• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.511-512
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• 2008

EPLA 패킷 스케줄링 알고리즘은 IEEE 802.22 WRAN 시스템의 실시간 트래픽 전송 지원을 위한 패킷 스케줄링 알고리즘으로 참고문헌[4]으로 제안되었다. 패킷 기반 무선 전송 시스템에서 실시간 트래픽의 경우 짧은 데이터 유효 시간을 가지며, 만약 데이터 유효 시간이 초과할 경우 실시간 트래픽 데이터로써의 가치를 상실하기 때문에 시스템에서는 해당패킷을 전송하지 않고 제거해 버린다. 그러므로 실시간 트래픽의 중요한 서비스 품질(QoS) 인자인 요구된 패킷 손실율을 보장하기 위해서는 실시간 트래픽의 데이터 유효 시간을 고려하여 자원을 할당하여야 한다. 기존의 패킷 스케쥴링 알고리즘들은 많은 경우 큐의 맨 앞에 위치한 패킷의 지연 시간을 고려하지만 EPLA는 패킷이 저장되는 큐 내의 다음 프레임에서 제거 될 것으로 예상되는패킷의 손실양을 고려하여 자인을 할당함으로 기존의 실시간 패킷 스케줄링 알고리즘에 비해 훨씬 좋은 성능을 보인다. 하지만 EPLA는 예상되는 패킷 손실양을 계간하기 위해서 모든 사용자의 큐에 저장된 패킷들을 확인해야하므로 높은 복잡도를 가지는 문제점이 있다. 본 논문에서는 각 사용자로부터 피드백 받은 부채널의 상태 정보를 기반으로 사용자 큐를 확인하여 횟수를 제한함으로써 패킷 손실을 성능의 손실 없이 복잡도를 줄이는 간단한 방법을 제안하고, 실시간 트래픽인 음성 트래픽과 비디오 트래픽에 대한 시뮬레이션 결과를 통해 이를 확인한다.

• Journal of Internet Computing and Services
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• v.6 no.4
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• pp.115-124
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• 2005

The packet loss problem that occurs in the mobile multicast (MoM) protocol due to designated multicast service provider(DMSP) handoff is investigated through simulation experiments for several DMSP selection policies. Then, two enhanced DMSP schemes are proposed to minimize the packet loss of the MoM protocol with single DMSP. The first scheme uses a backup DMSP and greatly reduces the packet loss rate at the expense of the increased network traffic. The second scheme utilizes the extended DMSP operation and shows many desirable features such as the almost zero packet loss rate and relatively low network traffic.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.6
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• pp.1327-1332
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• 2015

In this paper, we studied several aspects that affect the quality of VoIP using WiFi network. It's clear that the background data traffic within an AP, the end-to-end delay and the traffic loss of TCP/IP network gives serious effects on the voice quality. A kind of access control for the VoIP connection within an AP should be done for the acceptable voice quality.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.10
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• pp.24-33
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• 2001

Delay, delay variation and packet loss rate are principal QoS(Quality of Service) elements of packet communication. This paper proposes a new buffer allocation mechanism to improve the packet loss performance in such a situation that multiple logical buffers share a single physical memory buffer. In the proposed buffer allocation mechanism, the movement of dynamic threshold follows a curved track instead of a straight line which is used in the DT(dynamic threshold) mechanism. In order evaluate the effectiveness of the proposed mechanism, it is compared with the existing previously proposed mechanisms in several aspects including NC(no control), ST(Static Threshold) and DT mechanisms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.8A
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• pp.858-870
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• 2007

TCP is popular protocol for reliable data delivery in the Internet. In recent years, wireless environments with transmission errors are becoming more common. Therefore, there is significant interest in using TCP over wireless links. Previous works have shown that, unless the protocol is modified, TCP may perform poorly on paths that include a wireless link subject to transmission errors. The reason for this is the implicit assumption in TCP that all packet losses are due to congestion which causes unnecessary reduction of transmission rate when the cause of packet losses are wireless transmission errors. In this paper, we propose a new LDA that monitors the network congestion level using ROTT. And we evaluate the performance of our scheme and compare with TCP Veno, Spike scheme with NS2(Network Simulator 2). In the result of our experiment, our scheme reduces the packet loss misclassification to maximum 55% of other schemes. And the results of another simulation show that our scheme raise its transmission rate with the fairness preserved.

