• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.45-57
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• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.05a
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• pp.1286-1287
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• 2007

무선 네트워크에서 TCP Vegas는 다양한 원인으로 발생하는 패킷 손실로 인해 전송률이 떨어진다. 다수의 패킷 손실이 발생하면 TCP Vegas의 혼잡제어 기법은 패킷 손실로 인해 혼잡윈도우가 증가되지 않는다. 본 논문은 TCP Vegas의 혼잡제어 기법을 수정하여 무선 네트워크에서 TCP Vegas의 성능을 향상시킨 wirelessVegas를 제안한다. wirelessVegas는 NS-2를 이용하여 시뮬레이션 한 결과 TCP Vegas 보다 1%의 무선 링크 에러율에서 약 20% 높은 성능을 보인다.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.10b
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• pp.1013-1016
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• 2000

기존의 end-to-end 방식에서는 네트워크 내부에서 혼잡(congestion)이 발생했을 경우 각 endpoint에서 즉시 알아 낼 수 없기 때문에 일정시간 동안 수신된 packet의 순서에 대한 정보로 혼잡이 발생했는지에 대해 추론하는 것이다. 따라서 네트워크 내부에서 직접적으로 정보를 얻거나 처리를 해 줄 수 있다면 기존 방식보다 처리율이 향상될 것이다. 본 논문에서는 Active Network 메커니즘을 이용하여 라우터를 구성하고 관련 모의실험결과를 보인다. 이를 통해서 기존의 방식보다 향상된 혼잡제어를 제시한다.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.562-564
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• 2005

비대칭 율이 높은 비대칭 망에서 동작하는 TCP의 성능은 상향 링크의 혼잡 발생 여부에 좌우된다. 상향 링크에 실리는 응답 패킷 트래픽이 링크의 대역폭을 넘어서게 되면 응답 패킷이 손실되거나 전송 지연이 길어져서 성능을 저하시킨다. 상향 링크의 혼잡 문제를 해결하는 대표적인 방법으로 응답 패킷 필터링/재생성 기법을 들 수 있다. 본 논문에서는 해당 기법의 문제점을 해결한, 송신측 TCP의 상태에 따라 응답 패킷 재생성 작업을 수행하는 동적 패킷 재생성 기법을 제안한다. 이 기법은 정확한 응답 패킷을 재생성하고, 이를 순서 뒤바뀜 없이 송신측에 전송하도록 하여 혼잡 제어 구간에서의 성능을 향상시킨다. 또, TCP가 혼잡 제어 상태에 있는 동안에만 응답 패킷 재생성 작업을 수행하도록 하여 불필요한 응답 패킷 트래픽과 패킷 재생성에 드는 부하를 줄인다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1B
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• pp.154-161
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• 2010

In wireless sensor networks, due to the many-to-one convergence of upstream traffic, congestion more probably appears. The existing congestion control protocols avoid congestion by controlling incoming traffic, but the duty-cycle operation of MAC(Medium Access Control) layer has not considered. In this paper, we propose DCA(Duty-cycle Based Congestion Avoidance), an energy efficient congestion control scheme using duty-cycle adjustment for wireless sensor networks. The DCA scheme uses both a resource control approach by increasing the packet reception rate of the receiving node and a traffic control approach by decreasing the packet transmission rate of the sending node for the congestion avoidance. Our results show that the DCA operates energy efficiently and achieves reliability by its congestion control scheme in duty-cycled wireless sensor networks.

• 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.

• Journal of Korean Society of Transportation
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• v.27 no.6
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• pp.7-18
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• 2009

In 2008, there are 135 service areas, which provide for drivers to convenience on expressways. But there are some problems on parking capacity, because it is failed to calculate the appropriate parking lots of the service area. The fundamental problem is that the size of parking lots is calculated by simple value; despite there are various factors of utilization, congestion, and turnover. This paper describes a study on proper coefficient for calculating the appropriate service area, considering the characteristics of drivers' behavior. We conducted statistical analysis with the surveyed data in "Status of the expressway facilities in 2004." Analysis results show there are some difference by day and night, and vehicle type with the factor of utilization, congestion, and turnover, and there are different usages according to expressway route. The results indicate that the usage of a service area have different characteristics by time and space, and vehicle type.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.837-840
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• 2003

Recently the multimedia streaming traffic such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic through the Internet, does not apply any congestion control mechanism to regulate the data flow through the shared network. If this trend is let go unchecked, these traffic will effect the performance of TCP, which is used to transport data traffic, and may lead to congestion collapse of the Internet. To avoid any adverse effort on the current Internet functionality, A study on a new protocol of modification or addition of some functionality to existing transport protocol for transmitting streaming traffic in the Internet is needed. TCP-frienly congestion control mechanism is classified with window-based congestion control scheme and rate-based congestion control scheme. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and rate-based congestion control for multimedia streaming in the internet.

• 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.