• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.721-724
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• 2011

UDT(UDP-based Data Transfer protocol)는 연구망과 같은 대용량 고속 네트워크에서 신속하고 안정적인 데이터 전송에 효율적인 전송 프로토콜이다. 하지만 네트워크가 혼잡한 상황이거나 데이터 패킷 손실이 발생할 경우 매우 공격적인 전송 메커니즘으로 인해 오히려 성능이 감소하는 경우가 발생한다. 본 논문에서는 네트워크 지연 예측을 통해 혼잡을 제어하는 TCP-Vegas 기법을 UDT에 적용하여 네트워크 상황에 보다 적응적인 UDT 혼잡제어 기법을 제안한다. 즉, 네트워크 라우터 큐에 빌드된 Diff 값을 미리 예측하고 Diff 값에 따라 UDT rate control을 조절하여 기존 방법보다 향상된 성능을 얻는다. 네트워크 시뮬레이터인 NS2를 통해 실험한 결과 패킷 손실이 없는 상황에서 최대 11%, 손실률 0.01%인 상황에서는 최대 31%의 성능 향상을 확인하였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.717-720
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• 2011

고속 네트워크 환경에서 AR UDT(Adaptive Rate control UDT)는 표준 전송 프로토콜인 TCP에 비해 뛰어난 성능을 보인다. UDT(UDP-based Data Transfer)를 기반으로 하는 AR UDT의 혼잡제어는 네트워크 상태를 예측하여 패킷 간 전송시간을 변화시킴으로써 기존 UDT보다 향상된 성능을 보인다. 그러나 AR UDT는 네트워크 상태 예측의 오차가 클 뿐만 아니라 rate control만을 공격적으로 조절하기 때문에 수신 버퍼의 초과로 인해 안정적인 성능을 기대하기 어렵다. 본 논문에서는 AR UDT환경에서 네트워크 상태에 따라 적응적으로 혼잡제어를 하는 기법을 제안한다. RTT(Round Trip Time)의 변화량에 따라 네트워크 상태를 예측하여 flow control과 rate control을 적응적으로 조절한다. 네트워크 시뮬레이션 결과를 통하여 AR UDT에 비해 전송속도와 안정성이 향상되었음을 보였다.

• Journal of Korean Society of Transportation
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• v.17 no.2
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• pp.179-191
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• 1999

To device effective freeway traffic flow control strategies and freeway traffic information dissemination strategies, it is very important to estimate real physical queue length on the freeway. Shock wave theory and queueing theory have limitation to be used to estimate the queue length. The primary objective of this study is to develop a reliable method for estimating the physical queue length and level of congestion. Queueing propagation processes were analysed by using such traffic data as main line traffic volume, ramp volume, density. speed, and physical queue length collected by video photographing on Olympic Freeway. As a result of analysis, it has been confirmed that the real queue length can be estimated by using the traffic counts arriving the congested region and passing a bottleneck location. Further more, a reliable method for estimating the level of congestion could be developed on the basis of real-time traffic counts.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.20 no.6
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• pp.100-109
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• 2021

Flooding usually brings in disruptions and aggravated congestions to the roadway network. Hence, right information should be provided to road users to avoid the flood-impacted areas and for city officials to recover the network. However, the information about individual link congestion may not be conveyed to roadway users and city officials because too many links are congested at the same time. Therefore, more significant information may be desired, especially in a disastrous situation. This information may include 1) which places to avoid during flooding 2) which places are feasible to drive avoiding flooding. Hence, this paper aims to develop a framework to identify the flood-impacted areas in a roadway network and their criticality. Various impacted clusters and their spatiotemporal properties were identified with field data. From this data, roadway users can reroute their trips, and city officials can take the right actions to recover the affected areas. The information resulting from the developed framework would be significant enough for roadway users and city officials to cope with flooding.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.11a
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• pp.978-981
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• 2007

TCP(Transmission Control Protocol)는 흐름제어와 오류제어 그리고 혼잡제어 등을 통해 목적지까지 효율적으로 데이터를 전송하는 프로토콜이다. 최근 라우터 성능 및 라우팅 알고리즘 등의 발달로 인해 패킷이 비순차적으로 전송되는 패킷 재배치(packet reordering)가 빈번한 발생하여 TCP 성능 저하의 주요 원인이 되고 있다. 패킷 재배치가 TCP 로 하여금 불필요한 재전송(Spurious Retransmission) 및 혼잡제어를 수행하도록 하기 때문이다. 본 논문에서는 TCP 의 신속 재전송(Fast Retransmission)의 중복 응답 임계값(dupthresh)를 동적으로 조절함으로써, 불필요한 재전송 및 혼잡제어의 발생횟수를 효율적으로 줄이는 TCP-DAD(TCP Dynamically Adjusted Dupthresh)기법을 제안한다. 본 논문은 NS-2를 이용한 시뮬레이션을 통해 TCP-DAD 가 다른 TCP 메커니즘보다 평균적으로 약 34%의 성능 향상을 보임을 증명하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.340-342
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• 2005

