• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1485-1488
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• 2003

IEEE 802.11 무선 랜 환경에서 멀티미디어 트래픽이 효과적으로 전송퇴기 위해서는 정해진 딜레이 바운드내에서 전송이 완료되어야 한다 대표적인 멀티미디어 트래픽인 음성신호를 전송할 때의 단방향 딜레이 바운드는 echo canceller를 쓰지 않았을 경우 $25ms{\sim}30ms$ 이다. 딜레이 바운드를 지키지 못하고 전송된다면 시간에 민감한 음성신호의 특성 때문에 음성품질이 유지되지 않을 뿐만 아니라 채널의 혼잡을 유발하게 된다. 본 논문에서는 음성의 품질이 보장되는 기준을 95%이상의 패킷이 성공적으로 전달되는 경우로 제한하여 음성의 딜레이 바운드에 관한 분석을 시도하였다. 이를 위해 음성패킷이 drop될 확률을 수학적인 분석을 통해 유도하고 시뮬레이션을 통한 검증을 시도하였다. 시뮬레이션에서는 IEEE 802.11의 두 가지 기본적인 MAC(Multiple Access Control) 프로토콜인 DCF와 PCF를 사용해서 음성신호를 전송할 때 딜레이 바운드를 지키지 못하는 음성 패킷을 사전에 drop 시킴으로써 몇 개의 음성 노드가 손실율 5% 이내 (음성의 품질이 유지되는 한계)를 만족시키는지를 음성신호를 발생시키는 STA 수와 손실율의 관계를 통해 알아보았다.

• Journal of Internet Computing and Services
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• v.9 no.4
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• pp.1-10
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• 2008

In this paper, a packet scheduling algorithm, called the modified PLFS, is proposed for real-time traffic in IEEE 802.22 WRAN systems. The modified PLFS(Packet Loss Fair Scheduling) algorithm utilizes not only the delay of the Head of Line(HOL) packets in buffer of each user but also the amount of expected loss packets in the next-next frame when a service will not be given in the next frame. The performances of the modified PLFS are compared with those of PLFS and M-LWDF in terms of the average packet loss rate and throughput. The simulation results show that the proposed scheduling algorithm performs much better than the PLFS and M-LWDF algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1A
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• pp.30-37
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• 2006

In this paper, we propose a QoS-aware efficient bandwidth re-provisioning scheme in a next generation wireless packet transport network. At the transport network layer, it classifies the traffic of the radio network layer into a real time class and a non-real time class. Using an auto-regressive time-series model and a given packet loss probability, our scheme predicts the needed bandwidth of the non-real time class at every re-provisioning interval. Our scheme increases the system capacity by releasing the unutilized bandwidth of the non-real time traffic class for the real-time traffic class while insuring a controllable upper bound on the packet loss probability of a non-real time traffic class. Through empirical evaluations using the real Internet traffic traces, our scheme is validated that it can increase the bandwidth efficiency while guaranteeing the quality of service requirements of the non-real time traffic class.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.11C
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• pp.1068-1075
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• 2005

In this paper we propose a video coding method and associated packetization and decoding methods for error resilient transmission over the Internet. The proposed method re-organizes the input image into several mutually similar subimages. For this case, if the one of the subimage is lost in the network, the lost one is recovered by the proposed error concealment method which uses the correctly received other subimages. The performance of the proposed method is confirmed by the empirical results. The proposed method is not limited to the Internet communications but is applicable to the other packet-based networks.

• Journal of Internet Computing and Services
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• v.8 no.5
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• pp.125-132
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• 2007

Transmission mechanisms that include an available bandwidth estimation algorithm and a packet loss differentiation scheme, in general, exhibit higher TCP performance in wireless networks. TCP New Jersey, known as the best existing scheme in terms of goodput, improves wireless TCP performance using the available bandwidth estimation at the sender and the congestion warning at intermediate routers. Although TCP New Jersey achieves 17% and 85% improvements in goodput over TCP Westwood and TCP Reno, respectively, we further improve TCP New Jersey by exploring improved available bandwidth estimation, retransmission timeout, and recovery mechanisms. Hence, we propose TCP New Jersey PLUS (shortly TCP NJ+), showing that under 1% packet loss rate, it outperforms 3% by TCP New Jersey and 5% by TCP Wes1wood. In 5% packet loss rate, a characteristic of high bit-error-rate wireless network, it outperforms other TCP variants by 19% to 104% in terms of goodput even when the network is in bi-directional congestion.

