• pp.86-97
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• 1998

본 논문은 스테레오 음향 반향 제거기에 적용되는 고속 Affine Projection 알고리듬 을 제안한다. 최근 스테레오 원격 회의 시스템은 보다 현실감 있는 원격 회의를 가능케 하 는 장점으로 인해 많은 관심을 끌고 있다. 그러나, 회의실의 원단화자와 마이크로폰사이의 상호교차(cross-coupling)로 인해 음향 반향이 발생하게 된다. 만약 이 반향 신호가 제거되 지 않은채 수신 룸으로 전달되면 결국 음성 통화 품질이 저하된다. 이를 방지하기 위하여 추정 반향 신호를 만들어 내고 통신 품질의 손실 없이 이 반향을 제거하는 음향 반향 제거 기가 필수적이다. 단 채널 음향 반향 제거기와 다르게 스테레오 환경하에서의 음향 반향 제 거기는 전송실의 환경변화로 인한 성능 저하와 각 반향 경로를 추정하기 위해 사용하는 각 적응 필터의 임펄스응답이 반향 경로와 일치하지 않는 등의 각종 문제점들이 발생하게 된 다. 본 논문에서는 서로 상관관계 없는 입력신호를 만들어내고 전송실의 환경변화로 인한 성능저하를 보완하기 위해 전처리단(pre-processing block)을 제안하여 일반적인 방법에 대 해 3-10dB정도의 향상된 성능을 보이며 적은 계산량으로 빠른 수렴성능을 갖는 새로운 형 태의 스테레오 음향 반향 제거기를 제안한다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.242-249
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• 2003

When special voice control application services (VCS) such as voice-controlled web browsing or voice-controlled stock transactions are introduced in cellular systems, a channel quality better than that for ordinary voice communications service (OVS) is necessary in order to keep a suitable grade of VCS. To avoid ai. congestion, calls are normally admitted if there exists a channel-processing resource not occupied by other calls in the base as well as the interference level at the receiver is not higher than a predefined threshold. The threshold is usually 10㏈ noise-rise over the background noise level for voice communications service. When the base admits VCS attempts in exactly the same manner as it handles OVS calls. the same fraction of those will be not successful in taking the channel and then blocked. If the same noise-rise threshold is used as 10 ㏈, however, the admitted VCS calls might suffer from bad channel qualify and finally be dropped. From the user's point of view, the forced termination of ongoing calls is significantly undesirable than blocking new call attempts. When using a lower noise-rise threshold for VCS. on the other hand, the blocking probability of VCS gets higher than that of OVS. In this paper, a call admission policy that gives a priority to VCS is considered in order to reduce the blocking probability and keep an adequate channel quality.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.4
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• pp.53-58
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• 2001

In this paper to develop objective speech quality measure for CDMA telephone network environments, recent developed measures are investigated first. But those measures show low performances in CDMA telephone networks. To solve this problem, new objective speech quality measure adopting noise masking threshold is proposed and studied. To acquire better performance, scaled noise masking threshold calculation for speech signals is employed instead of conventional tone signals. To verify effectiveness of proposed method performance comparison experiments are carried out for CDMA telephone network speech databases, for the results proposed methods show improved performances compared to existing meaures.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• The Proceedings of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.7 no.3
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• pp.49-56
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• 1993

This paper describes design and implementation of signal transmission system using LCX as communication media, which has characteristics of high reliability easy for expansion and complex transmission of voice, data and video signal in Electric culvert. In this system, we estimated system performance as result of variable transmission characteristics test. In case of voice signal, transmission loss characteristics improved 5-10(dB] than designed Values in received signal level. In the test of speech quality estimation, we obtained satisfactory result as speech intensity = 3 (QSA value), speech atriculation = 4 (QRK value). As for data and video signal transmission, communication success rates were 981% 1 in monitoring and control functional test. As a result of transmission characteristics test in transmission line and system, transmission range by LCX communication system without repeater can reach in 6Km. This paper presents basic construction method using LCX communication system for total management in Electric culvert.

• Journal of Digital Contents Society
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• v.19 no.4
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• pp.661-671
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• 2018

The avoidance tendency of voice call in younger generation facilitates the growth of mobile O2O service that utilizes Messaging. This study adopted and elaborated original concepts of telephone apprehension to apply phone anxiety to O2O. We examined previous researches related to the ubiquity of service quality, communication ability and anxiety, and then conducted a survey to understand the consciousness of mobile O2O service users. The results show that users with high message intimacy and users with poor logicality are afraid of voice calls. The mobile O2O service is considered useful as a means to replace an unstable situation. In this study, we considered individual characteristics of acceptance factor of O2O mobile service technology, and it was a new attempt to expand telephone apprehension characteristic reflecting current situation. In addition, the different results of this research from prior studies that examined the relationships among apprehension, usefulness, and continuous use intentions might be helpful to expand and sophisticate the research area.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.945-951
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• 2007

Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.1
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• pp.81-89
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• 2005

In this paper, we have described the implementation of a practical simulation tool to design and analyze communication networks. Especially, this study is focused on the implementation and application methods of a simulator supporting VoIP The key characteristics of this particular system are its easy and intuitive usage, the real behaviors implementation of equipment and protocols, the actual generation and transmission of traffic for simulation, supporting of VoIP and so forth. Our system is distinguished from the existing tools which define only the nature of voice traffic, process those packets in the same way as general data, and analyze only the quality of packet transmission such as delay. Our tool presented in this paper generates and processes packets in different way according to the types of traffic distinguishing call signal from voice information traffic. Also, we equipped this system with the various devices such as VoIP gateway and gatekeeper, which enabled this system to analyze the performance of devices and the quality of voice traffic transmission between PSTN and Internet. By presenting the implementation methods and application of this system, we managed to propose the utilization scheme of a simulation tool.