• Journal of Advanced Navigation Technology
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• v.13 no.4
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• pp.523-531
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• 2009

In this study, we propose a joint source/channel rate control algorithm for video encoder targeting packet erasure channel. Based on the buffer constraints of video communication systems, encoding rate constraint is presented. After defining source distortion models for coded and skipped video frames and a channel distortion model for packet errors and their propagation, an average distortion model of received video is proposed for a given encoding window. Finally, we define an optimization problem to minimize the average distortion for given channel rates and packet loss rates by controlling spatio-temporal parameters of source video and FEC block sizes. Then, we propose a window-based algorithm to solve the problem in real-time.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.19-22
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• 1997

음성이나 동화상 같은 실시간 정보를 인터넷상으로 전송하는 인터넷방송 시스템은 인터넷 서비스의 중요한 이슈중에 하나지만 구현시에 가장 커다란 문제가 되는 것은 패킷 손실이다. 인터넷라디오나 AOD에서 발생하는 손실된 패킷을 복구하기 위해 여러 가지 방법이 제안되고 있으나, 현재 인터넷폰 등을 위해서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘이 가장 효과적으로 사용되고 있으나 단방향 통신이면서 음질이 가장 중요한 요소인 방송시스템에서는 그대로 적용하기에 문제가 있다. 본 논문에서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘에 사용될 수 있는 인터리빙 방법을 제안하였다. 연구결과, 인터리빙 처리를 위하여 추가적인 지연이 발생하는 단점은 있지만 연집 에러를 효과적으로 분산시켜 패킷 손실 복구율을 크게 개선시킬 수 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2004.10c
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• pp.313-315
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• 2004

본 논문은 IEEE 802.l1b 유무선 환경 에 서 TCP를 이용한 데이터 전송 시 에이젼트를 이용하여 패킷 손실의 원인을 분석, 무선 링크에서 발생한 패킷 손실에 대해서는 혼잡 윈도우 크기를 유지하고, 유선 링크에서 발생한 패킷 손실에 대해서는 지역 재전송을 수행하는 저전력 전송 기법을 제안하고 실제 구현한다. 제안하는 저전력 전송기법 은 전송 후 WNIC를 저 전력 모드로 전환 시킴으로써 WNIC 전력 소비를 최소화 한다. 실험 결과 높은 무선 링크 에러율(1~2%)에서 기존 TCP-Reno 보다 약 18% 에너지 감소 효과를 나타냈다.

• Journal of Korea Society of Industrial Information Systems
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• v.10 no.3
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• pp.48-56
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• 2005

This paper proposes a dynamic end-to-end flow control algorithm that is more effective than previous methods considering either the Rate of Packet Loss (RPL) or Round Trip Time (RTT). When the RPL is under normal conditions, the current network status will be in one of three defined states by using the RTT and this makes bandwidth control precise before serious packet loss occurs. If the RPL exceeds a critical level, then the network is considered to be in a fourth state. Suitable transmission rates are determined depending on the network status and are controlled by adjusting not only the number of transmitted frames but also the quality of the frames. In this paper, we present some experimental results of the proposed algorithm. According to our quantitative analysis, the proposed method performs 1.6 to 6 times better than the previous method in terms of the RPL. At the same time, the total number of transmitted packets is increased, which indicates that the proposed method can provide greater bandwidth capacity than previous methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.9
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• pp.700-706
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• 2013

In this paper, we propose a method for applying random linear network coding in satellite communication to improve reliability. In the proposed protocol, network-coded redundancy (NC-R) packets are transmitted in the PEP (Performance Enhancement Proxy). Therefore, if data packets is lost by wireless channel error, they can be recovered by NC-R packets. We also develop the TCP performance model of the proposed protocol and evaluate the performance of the proposed protocol. In the simulation results, It is shown that the proposed protocol can improve the TCP throughput as compared with that of the conventional TCP because the NC-R packets is sent by the sender-side PEP and the receiver-side PEP use these packets to recover the lost packets, resulting in reducing the packet loss in TCP.