• 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 Transactions of the Korea Information Processing Society
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• v.5 no.8
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• pp.2102-2116
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• 1998

본 논문에서는 TCP/IP 프로토콜을 기반으로 하는 인터넷에서 SNMP의 MIB-II를 활용하여 분석 파라미터를 정의하고 이를 계산하는 알고리즘을 제안하고자 한다. TCP/IP 망의 사용자에게 적절한 QoS를 제공하기 위해서는 성능과 장애에 관련된 파리미터를 기반으로 한 망 관리 행위를 수행해야 한다. 이를 위해서 인터넷 관리 표준을 정의된 MIB-II의 관리 정보를 기반으로 분석 파라미터를 정의하고 이를 계산하기 위한 알고리즘을 제시하고자 한다. MIB-II에서 system, interface, ip, snmp 그룹의 관리 변수를 Case 다이어그램에 따라 분석하여 선로 이용률, 에러 수신율, 인터페이스 패킷 송수신율, 인터페이스 패킷 송수신 손실률, 입출력 트래픽률, 방송형 송수신 트래픽 비율, 시스템 패킷 입출력률, 시스템 패킷 송수신 손실률, 시스템 자원 부하율, 패킷 전달률, 경로 설정 실패율, 관리 트래픽 이용률 등의 분석 파라미터와 계산 알고리즘을 제안한다. 분석 파라미터 계산 알고리즘에 대한 적용성을 실험하기 위해서 실존하는 라우터를 대상으로 분석 결과를 제시하고 진단하였다. 이러한 인터넷 분석 파라미터 계산 알고리즘은 망 관리자가 전체 TCP/IP 통신 네트워크를 진단하고 분석할 수 있는 자료로 활용될 수 있을 뿐만 아니라 인터넷 사용자에게 QoS를 제공하루 수 있을 것으로 기대된다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.3
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• pp.1-7
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• 2012

Cognitive radio, a new paradigm for wireless communication, is being recently expected to support various types of multimedia traffics. To guarantee Quality of Service (QoS) from SUs, a static packet priority policy can be considered. However, this approach can easily satisfy Quality of Service of high priority application while that of lower priority applications is being degraded. In the paper, we propose a fuzzy-based dynamic packet scheduling algorithm to support multimedia traffics in which the dynamic packet scheduler modifies priorities of packets according to Fuzzy-rules with the information of priority and delay deadline of each packet, and determines which packet would be transmitted through the channel of the primary user in the next time slot in order to reduce packet loss rate. Our simulation result shows that packet loss rate can be improved through the proposed scheme when overall traffic load is not heavy.

• The KIPS Transactions:PartB
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• v.9B no.5
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• pp.571-578
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• 2002

In this paper, we propose a video sequence coding scheme called AMV (Auxiliary Motion Vector) to minimize error propagation caused by transmission errors over the Internet. Unlike the conventional coding schemes the AMY coder, for a macroblock in a frame, selects two best matching blocks among several preceding frames. The best matching block, called a primary block, is used for motion compensation of the destination macroblock. The other block, called an auxiliary block, replaces the primary block in case of its loss at the decoder. When a primary block is corrupted or lost during transmission, the decoder can efficiently and simply suppress error propagation to the subsequent frames by replacing the block with an auxiliary block. This scheme has an advantage of reducing both the number and the impact of error propagations. We implemented the proposed coder by modifying H.263 standard coding and evaluated the performance of our proposed scheme in the simulation. The simulation results show that AMV coder is more efficient than the H.263 baseline coder at the high packet loss rate.