• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.4
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• pp.125-132
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• 2015

PCI Express is high-speed point-to-point serial protocol, the system is designed by analysing loss and jitter through Eye Diagram. It is necessarily analyzing high speed serial signals when the PCI Express Gen3 which has 8Gbps physical signal speed is designed especially. This paper deals with topology extraction, channel analysis, extraction of s-parameters and system signal integrity simulation within transceiver buffer models through PCI Express Gen3 server connecting switch system design. Optimal parameters of transmitter buffer equalizer are found through solution space simulation of de-emphasis and preshoot parameters to compensate channel loss.

• Proceedings of the IEEK Conference
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• 2002.06c
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• pp.111-114
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• 2002

This thesis introduces a buffer management method that could be used to provide better sound quality for Internet phone terminals. Proposed method used the fuzzy system for dynamically adjust the number of jitter This method actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management method and this intelligent method in various network settings.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.365-371
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• 2015

In wireless network, the amount of video streaming packet information in receiver replay buffer can be varied according tothe wireless network condition. By the effect, unforeseeable delay and jitter are generated and then busty video traffics can be made. If the amount of buffer information coming in receiver replay buffer is larger than the amount of a specific buffer information, buffer overflow is generated. Such a problem makes the image skip effect and packet loss, and then causes the quality degradation and replay discontinuity of the video streaming service in destination receiver. To solve the buffer overflow problem, this paper applies the token bucket for the busty traffic to the receiver terminal and analyzes the effect of the token bucket. The simulation result using NS-2 and JSVM shows that the proposed scheme with the token bucket has significantly better performance than the conventional scheme without the token bucket in terms of overflow generation number, packet loss rate and PSNR.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.953-960
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• 2002

Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

• Proceedings of the IEEK Conference
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• 2001.06b
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• pp.281-284
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• 2001

This paper describes an analog multi-phase delay-locked loop (DLL) to solve the harmonic lock problem using current-starved inverter and shunt-capacitor delay cell. The DLL can be used not only as an internal clock buffer of microprocessors and memory It's but also as a multi-phase clock generator for gigabit serial interfaces. The proposed circuit was simulated in a 0.25${\mu}{\textrm}{m}$ CMOS technology to solve harmonic lock problem and to realize fast lock-on time and low-jitter we verified time interval less than 40 ps as the simulation results.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.430-433
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• 2003

In this paper, we propose the method of voice packet transformation for the improvement of voice quality in embedded VoIP system terminals. For this purpose, it was analyzed about the RTP header in the voice packet receiving side and designed about the handling method of voice packets by using jitter buffer. Through this analyzed and designed results, by implementing the voice packet transformation method and testing on the self-product VoIP system, we reached that a conclusion is satisfied with performance of VoIP services.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.04a
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• pp.87-89
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• 2015

최근 빠른 속도와 사용자의 증가로 VoIP 기술에 대한 관심이 높아지고 있다. 본 논문에서는 VoIP의 개요와 VoIP 환경에서 전송 품질에 영향을 주는 요소인 지터의 정의와 지터를 개선하기 위한 지터 버퍼에 대해 설명한다. 지터 버퍼는 크게 고정형과 적응형, 두 가지로 나누어지며 지터 버퍼에 대한 분석을 통해 지터 버퍼 성능 향상 방안을 제안한다.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.649-655
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• 2000

Real-time application programs have constraints which need to be met between media-data. These constraints represents the delay time ad quality of service between media-data to be presented. In order to efficiently describe the delay time and quality of service, a new synchronization mechanism is needed. Proposed paper is a dynamic synchronization that minimized the effects of adaptive transmission delay time. That is, the method meets the requirements of synchronization between media-dat by handling dynamically the adaptive waiting time resulted from variations of delay time. In addition, the mechanism has interval adjustment using maximum delay jitter time. This paper decreases the data loss resulted from variation of delay time and from loss time of media-data by means of applying delay jitter in order to deal with synchronization interval adjustment. Plus, the mechanism adaptively manages the waiting time of smoothing buffer, which leads to minimize the gap from the variation of delay time. The proposed paper is suitable to the system which requires the guarantee of high quality of service and mechanism improves quality of services such as decrease of loss rate, increase of playout rate.