• 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:Information Networking
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• v.31 no.1
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• pp.91-100
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• 2004

This paper presents a receiver-driven TCP flow control mechanism, which is adaptive to the wireless condition, for memory constrained mobile receiver. A receiver-driven TCP flow control is, in general, achieved by adjusting the size of advertised window at the receiver. The proposed method constantly measures at the receiver both the available wireless bandwidth and the packet round-trip time. Depending on the measured values, the receiver adjusts appropriately the size of advertised window. Constrained by the adjusted window which reflects the current state of the wireless network, the sender achieves an improved TCP throughput as well as the reduced round-trip packet delay. Its implementation only affects the protocol stack at the receiver and hence neither the sender nor the router are required to be modified. The mechanism has been implemented in real environments. The experimental results show that in CDMA2000 1x networks the TCP throughput of the proposed method has improved about 5 times over the conventional method when the receiver's buffer size is limited to 2896 bytes. Also, with 64Kbytes of buffer site, the packet round-trip time of the proposed method has been reduced in half, compared the case with the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.3B
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• pp.487-492
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• 2000

The ABR(available bit rate) service has been developed to support data application over ATM(asynchronous Transfer mode) networks, and TCP is the most widely used transport layer protocol in existing data networks. Thus, the performance improvement of TCP over ABR is an issue of cardinal importance. In this paper, we propose algorithm of the adjustment of congestion window size utilizing RM(resource Management) cell information. The motivation is to use more reliable information in ER(explicit rate) field within RM cell to determine TCP window size. Simulations are performance using the suggested algorithm is improved, as compared to TCP-reno.

• 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 KIEE Conference
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• 2004.11c
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• pp.234-236
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• 2004

Transmission control protocol(TCP) is protocol used in internet. TCP is seldom transmission error and is protocol based on wire environment. TCP uses 3 way handshake ways, data transmission control through windows size, data transmission control through reception confirmation, sliding window for packet delivery. In this study, designed TCP packet ion module for analyze the TCP segments & correct information about TCP. TCP capture in internet using designed TCP module and analysed TCP segments composition. Through this, could analyze the correct information of protocol in network.

• 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.

• 대한공업교육학회지
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• v.32 no.2
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• pp.199-216
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• 2007

Stream Control Transmission Protocol(SCTP) is a transport layer protocol to support the data transmission. SCTP is similar to Transmission Control Protocol(TCP) in a variety of aspects. However, several features of SCTP including multi-homing and multi-streaming incur the performance difference from TCP. This paper highlights the data transfer during the initial slow start phase in SCTP congestion control composed of slow start phase and congestion avoidance phase. In order to compare the mean transfer time between SCTP and TCP, we experiment with different performance parameters including bandwidth, round trip time, and data length. By varying data length, we also measure the corresponding initial window size, which is one of factors affecting the mean transfer time. For the experiment, we have written server and client applications by C language using SCTP socket API and have measured the transfer time by ethereal program. We transferred data between client and server using round-robin method. Analysis of these experimental results from the testbed implementation shows that larger initial window size of SCTP than that of TCP brings the reduction in the mean transfer time of SCTP compared with TCP by 15 % on average during the initial slow start phase.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• 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 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.