• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.8
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• pp.331-339
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• 2003

In this paper, we propose an modified TCP congestion control algorithm using estimated RTT with congestion window parameter CWnd. Congestion window size is controlled with memorized RTT value on the congestion status. It can avoid occurrence of frequent congestion and reduce CWnd fluctuation. From simulation results, proposed algorithm shows great improvement on network efficiency and buffer utilization compared with original TCP algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1653-1660
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• 2003

In this paper, we proposed an enhanced TCP congestion control algorithm using RTT with congestion window parameter cwnd to minimize the effect of TCP congestion. The proposed scheme could avoid the occurrence of frequent congestion and decrease the delay caused by the recovery time and the using amount of switch buffer. Through the simulation, we showed that the proposed scheme cm acquire higher performance than the existing scheme. There are 22.56％ improvement at the average using buffer efficiency, and packet drop rate is 0.1％ which is less than existing scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.58-65
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• 2016

Satellite communication is a wide area network (WAN) which provides communication service worldwide. However, the performance of TCP can be seriously degraded in the satellite networks due to limited bandwidth, long round-trip time (RTT) and high bit error rate (BER) over satellite links. In order to improve the performance of TCP, this paper proposes cross-layer Performance Enhancing Proxy (PEP) in digital video broadcasting-return channel via satellite (DVB-RCS) networks. The proposed protocol sets TCP Congestion Window (CWND) size by using satellite resource allocation information exchanged between TCP and the link-layer. we implement PEP testbed based on Linux to evaluate the performance of the proposed protocol. The simulation results show that the proposed protocol performs better than standard TCP both in single and multiple sessions in variant BER, because the proposed protocol sets TCP CWND size by using satellite resource allocation.

• Proceedings of the Korean Information Science Society Conference
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• 2004.10c
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• pp.541-543
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• 2004

TCP does not distinguish between congestion and packet losses due to route change and link failures, which are prevalent in mobile ad hoc networks. So, TCP does not show satisfactory performance in ad hoc networks since it assumes that all packet losses are due to network congestions. In particular, when a route is reestablished it needs to be adaptively determined CWND according to the new route features. In this paper, we proposed CWND adjustment scheme to improve the TCP performance overad hoc networks. TCP sender effectively adjusts CWND by monitoring the network situation using control packets. Simulation results using NS-2 show that the proposed scheme increases TCP throughput compared with those of general TCP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.2B
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• pp.124-132
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• 2007

Where a wireless LAN and a cellular network coexist, a mobile node has to experience vertical handoffs to move between them. Immediately after the vertical handoffs, TCP must need adaptation latency to adjust its congestion window to the proper size at a newly arrived network to use full of a new end-to-end available bandwidth. Even though SACK TCP has the best performance among other regular TCPs in the previous studies, it still cannot use full of the new available bandwidth quickly due to its inefficient increasing way of congestion window. BIC TCP, that becomes a popular TCP in long fat networks, has great feature working well against vertical handoffs by increasing congestion window exponentially with TCP connection sustained. In this paper, we derive adaptation latency of SACK TCP and BIC TCP numerically, and verify them by simulations. We also find that the shorter adaptation latency of BIC TCP produces higher throughput than SACK TCP on vertical handoffs. Consequently, to get higher performance on vertical handoff situations, we propose to use BIC TCP.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.235-244
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• 2004

TCP Vegas version provides better performance and more stable services than TCP Tahoe and Reno versions, which are widely used in the current Internet. However, in the situation where TCP Vegas and Reno share the bottleneck link, the performance of TCP Vegas is much smaller than that of TCP Reno. This unfairness is due to the difference of congestion control mechanisms of each TCP use. Several studies have been executed in order to solve this unfairness problem. In this paper, we analyze the minimum window size to maintain the maximum TCP performance of link bandwidth. In addition, we propose an algorithm which maintains the TCP performance and improves fairness by selective packet drops in order to allocate proper window size of each TCP connections. To evaluate the performance of the proposed algorithm, we have measured the number of data bytes transmitted between end-to-end systems by each TCP connections. The simulation results show that the proposed algorithm maintains the maximum TCP performance and improves the fairness.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1147-1150
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• 2006

Stream Control Transmission Protocol (SCTP) uses the 32-bit checksum in the common header, by which a corrupted SCTP packet will be regarded as a lost packet and then discarded. This may result in degradation of SCTP's throughput performance over wireless networks. This paper proposes a new chunk checksum scheme for SCTP, in which each data chunk contains its own checksum field and SACK chunk carry corresponding Transmission Sequence Number (TSN) with timestamp for every corruption event. The proposed chunk checksum scheme is introduced with the following three purposes: 1) to distinguish the chunk corruptions from the chunk losses; 2) to avoid the unnecessary halving of the congestion window (cwnd) in the case of chunk corruption; 3) to avoid the unwanted timeouts which can be induced in conventional SCTP in the case that the retransmitted data chunks are corrupted again in wireless networks. Simulation results show that the proposed chunk checksum scheme could improve the SCTP throughput in the wireless environments with a high bit error rate.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.271-273
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• 2000

본 논문에서는 TCP 수신자가 지연 ACK 옵션(Delayed ACK Option)을 사용할 경우에 TCP 송신자에게 발생하는 성능 저하요인들을 분석하고 다음과 같은 해결책을 제시하였다. 먼저, 느린 시작 구간(Slow Start phase) 처음에 생기는 ACK 타임아웃은 큰 초기 윈도우(large initial window)또는 1-bit 마킹 기법을 통해 해결할 수 있다. 그리고, 느린 시작 구간과 혼잡 회피 구간(Congestion Avoidance phase)에서 혼잡 윈도우(cwnd)가 천천히 증가하는 문제는 적절히 바이트 카운팅 기법을 사용함으로써 해결할 수 있다. 마지막으로, 송신자가 버스트(burst)한 데이터를 네트웍에 발생시키는 문제는 트래픽을 평활(pacing)함으로써 해결할 수 있다. 또한 본 연구에서는 분석적 모델링을 통하여 TCP가 보내는 평균 전송률을 구하였으며 이 결과는 TCP에 친화한 전송률 기반 전송방법(TCP Friendly Rate Based Control)에 응용될 수 있을 것이다. 그리고 시뮬레이션을 통해서 제시한 방법의 성능이 향상됨을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.8
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• pp.1542-1550
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• 2015

Delayed ACK is an algorithm implemented to decrease the number of ACK transmissions by delaying an ACK transmission and by waiting for additional ACK instead of transmitting the ACK immediately. By using Delayed ACK in TCP, the congestion of network and the overhead of handling ACKs can be reduced. Waiting time of Delayed ACK is defined as the Delayed ACK timeout, and it is fixed in Window OS basically. However, the fixed value of Delayed ACK timeout is not suitable for dynamic network circumstance, and it may cause unnecessary delay. This paper proposes a regulating Delayed ACK timeout algorithm to reduce the aforementioned unnecessary delay caused by the lengthy default value of the Delayed ACK timeout. We confirm that TCP transmission performance in dynamic network circumstance can be improved using the proposed algorithm.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.