• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.39-44
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• 2002

In this paper, in order to improve the performance of TCP short traffic application services in wireless Internet environments, the Split-ACKs (SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless lints results in congestion control procedure of TCP being invoked at the source. This causes severe end-to-end Performance degradation of TCP. In this paper, to alleviate the TCP Performance, the SPACK method, split acknowledgement Packets in the base station, is proposed. Using computer simulation, the performance of TCP using SPACK is analysed and shows better performance than traditional TCP Protocol.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.04a
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• pp.7-16
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• 2000

초고속 통신의 실현과 다양한 멀티미디어 서비스를 위한 TCP/IP 통신의 전송 메카니즘으로서 ATM이 널리 사용되고 있다. TCP층과 ATM층의 상호작용이 TCP층 뿐만 아니라 ATM층에도 영향을 미쳐 처리율과 공정성의 성능이 저하되므로, ATM상의 TCP 트래픽 파라미터, 즉 TG(timer granularity)와 윈도우 크기를 적절하게 조정함으로써 TCP 흐름제어가 ABR 율제어와 상호작용하지 않는다. 그 결과로 상호작용의 영향을 최소화할 수 있으며, TCP의성능향상을 가져올 수 있다. 본 논문에서는 윈도우 크기와 TG와 같은 TCP 파라미터를 조절하여 VBR서비스상에서의 TCP 트랙픽의 성능평가를 통해 높은 처리율과 공정성을 나타내었다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11a
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• pp.429-432
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• 2003

최근 TCP와 유사하게 멀티미디어 데이터를 전송하기 위한 rate control 프로토콜에 관한 연구가 활발하게 진행되고 있다. 본 논문에서는 멀티미디어 전송 방식에 있어서 TCP와 공정성을 유지하며 TCP와 유사하게 동작하도록 하는 TRCP 프로토콜을 제안한다. TRCP는 TCP vegas 이론을 응용하였으며 RTT와 패킷손실율에 의해 네트워크 혼잡 상태를 미리 예측하고 TCP의 AIMD 방식을 사용하여 TCP와 유사하게 전송율을.조절하는 프로토콜이다. 이 프로토콜에 대한 TCP와 공정성을 검증하기 위하여 시뮬레이션을 실시하고 그 결과를 분석한다.

• Journal of Advanced Navigation Technology
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• v.12 no.2
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• pp.154-163
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• 2008

In a mobile broadband wireless access (MBWA) network, many users access a base station (BS), which relays data transferred from high-speed wired network to low-speed wireless network. For this difference of their data rate, a BS suffers from the lack of its buffer space when many users run multiple applications at the same time, and thus packet losses occur. TCP, which guarantees end-to-end reliability, is used as transport protocol also in wireless networks. But TCP lowers their transmission rate incorrectly and frequently whenever packet losses occur. And they increase their transmission rate differently with each other; finally TCP throughput of each TCP flow varies largely, and then TCP fairness goes worse. In this paper, a scheme that controls packet transmission rate adaptively according to TCP flows' transmission rate, that prevents buffer overflows at BS, and that guarantees TCP fairness at a certain degree is proposed. As it is analyzed by simulations, the proposed scheme enhances TCP fairness by maintaining TCP throughput of each TCP sender similarly with each other.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.667-674
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• 2004

The performance of transmission control protocol (TCP) is largely dependent upon its loss recovery. Therefore, it is a very important issue whether the packet losses may be recovered without retransmission timeout (RTO) or not. Although TCP SACK can recover multiple packet losses in a window, it cannot avoid RTO if a retransmitted packet is lost again. In order to alleviate this problem, we propose a simple change to TCP SACK, which is called TCP SACK+ in simple. We use a stochastic model to evaluate the performance of TCP SACK+, and compare it with TCP SACK. Numerical results evaluated by simulations show that SACK+ can improve the loss recovery of TCP SACK significantly in presence of random losses.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.4
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• pp.35-44
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• 2004