최근 기지국(Base Station)의 도움 없이 이동 단말기(Mobile Device) 간의 다중 무선 홉을 사용하여 송수신자 간의 데이터 전송을 가능하게 하는 Ad-hoc 통신망에 관한 연구가 활발히 진행되고 있다. Ad-hoc 통신망을 구성하는 이동 단말은 이동성을 고려하여 한정된 배터리로 동작을 하게 된다. Ad-hoc 통신망에서 이동 단말은 종단(End Host)으로만 동작하는 것이 아닌 중계 노드(Intermediate Node)로도 동작을 하기 때문에 통신상에 경로를 제공하는 중요한 역할도 한다. 하지만, 현재 인터넷 상에서 광범위하게 사용되고 있는 전송 규약인 TCP(Transmission Control Protocol)는 수동적인(Reactive)혼잡 제어(Congestion Control)방식으로 망의 혼잡으로 인한 패킷 손실 발생 이전까지 전송 윈도우의 크기를 증가 때문에 반복적인 혼잡과 그로 인한 패킷 손실로 인해 불필요한 재전송을 반복하게 된다. 이와 같이 기존 TCP는 무선 통신망에서 동작하는 이동 단말의 한정된 배터리 전원을 고려하지 않고 동작하기 때문에 이동 단말의 에너지를 불필요하게 낭비하는 문제를 가지게 된다. 본 논문은 Ad-hoc 망에서 이동 단말의 에너지 효율을 개선하기 위해 불필요한 재전송 방지 및 망 상태에 따라 전송률을 적절하게 조절하는 TCP의 새로운 혼잡 제어 기법을 제시하였다. 또한 ns-2 시뮬레이터를 이용한 실험을 통해 이동 단말의 에너지 효율이 제안된 혼잡 제어 기법에 의해 향상되었음을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11B
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• pp.900-910
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• 2004

Route failure is mainly caused by mobility of mobile host in ad hoc networks. Route failure, which may lead to sudden packet losses and delays, is losing the route from source to destination. In this situation, TCP assumes that congestion has occurred within the network and also initiates the congestion control procedures. Congestion control algorithm provides the means for the source to deal with lost packets. TCP performance in ad hoc environments will be degraded as TCP source cannot distinguish congestion from route failure. In this paper, we propose TCP-P as pre-detection approach to deal with route failure. TCP-P freezes TCP through pre-detection of route failure. Route failure information of the proposed mechanism is obtained not by routing protocol but by MAC protocol. The intermediated node, obtaining route failure information by its MAC layer, relays the information to TCP source and lets TCP source stop the congestion control algorithm. Results reveal that TCP-P responding with proactive manner outperforms other approaches in terms of communication throughput under the presence of node mobility.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.527-531
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• 2003

Previous researches on scalability problem of distributed virtual environment (DVE) have been mainly focused on spatial partitioning of area of interest (AOI). Congestion phenomena by avatar groups in AOI have been neglected relatively. However, AOI congestion is highly related to scalability of DVT because it exhausts system resources such as network bandwidth and rendering time, and could be a bar to perform collaboration among participants. In this paper, this will be defined as the problem that must be solved for the realization of the scalable DVE, and a model will be proposed to measure and control congestion situation in AOI. The purposes of the proposed model are to prevent high density of participants in AOI, and to protect stable collaboration in DVE. For evaluation of the performance it is compared with a previous method by defining the resource cost model which is dynamically activated to AOI congestion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.477-487
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• 2008

Congestion control of the TCP reduces transmission rate when it detects packet loss because packet loss origines from congestion in the wired network. In the wireless network, packet loss comes from channel errors. Wired TCP degrades performance when there are wireless losses because it does not classify type of loss. These day, there are many researches which classify type of loss between congestion loss and wireless loss for wired-wireless hybrid network. For wireless TCP, many of existing algorithms are based on the estimated bandwidth or variations of packet arrival time. In this paper, we propose a new TCP scheme to distinguish the wireless packet losses from the congestion packet losses using MIB of the IEEE 802.11 MAC. We perform excessive simulations using the NS-2 network simulator and analyze the simulation results to compare the performance of the proposed algorithm to other well-known algorithms. From simulation results, we know that proposed algorithm improves performance about 12% and 32% compared with Spike algorithm and mBiaz algorithm, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1599-1609
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• 2014

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm. Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.