• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1139-1145
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• 2010

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

• 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.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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• pp.1482-1484
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• 2010

A new TCP protocol can succeed for large bandwidth delay product when it meets network bandwidth utilization efficiency and fair sharing. We introduce a novel congestion control algorithm which employs queueing delay information in order to calculate the amount of congestion window increment in increase phase, and reduces congestion window to optimal estimated bound as packet loss occurs. Combination of such methods guarantees that the proposal utilizes fully network bandwidth, recovers quickly from packet loss in wireless link, and preserves fairness for competing flows mixed short RTT and long RTT. Our simulations show that features of the proposed TCP meet the desired requirements.

• Journal of Communications and Networks
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• v.6 no.4
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• pp.376-386
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• 2004

In ad hoc networks, loss-based congestion window progression by the traditional means of duplicate ACKs and timeouts causes high network buffer utilization due to large bursts of data, thereby degrading network bandwidth utilization. Moreover, network-oriented feedbacks to handle route disconnection events may impair packet forwarding capability by adding to MAC layer congestion and also dissipate considerable network resources at reluctant intermediate nodes. Here, we propose a new TCP scheme that does not require the participation of intermediate nodes. It is a purely end-to-end scheme using TCP timestamps to deduce link conditions. It also eliminates spurious reductions of the transmission window in cases of timeouts and fast retransmits. The scheme incorporates a receiver-oriented rate controller (rater), and a congestion window delimiter for the 802.11 MAC protocol. In addition, the transient nature of medium availability due to medium contention during the connection time is addressed by a freezing timer (freezer) at the receiver, which freezes the sender whenever heavy contention is perceived. Finally, the sender-end is modified to comply with the receiver-end enhancements, as an optional deployment. Simulation studies show that our modification of TCP for ad hoc networks offers outstanding performance in terms of goodput, as well as throughput.

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.427-434
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• 2003

This paper suggests an improved TCP congestion algorithm, which is more robust to lossy wireless environment than other algorithms such as TCP-Reno. The suggested algorithm decides on the size of a congestion window depending on both PER (Packet Error Rate) and its state, which is one of fast recovery state and slow start state. Some simulations are given to validate the suggested algorithm and the algorithm is compared with other TCP congestion algorithm from the point of view of performance measures such as a congestion window and throughput. The suggested algorithm has better throughput than other algorithm over wireless links with high PER and similar throughput to others over wireless links with low BER.

• 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.

• 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 the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.304-309
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• 2004

Recently, the needs for multimedia stream 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 thorough the Internet, does not apply any congestion control mechanism to regulate the data flow thorough the shared network And it leads to congestion collapse of the Internet and results in long-term transmission delay. To avoid any adverse effect on the current Internet functionality, a now protocol of modification or addition of some functionality to perform congestion control and to reduce huge transmission delay in transmitting of multimedia stream are in study. TCP-friendly congestion control mechanism is classified into two : one is window-based congestion control scheme using general window management functionalities, the other is rate-based congestion control scheme using TCP modeling equation. 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 late-based congestion control for multimedia stream. And we also simulate the performance of improved TEAR implementation using NS. With He simulation results, we show that the improved TEAR can provide better fairness and lower rate fluctuations than TCP.