• 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.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.

• 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.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.865-874
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• 2002

HTTP (HyperText Transfer Protocol) is a transfer protocol used by the World Wide Web distributed hypermedia system to retrieve the objects. Because HTTP is a connection oriented protocol, it uses TCP (Transmission control Protocol) as a transport layer. But it is known that HTTP interacts with TCP badly. it is discussed about factors affecting the performance or HTTP over TCP, the transaction time obtained by the per-transaction TCP connections for HTTP access and the TCP slow-start overheads, and the transaction time for T-TCP (Transaction TCP) which is one or methods improving the performance or HTTP over TCP. Average waiting time and average turnaround time are important parameters to satisfy QoS (Quality of Service) of end users. Formulas for calculating two parameters are derived. Such formulas can be used for the environment in which each TCP or T-TCP transaction time is same or different. Some experiments and computational experiences indicate that the proposed formulas are well acted, can be applied to the environment which the extension of bandwidth is necessary, and time characteristics of T-TCP are superior to that of TCP. Also, the load distribution method of web server based on the combination of bandwidths is discussed to reduce average waiting time and average turnaround time.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.229-234
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• 2004

In this study, the performance is compared between RFC compatible normal TCP and several speed constraints Ignored TCP. To do these, the main algorithms that constraints the transmit rate of TCP are modified. We have modified TCP protocol stack in a Linux kernel to compare the speeds between the standard TCP and our modified TCP. We find that if the destination is short distance away from the source and packet error is scarce then the speed differences between normal and modified TCP nay be negligible. However, if the destination is far away from the source and slow start algorithm is not adopted then the transfer time for small file is different greatly In addition, if packet error occurred frequently, our modified TCP is faster than the standard TCP regardless of distance.

• The KIPS Transactions:PartC
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• v.17C no.6
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• pp.451-458
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• 2010

To detect and recover packet losses over wireless links is very important in terms of reliability in packet transmission. Most wireless communication systems adopt an automatic repeat request (ARQ) protocol operating at link layer. However, it has been constantly addressed that the interaction not harmonized sufficiently between ARQ and TCP rather degrades TCP performance. In this paper, therefore, we propose an improved scheme from the aspect of the interaction with TCP loss recovery mechanism that can be applied to selective repeat ARQ (SR-ARQ) protocol and prove that the proposed scheme improves TCP performance significantly by OPNET simulations.

• 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.

• The KIPS Transactions:PartC
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• v.8C no.3
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• pp.335-344
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• 2001

최근 인터넷 환경이 반도체, 광통신 그리고 차세대 인터넷 기술의 발달로 고성능화 되어가고 있다. 따라서 고성능 인터넷을 위한 TCP의 성능 향상 연구가 매우 중요해졌다. 그러나 기존 TCP는 수신위도 버퍼의 물리적 크기에 의하여 최대 전송 성능과 대역폭 탐색 기능이 제한을 받는 구조적인 문제점을 갖고 있다. 본 논문에서는 이를 해결하기 위하여 수신 호스트에 가상 윈도 개념을 도입하였다. 이는 송신 호스트가 RTT 동안 균일하게 세그먼트를 분산시켜서 패킷을 전송할 때 세그먼트 간격 시간 동안 수신 호스트의 처리 능력을 가상윈도로 나타내는 것이다. 따라서 가상 윈도의 크기는 수신 호스트의 성능에 비례하기 때문에 수신 호스트가 고성능일 경우 TCP의 전송 능력 성능이 더 높아질 수 있다. 초고속 인터넷일 경우 제안 알고리즘이 기존 TCP보다 전송능력에 있어 1.5∼5배 개선되는 것을 네트워크 시뮬레이션인 NS2를 이용하여 확인하였다.

• The KIPS Transactions:PartC
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• v.12C no.7 s.103
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• pp.1039-1046
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• 2005

TCP SACK is the unique mechanism to reflect the situation of sink's sequence space, some TCP variants and proposals can perform in conjunction with SACK mechanism for achieving optimal performance. By definition of RFC 2018, however, each contiguous block of data queued at the data receiver is defined in the SACK option by two 32-bit unsigned integers in network byte order. Since TCP Options field has a 40-byte maximum length, when error bursts now, we note that the available option space may not be sufficient to report all blocks present in the receiver's queue and lead to unnecessarily force the TCP sender to retransmit Packets that have actually been received by TCP sink. For overcoming this restriction, in this thesis, a new solution named 'one-byte offset based SACK mechanism' is designed to further improve the performance or TCP SACK and prevent those unwanted retransmissions. The results or both theory analysis and simulation also show that his proposed scheme operates simply and more effectively than the other existing schemes by means of the least bytes and most robust mechanism to the high packet error rates often seen in networks environment.

• Journal of the Korean Ceramic Society
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• v.50 no.4
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• pp.275-279
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• 2013

The purpose of this study was to prepare calcium phosphate cement [CPC] for use in artificial bone. Nano-crystalline calcium phosphate [CaP] was precipitated at $37^{\circ}C$ using highly active $Ca(OH)_2$ in DI water and an aqueous solution of $H_3PO_4$. From the XRD measurements, the nano-CaP powder was close to apatitic TCP phase and the powders fired at $800^{\circ}C$ showed a critical ${\beta}$-TCP phase. A mixture of one mole $CaCO_3$ and two moles di-calcium phosphate was calcined at $1100^{\circ}C$ to make a reference ${\beta}$-TCP material. The nano-CaP powders were added to the normal ${\beta}$-TCP matrix and fired at $900^{\circ}C$ to make a ${\beta}$-TCP block. The sintered block showed improved mechanical strength, which was caused by the solid state interaction between nano-CaP and normal ${\beta}$-TCP.