• Journal of Communications and Networks
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• v.4 no.2
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• pp.108-117
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• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.382-390
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• 2005

As today's networks evolve towards an If-based integrated network, the role of transmission control protocol(TCP) has been increasing as well. As a well-known issue, the performance of TCP is affected by its loss recovery mechanism that is comprised of two algorithms; fast retransmit and fast recovery. Although retransmission timeout(RTO) caused by multiple packet losses can be avoided by using selective acknowledgement(SACK) option, RTO cannot be avoided if a retransmitted packet is lost. Therefore, we propose a simple modification to make it possible for a TCP sender using SACK option to detect a lost retransmission. In order to evaluate the proposed algorithm, simulations have been performed for two scenarios where packet losses are random and correlated. Simulation results show that the proposed algorithm can improve TCP performance significantly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.660-666
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• 2004

Overall TCP performance represented by end-to-end throughput is largely dependent upon its loss recovery performance. In particular non-congestion packet losses caused by transmission errors degrade TCP performance seriously. Using Markov process, we analyze TCP loss recovery performance for correlated packet losses caused by multipath fading. The results show that loss recovery performance can be severely affected by burstiness in packet losses, even if overall packet loss ratio is very low.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.10
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• pp.4977-4996
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• 2016

The existence of excessively large and too filled network buffers, known as bufferbloat, has recently gained attention as a major performance problem for delay-sensitive applications. Researchers have made three types of suggestions to solve the bufferbloat problem. One is End to End (E2E) congestion control, second is deployment of Active Queue Management (AQM) techniques and third is the combination of above two. However, these solutions either seem impractical or could not obtain good bandwidth utilization. In this paper, we propose a Transmission Control Protocol（TCP）delayed window update mechanism which uses a congestion detection approach to predict the congestion level of networks. When detecting the network congestion is coming, a delayed window update control strategy is adopted to maintain good protocol performance. If the network is non-congested, the mechanism stops work and congestion window is updated based on the original protocol. The simulation experiments are conducted on both high bandwidth and long delay scenario and low bandwidth and short delay scenario. Experiment results show that TCP delayed window update mechanism can effectively improve the performance of the original protocol, decreasing packet losses and queuing delay while guaranteeing transmission efficiency of the whole network. In addition, it can perform good fairness and TCP friendliness.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.45-50
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• 2008

Many applications of sensor network require connection to the Internet. The transmission protocol of traditional sensor network was designed within the sensor network itself. However, based on 6LoWPAN which can be accessed using IPv6, direct connection is possible between the sensor network and the TCP/IP network outside. Transmission of data in applications of sensor network falls into two main categories. One is a small packet that is periodically produced such as packet related to temperature and humidity. The other is a relatively large packet that brings about network overheads such as images. We investigated the conformance test and pros and cons of application data over the transmission protocol of Zigbee and 6LoWPAN. As a result, both Zigbee and 6LoWPAN have shown low rate of loss for periodic data and have in creased reliability of data transfer. When transmitting streaming image data, both ACK, non ACK mode of Zigbee and UDP of 6LoWPAN minimized transmission time but suffered the consequences of high packet loss. Even though TCP of 6LoWPAN required a long transmission time, we were able to confirm that no loss has occurred.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.281-288
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• 2008

Recently, the advance of multimedia hardware has fostered the development of wireless multimedia sensor network which is able to ubiquitously obtain multimedia content such as image or audio from the environment. The multimedia data which has several characteristics such as large size and correlation between the data requires reliability in transmission. However, the existing solution which take the focus on the efficiency of network mainly, is not appropriate to transmit the multimedia data. In the paper, we proposes a reliable asynchronous image transfer protocol, RAIT. RAIT applies double sliding window method in node-to-node image tansfer to prevent the packet loss caused by network congestion. The double sliding window consists of one sliding window for the receiving queue, which is used for prevention of packet loss caused by communication failure between nodes and the other sliding window for the sending queue which prevents the packet loss caused by network congestion. the routing node prevents the packet loss and guarantees the fairness between the nodes by scheduling the packets based on the image non-preemptively. The RAIT implements the double sliding window method by cross layer design between RAIT layer, routing layer, and queue layer. The experiment shows that RAIT guarantees the reliability of image transmission compared with the existing protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.7
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• pp.3176-3183
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• 2011

Traffic and congestion control in the wireless sensor network is an important parameter that decides the throughput and QoS (Quality of Service). This paper proposes a transmission rate priority-based traffic control scheme to serve digital contents streaming in wireless sensor networks. In this paper, priority for transmission rate decides on the real-time traffic and non-real-time with burst time and length. This transmission rate-based priority creates low latency and high reliability so that traffic can be efficiently controlled when needed. Traffic control in this paper performs the service differentiation via traffic detection process, traffic notification process and traffic adjustment. The simulation results show that the proposed scheme achieves improved performance in delay rate, packet loss rate and throughput compared with those of other existing CCF and WCA.

• Journal of IKEEE
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• v.13 no.3
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• pp.7-17
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• 2009

WiMedia UWB technology is a fully distributed data communication technology developed for the application demanding a high data transmission rate in the wireless PAN area. In general, devices can send data either by reserving time slots or by using prioritized CSMA/CA. If the PCA protocol of prioritized CSMA/CA is used, they are suffered congestion as the number of devices increases. In this paper, we propose a Deterministic Access Protocol(DAP) in WiMedia WPANs. A DAP is a method to transmit data in the non-reserved DRP period without competition as each device informs the beacon order information in the beacon period and the queue information. In addition, the problem that the devices with a lower beacon slot number have more transmission opportunities is addressed by introducing the reference point. Simulation results are given to demonstrate that a DAP can improve the throughput and reduce the packet loss rate.