• Journal of Communications and Networks
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• v.12 no.5
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• pp.499-509
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• 2010

A plethora of transmission control protocol (TCP) congestion control algorithms have been devoted to achieving the ultimate goal of high link utilization and fair bandwidth sharing in high bandwidth-delay product (HBDP) networks. We present a new insight into the TCP congestion control problem; in particular an end-to-end delay-based approach for an HBDP network. Our main focus is to design an end-to-end mechanism that can achieve the goal without the assistance of any network feedback. Without a router's aid in notifying the network load factor of a bottleneck link, we utilize goodput and throughput values in order to estimate the load factor. The obtained load factor affects the congestion window adjustment. The new protocol, which is called TCP-goodput and throughput (GT), adopts the carefully designed inversely-proportional increase multiplicative decrease window control policy. Our protocol is stable and efficient regardless of the link capacity, the number of flows, and the round-trip delay. Simulation results show that TCP-GT achieves high utilization, good fairness, small standing queue size, and no packet loss in an HBDP environment.

• Journal of Advanced Navigation Technology
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• v.15 no.2
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• pp.241-247
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• 2011

Most of network congestions are caused by TCP packet losses in the wired network ecosystems. On the contrary, high BER (Bit Error Rate), which is characteristic of the wireless networks, is a main factor in wireless network environments. Many wireless networks carry out the congestion control mechanisms frequently because they estimate that packet losses are not due to the wireless signal qualities but the congestion controls in their networks. To solve this problem, we propose the improved EWLN (Explicit Wireless Loss Notification) algorithm that more efficiently utilize the congestion window size to increase the wireless network throughput than legacy EWLN algorithm.

• Journal of Institute of Control, Robotics and Systems
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• v.16 no.8
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• pp.816-822
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• 2010

In this paper, we propose an algorithm to improve TCP performance over wireless links. TCP is known to have poor performance over wireless links because TCP has no mechanism to differentiate congestion loss from wireless loss, and treats all losses as congestive. We present a simple method to determine the cause of packet loss using the successive ECN. In addition, we present an algorithm to control the congestion window size based on the estimated queue state in order to guarantee the fairness and high link utilization.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.81-86
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• 2011

In one of the most impacting schemes proposed to address the TCP throughput problem over network coding, the network coding layer sends an acknowledgement if an innovative linear combination is received, even when a new packet is not decoded. Although this scheme is very effective, its implementation requires a limit on the coding window size. This limitation causes low TCP throughput in the presence of packet reordering. We argue that a TCP variant detecting a packet loss relying only on timers is effective in dealing with the packet reordering problem in network coding environments as well. Also we propose a new network coding layer to support such a TCP variant. Simulation results for a 2-path environment show that our proposed scheme improves TCP throughput by 19%.

• The Journal of the Korea Contents Association
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• v.7 no.1
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• pp.131-138
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• 2007

In wired networks, packet losses mostly occur due to congestion. TCP reacts to the congestion by decreasing its congestion window, thus to reduce network utilization. In wireless networks, however, losses may occur due to the high bit-error rate of the transmission medium or due to fading and mobility. Nevertheless, TCP still reacts to packet losses according to its congestion control scheme, thus to reduce the network utilization unnecessarily. This reduction of network utilization causes the performance of TCP to decrease. In this paper, we predict packet loss by using RSS(Received Signal Strengths) on the wireless and suggest adding RSS flag bit in ACK packet of MH. By using RSS flag bit in ACK, the FH(Fixed Host) decides whether it adopt congestion control scheme or not for the maximum throughput. The result of the simulation by NS-2 shows that the proposed mechanism significantly increases sending amount and receiving amount by 40% at maximum.

• 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 the Korea Society of Computer and Information
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• v.14 no.6
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• pp.73-81
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• 2009

TCP assumes that packet loss is always happened by congestionlike wired networks because is can not distinguish between congestion loss and transmission error loss,. This assumption results in unnecessary TCP performance degradation in wireless networks by reducing sender's congestion window size and retransmitting the lost packets. Also, repeated retransmissions loed to waste the limited battery power of mobile devices. In this paper, we propose the new congestion control scheme that add the algorithms monitoring networks states and the algorithms preventing congestion to improve TCP throughput performance and energy efficiency in wireless ad-hoc networks. Using NS2, we showd our scheme improved throughput performance and energy efficiency.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.2
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• pp.264-270
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• 2006

In this paper we verified that the window based TCP performance of throughput can be improved by the traffic connection efficiency. and have studied the performance of traffic congestion control that is controlling transmission rate. In wireless network, the bidirectional node is run by estimating the usage rate of link of error control idle and the throughput is shown by transmitting segments. The throughput rate shows almost no delay due to the bidirectional traffic connection efficiency up to the allowable point as increasing the transport rate by the critical value, depending on the size of end-to-end node queue of the increase of transport rate. This paper reports the performance improvement as the number of feedback connection traffic congestion control increases because of the increase of the number of asynchronous transport TCP connections.

• IEMEK Journal of Embedded Systems and Applications
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• v.1 no.1
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• pp.24-30
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• 2006

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

• Journal of Korea Multimedia Society
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• v.8 no.1
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• pp.71-78
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• 2005

When a PDA is used as a mobile terminal in an infrastructure network based on 802.11b standard wireless LAN, we measured and evaluated the performance of multimedia data transmission. In general, PDAs are mainly used tot receiving bulk data like multimedia data. However, while a PDA as a mobile host transfers multimedia data to a desktop PC as a fixed host in such a network, the time taken to receive data from the PC to the PDA is always longer than the time taken to send data from the PDA to the PC by at least 53%. Thus, we consider some critical factors that affect the time taken to receive Our experimental results show that with the PDA the time taken to access a file for writing is longer than the time taken to access a file for reading of to 4.11 times. In addition, a dramatic change of the site of the receiver window, from 686 bytes to 32,383 bytes, prolongs the time taken to receive. To transfer data without any inter-packet interval for the PDA and to increase the size of the TCP receive buffer will be effective in improving the delay to receive data.