• Journal of KIISE
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• v.42 no.12
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• pp.1600-1610
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• 2015

While today's networks have been shown to exhibit HBDP (High Bandwidth Delay Product) characteristics, the legacy TCP increases the size of the congestion window slowly and decreases the size of the congestion window drastically such that it is not suitable for HBDP Networks. In order to solve this problem with the legacy TCP, many congestion control TCP mechanisms have been proposed. C-TCP (Compound-TCP) is a hybrid TCP which is a synergy of delay-based and loss-based approaches. C-TCP adapts the decreasing rate of the delay window without considering the congestion level, leading to degradation of performance. In this paper, we propose a new scheme to improve the performance of C-TCP. By controlling the increasing and decreasing rates according to the congestion level of the network, our proposed scheme can improve the bandwidth occupancy and fairness of C-TCP. Through performance evaluation, we show that our proposed scheme offers better performance in HBDP networks as compared to the legacy C-TCP.

• The KIPS Transactions:PartC
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• v.12C no.4 s.100
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• pp.571-580
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• 2005

In fast long-distance networks, TCP's congestion control algorithm has the problem of utilizing bandwidth effectively. Several window-based congestion control protocols for high-speed and large delay networks have been proposed to solve this problem. These protocols deliberate mainly three properties : scalability, TCP-friendliness, and RTT-fairness. These protocols, however, cannot satisfy above three properties at the same time because of the trade-off among them This paper presents a new window-based congestion control algorithm, called EM (Exponential Increase/ Multiplicative Decrease), that simultaneously supports all four properties including fast convergence, which is another important constraint for fast long-distance networks; it can support scalability by increasing congestion window exponentially proportional to the time elapsed since a packet loss; it can support RTT-fairness and TCP-friendliness by considering RTT in its response function; it can support last fair-share convergence by increasing congestion window inversely proportional to the congestion window just before packet loss. We evaluate the performance of EIMD and other algorithms by extensive computer simulations.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.10a
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• pp.387-389
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• 2015

IEEE 802.11e EDCA(Enhanced Distributed Channel Access)는 4개의 AC(Access Category)를 이용하여 트래픽에 따른 우선순위를 부여하고 QoS(Quality of Service)를 제공하기 위해 표준화되었다. EDCA는 이진 백오프 알고리즘을 갖는 CSMA/CA(Carrier Sense Multiple Access with Collision Avoidance) 방법을 이용한다. EDCA에서 패킷 전송에 실패할 경우 경쟁 윈도우 값은 두 배씩 증가 되고, 성공할 경우에는 최소 경쟁 윈도우 값으로 초기화된다. 따라서 경쟁 윈도우 값이 최적의 값을 유지하지 못해 많은 패킷 충돌을 야기하여 네트워크 성능을 감소시킨다. 이 문제를 해결하기 위해 기존에 제안된 논문에서는 패킷 전송 성공 후 경쟁 윈도우 값을 최소 경쟁 윈도우 값이 아닌 채널 혼잡 정도에 따라 계산된 값으로 설정한다. 그러나 이 방법은 트래픽 종류와 상관없이 같은 방법으로 동작하기 때문에 트래픽 종류에 따른 차별적 QoS를 보장하지 않는다. 또한 계산된 경쟁 윈도우 값은 현재 값에 비해 상대적으로 낮은 값을 갖기 때문에 여전히 높은 충돌율을 갖는다. 본 논문에서는 이 문제를 해결하기 위해 새로운 프로토콜을 제안한다. 제안된 방법에서는 네트워크의 혼잡 정도를 잘 반영하기 위한 새로운 경쟁 윈도우 계산 알고리즘을 제시한다. 또한 제안된 알고리즘은 트래픽 종류에 따른 QoS 보장을 위해 트래픽 종류에 따른 차별화 파라미터를 이용한다.

• Journal of Internet of Things and Convergence
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• v.6 no.3
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• pp.1-6
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• 2020

This paper aims to integrate the previous models about mean object transfer latency in one framework and analyze the result through the computational experience. The analytical object transfer latency model assumes the multiple packet losses and the Internet of Things(IoT) environment including multi-hop wireless network, where fast re-transmission is not possible due to small window. The model also considers the initial congestion window size and the multiple packet loss in one congestion window. Performance evaluation shows that the lower and upper bounds of the mean object transfer latency are almost the same when both transfer object size and packet loss rate are small. However, as packet loss rate increases, the size of the initial congestion window and the round-trip time affect the upper and lower bounds of the mean object transfer latency.

• 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 the Korea Institute of Information and Communication Engineering
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• v.17 no.11
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• pp.2553-2562
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• 2013

It is difficult for TCP congestion control algorithm to ensure the bandwidth and delay bound required for media streaming services in broadband wireless network environments. In this paper, we propose the COIN TCP (COncave INcrease TCP) scheme for providing a high-quality media streaming services. The COIN TCP concavely increases the congestion window size by adjusting the increment rate of congestion window, that is inversely proportional to the amount of data accumulated in the router queue. As a result, our scheme can quickly occupy the available bandwidth and prevent the heavy congestion. It also improves the link utilization by adjusting the decrement rate of congestion window according to the packet loss rate with the random loss. Through the simulation results, we prove that our scheme improves the total throughput in broadband wireless network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1139-1145
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• 2010

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.427-434
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• 2003

This paper suggests an improved TCP congestion algorithm, which is more robust to lossy wireless environment than other algorithms such as TCP-Reno. The suggested algorithm decides on the size of a congestion window depending on both PER (Packet Error Rate) and its state, which is one of fast recovery state and slow start state. Some simulations are given to validate the suggested algorithm and the algorithm is compared with other TCP congestion algorithm from the point of view of performance measures such as a congestion window and throughput. The suggested algorithm has better throughput than other algorithm over wireless links with high PER and similar throughput to others over wireless links with low BER.

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.336-343
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• 2002

Nowadays, after appearance of wireless network the existent internet environment is changing into the united wire/wireless network. But the present TCP regards all of the packet losses on transmission as the packet tosses due to the congestion. When it is applied on the wireless path, it deteriorates the end-to-end TCP throughput because it regards the packet loss by handoff or bit error as the packet loss by the congestion and it reduces the congestion window. In this paper, for solving these problems we propose the method that controls the performance of TCP on the wireless link by extending ECN which is used as a congestion control mechanism on the existent wire link. This is the method that distinguished the packet loss due to the congestion from due to bit error or handoff on the wireless network, so it calls the congestion control mechanism only when there occurs the congestion in the united wire/wireless network.