Recently announced TCP Vegas predicts the degree of congestion in the network and then control the congestion window size. Thus it shows better performance than TCP Reno. however, TCP vegas does not assume any network model, its congestion window control is very limited. Because or this limitation, TCP vegas still can not adapt to fast changing available bandwidth. In this paper, we introduce a new TCP algorithm which adapts to fast changing available bandwidth well. To devise such a TCP, we model the end to end network of TCP connection as a queueing system and finds congestion window size which can utilize the available bandwidth sufficiently but not make the network congested. The simulation results show that our algorithm adapts to the avaliable bandwidth faster than TCP vegas and as a results, when the available bandwidth is changing rapidly, our algorithm not only operates more stably than TCP Vegas, but also it shows higher thruput than TCP Vegas.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.10
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• pp.491-498
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• 2005

The Denial of Service(DoS) attacks, which are done by consuming all of the computing or communication resources necessary for the services, are known very difficult to be protected from. TCP has drawbacks in its connection establishment for possible DoS attacks. TCP maintains the state of each partly established connection in the connection queue until it is established completely and accepted by the application. The attackers can make the queue full by sending connection requests repeatedly and not completing the connection establishment steps for those requests. In this paper, we have designed and implemented the extended TCP for preventing from SYN Flood DoS attacks. In the extended TCP, the state of each partly established connection is not maintained in the queue until the connection is established completely. For the extended TCP, we have modified the 3-way handshake procedure of TCP and implemented the extended TCP in the Linux operating system. The test result shows $0.05\%$ delay more than original TCP, but it shows that the extended TCP is strong for SYN Flood attacks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12C
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• pp.1239-1247
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• 2005

It is important to improve TCP performance in Self-similar TCP network where signalling between the same end nodes through bidirectional traffic routes. In wireless link, the traffic limitation pattern occurred in two or more TCP connections is applied into MPEC video control as multi time-interval congestion control. For TCP update variable, we extend TCP and perform as function call, and we study a method of relating TCP with LTS module controlling with the information type that is overcoming the limit of feedback loop determined by RTT. For comparison, we measure the TCP throughput without LTS and verify the fairness by means of meta control. The improved TCP performance is shown by that the number of connections of traffic congestion control increases when RTT increases.

• Journal of KIISE
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• v.42 no.9
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• pp.1162-1174
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• 2015

These days, the networks have exhibited HBDP (High Bandwidth Delay Product) characteristics. The legacy TCP slowly increases the size of the congestion window and drastically decreases the size of a congestion window. The legacy TCP has been found to be unsuitable for HBDP networks. TCP mechanisms for solving the problems of the legacy TCP can be categorized into the loss-based TCP and the delay-based TCP. Most of the TCP mechanisms use the standard slow start phase, which leads to a heavy packet loss event caused by the overshoot. Also, in the case of congestion avoidance, the loss-based TCP has shown problems of wastage in terms of the bandwidth and RTT (Round Trip Time) fairness. The delay-based TCP has shown a slow increase in speed and low occupancy of the bandwidth. In this paper, we propose a new scheme for improving the over shoot, increasing the speed of the bandwidth and overcoming the bandwidth occupancy and RTT fairness issues. By monitoring the buffer condition in the bottleneck link, the proposed scheme does congestion control and solves problems of slow start and congestion avoidance. By evaluating performance, we prove that our proposed scheme offers better performance in HBDP networks compared to the previous TCP mechanisms.

• The Transactions of the Korea Information Processing Society
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• v.7 no.5
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• pp.1520-1524
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• 2000

TCP congestion control algorithm prevents network congestion through the control of outgoing traffic size. The network, therefore, should monitor the incoming traffic size of a TCP to determine whether or not a TCP follows standard congestion control algorithms. Some TCP friendly test algorithms are proposed, But, these algorithms cannot be used in real environments because a router in a network does not know the RTT of a TCP flow. In this study, we propose a new RTT determination algorithm that can be used in a router. Our proposed algorithms is validated through the simulation studies.