• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.652-657
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• 2004

As the real-time multimedia applications through Internet increase, the bandwidth available to TCP connections is oppressed by the UDP traffic, result in the performance of overall system is extremely deteriorated. Therefore, developing a new transmission protocol is necessary. The TCP-friendly algorithm is an example meeting this necessity. The TCP-friendly (TFRC) is an UDP-based protocol that controls the transmission rate based on the available round transmission time (RTT) and the packet loss rate (PLR). In the data transmission processing, transmission rate is determined based on the conditions of the previous transmission period. If the one-step ahead predicted values of the control factors are available, the performance will be improved significantly. This paper proposes a prediction model of transmission rate control factors that will be used for the transmission rate control, which improves the performance of the networks. The model developed through this research is predicting one-step ahead variables of RTT and PLR. A multiplayer perceptron neural network is used as the prediction model and Levenberg-Marquardt algorithm is used for the training. The values of RTT and PLR were collected using TFRC protocol in the real system. The obtained prediction model is validated using new data set and the results show that the obtained model predicts the factors accurately.

• IEMEK Journal of Embedded Systems and Applications
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• v.10 no.4
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• pp.241-255
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• 2015

Recent network environment has been rapidly evolved to cloud computing environment based on the development of the Internet technologies. Furthermore there is an increasing demand on mobile cloud computing due to explosive growth of smart devices and wide deployment of LTE-based cellular networks. Thus mobile Desktop-as-a-Service(DaaS) could be a pervasive service for nomadic users. In addition, video streaming traffic is currently more than two-thirds of mobile traffic and ever increasing. All such trends enable that video streaming in mobile DaaS could be an important concern for mobile cloud computing. It should be noted that the performance of the Transmission Control Protocol(TCP) on cloud host servers greatly affects Quality of Service(QoS) of video streams for mobile users. With widely deployed Linux server platforms for cloud computing, in this paper, we conduct an experimental analysis of the twelve Linux TCP variants in mobile DaaS environments. The results of our experiments show that the TCP Illinois outperforms the other TCP variants in terms of wide range of packet loss rate and propagation delay over LTE-based wireless links between cloud servers and mobile devices, even though TCP CUBIC is usually used in default in the current Linux systems.

• 대한공업교육학회지
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• v.33 no.2
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• pp.287-299
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• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• Journal of Internet Computing and Services
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• v.1 no.2
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• pp.1-10
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• 2000

ATM network technology is generally used for the solution of integrating multimedia service in high-speed Internet. In Internet protocol based on ATM services, If single cell is lost in ATM layer, the entire TCP packet will be lost. Therefore, TCP performance will be degraded. In order to reduce cell loss, when congestion occur, UBR+EPD mechanism is proposed to improve the throughput in TCP over UBR, and ER scheme is suggested in TCP over ABR. In this paper, we analyzed the performance improvement effect of UBR+EPD with FRR (Fast Retransmission and Recovery), the adjusting EPD threshold parameter (R), and variation of MTU (Maximum Transport Unit) size. As a result, through the analysis of performance, we know that the improved throughput and fairness are shown by the proposed scheme.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.5
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• pp.1013-1027
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• 2011

In this paper we propose ATCS, a practical TCP protocol coding scheme based on network coding for satellite IP networks. The proposal is specially designed to enhance TCP performance over satellite networks. In our scheme, the source introduces a degree of redundancy and transmits a random linear combination of TCP packets. Since the redundant packets are utilized to mask packet loss over satellite links, the degree of redundancy is determined by the link error rates. Through a simple and effective method, ATCS estimates link error rates in real time and then dynamically adjusts the redundant factor. Consequently, ATCS is adaptable to a wide range of link error rates by coding TCP segments with a flexible redundancy factor. Furthermore, the scheme is compatible with traditional TCP variants. Simulation results indicate that the proposal improves TCP performance considerably.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.413-420
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• 2006

HMIPv6 is designed to reduce the signaling load to external network and improve handover speed of MN by including Mobility Anchor Point(MAP) in local handover. However in this case of macro handover, it's just used pervious MIPv6 handover algorithm. So, it occurs packet loss and transmission delay problem. In this paper, we propose the mechanism applying the HMIPv6 for Fast Handover to choose suitable to the condition buffering handover. The condition for the selection is result distance measurement between MN and CN, between MN and NAR. Furthermore, using F-SNOOP protocol, it is possible to improve wireless network performance. Wireless network has high Bit Error Rate(BER) characteristic because of path loss, fading, noise and interference. TCP regards such errors as congestion and starts congestion control. This congestion control makes packet transmission rate low. However, F-SNOOP improves TCP performance based on SNOOP and Freeze TCP that use Zero Window Advertisement(ZWA) message when handoff occurs in wireless network.

• 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 KIISE:Information Networking
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• v.34 no.6
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• pp.435-445
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• 2007

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem: it cannot perform local retransmission efficiently under the bursty-error prone wireless link. To solve this problem, SACK-Aware-Snoop and SNACK mechanism have been proposed. These approaches improve the performance by using SACK option field between base station and mobile host. However in the wireless channel with high packet loss rate, SACK-Aware-Snoop and SNACK mechanism do not work well because of two reason: (a) end-to-end performance is degraded because duplicate ACKs themself can be lost in the presence of bursty error, (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily because of SACK option field in the wireless link. In this paper, we propose a new local retransmission scheme based on Cross-layer approach, called Cross-layer Snoop(C-Snoop) protocol, to solve the limitation of previous localized link layer schemes. C-Snoop protocol includes caching lost TCP data and performing local retransmission based on a few policies dealing with MAC-layer's timeout and local retransmission timeout. From the simulation result, we could see more improved TCP throughput and energy efficiency than previous mechanisms.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3332-3339
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• 1999

TCP protocol is originally designed for the wired network in which the packet loss in mostly caused the network congestion, and thus the performance of protocol can be decreased in mobile Internet environment due to the comparatively high error rate of wireless link. This paper analyzes the influence of retransmission interference that is a major factor to the TCP performance decrease. We first build a model in order to simulate the mobile Internet environment that includes the wireless network, and then we implement a program using PARSEC simulation language and finally executed the program. Based on the simulation results, we suggest a method that reduce the effect of the retransmission interference.

• Journal of Digital Contents Society
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• v.15 no.2
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• pp.147-154
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• 2014

There have been company researches for TCP-Westwood algorithms in wireless TCP environment with high packet loss rate. Because the TCP-Westwood algorithm adjusts the congestion window according to the ABE(Available Bandwidth Estimation), the algorithm has a problem which the accuracy of ABE decreases as the error rate increases. To solve such a problem, the proposed scheme in this paper adopts the existing packet pattern based algorithm that the ABE is ignored when the arriving interval time of ACK is longer than a given interval time and uses new algorithm that the ABE is reallocated to a given allowable ABE when the ABE is over the allowable range. The proposed scheme shows the simulation result that the ABE is closest to the setting bandwidth for simulation compared to the existing algorithms.