• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.595-603
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• 2003

To provide mobile nodes with continuous communication services, it is important to reduce the packet losses during handoffs. The handoffs of mobile nodes cause packet losses and decrease of TCP throughput on account of a variety of factors. One of those is the congestion in the new cell. Due to the congestion, not only the node moving into the cell but also the already existing nodes that were successfully communicating in the cell suffer the performance degradation. In this paper we propose a new handoff mechanism called‘packet freeze control’, which avoids the congestion caused by handoffs by regulating the influx of traffic burst into the new cell. Packet freeze control is applicable to a wireless network domain in which FAs(Foreign Agents) are connected hierarchically and constitute a logical tree. It gradually increases the number of packets transferred to the new cell by buffering packets in the FAs on the packet delivery path over the wireless network domain. The simulation results show that the proposed mechanism not only reduces the packet losses but also enhances the TCP throughput of other mobile nodes in the cell.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.389-405
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• 2001

In these days, we have increasing demands on the real-time services, especially for the multimedia data transmission in both of wired and wireless environments and thus efficient and stable ways of transmitting realtime data are needs. Although RTP is widely used for internet-based realtime applications, it cannot avoid packet losses, due to the use of UDP stack and its underlying layers. In the case of mobile computing applications, the packet losses are more frequent and consecutive because of the limited bandwidth. In this paper, we first statistically analyze the characteristics of packet losses in the wired and wireless communications, based on Gilbert model, and a new packet recovery scheme for realtime data transmission is presented. To reflect the transmission characteristics of the present network environment, our scheme makes the sender to dynamically adjust the amount of redundant information, using the current packet loss characteristic parameters reported by the receiver. Additionally, we use relatively large and discontinuous offset values, which enables us to recover from both of the random and consecutive packet losses. Due to these characteristics, our scheme is suitable for the mobile computing environment where packet loss rates are relatively high and varies rapidly in a wide range. Since our scheme is based on the analytic model form statistics, it can also be used for other network environments. We have implemented the scheme with Mobile IP and RTP/RTCP protocols to experimentally verify its efficiency.

• Journal of KIISE:Information Networking
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• v.27 no.2
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• pp.160-174
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• 2000

Nowadays, most widely used transport protocol, TCP is tuned to perform well in traditional networks where packet losses occur mostly because of congestion. TCP performs reliable end-to-end packet transmission under the assumption of low packet error rate. However, networks with wireless links suffer from significant losses due to high error rate and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in inefficient use of network bandwidth and degraded end-to-end performance in that system. To solve this problem, several methods have been proposed. In this paper, we analyse and compare these methods and propose appropriate model for improving TCP performance in the network with wireless links. This model uses TCP selective acknowledgement (SACK) option between TCP ends, and also uses caching method at the base station. Our simulation results show that using TCP SACK option with base station caching significantly reduces unnecessary duplicate retransmissions and recover packet losses effectively.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.7
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• pp.3150-3171
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• 2018

In this paper, a heuristic method that leverages packet traces captured at the entire boarder of an ISP to distinguish and estimate the overall packet loss within an ISP's management domain (Intra_Path_Loss) and that in the outside Internet (Inter_Path_Loss) is proposed. Our method is inspired by that packet losses happened at different locations will cause different TCP sequence number patterns at the border of an ISP. Thereby, we leverage these TCP sequence number patterns to build a series of heuristic rules to estimate Intra_Path_Loss and Inter_Path_Loss, respectively. We do this work with an eye towards showing that the overall packet losses defined and estimated in this paper can provide the operators with some valuable information to help them precisely grasp the overall performance of network paths and narrow down the range of network anomalies. The proposed method is rigorously validated with simulations, and finally the results from a regional academic network JSERNET verify its effectiveness and practicability.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.9
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• pp.30-38
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• 2008

The traditional TCP was designed assuming wired networks. Thus, if it is used networks consisting of both wired and wireless networks, all packet losses including random losses in wireless links are regarded as network congestion losses. Misclassification of packet losses causes unnecessary reduction of transmission rate, and results in waste of bandwidth. In this paper, we present WTCP(wireless TCP) congestion control algorithm that differentiates the random losses more accurately, and adopts improved congestion control which results in better network throughput. To evaluate the performance of proposed scheme, we compared the proposed algorithm with TCP Westwood and TCP Veno via simulations.

• Journal of the Korea Society of Computer and Information
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• v.16 no.1
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• pp.133-142
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• 2011

When TCP operates in multi-hop wireless networks, it suffers from severe performance degradation due to the different characteristics of wireless networks and wired networks. This is because TCP reacts to wireless packet losses by unnecessarily decreasing its sending rate assuming the losses as congestion losses. Although several loss differentiation algorithms (LDAs) have been proposed to avoid such performance degradation, their detection accuracies are not high as much as we expect. In addition the schemes have a tendency to sacrifice the detection accuracy of congestion losses while they improve the detection accuracy of wireless losses. In this paper, we suggest a new sender-based loss differentiation scheme which enhances the detection accuracy of wireless losses while minimizing the sacrifice of the detection accuracy of congestion losses. Our scheme estimates the rate of queue usage which is highly correlated with the congestion in the network path between a TCP sender and a receiver, and it distinguishes congestion losses from wireless losses by comparing the estimated queue usage with a certain threshold. In the extensive experiments based on a network simulator, QualNet, we measure and compare each detection accuracy of wireless losses and congestion losses, and evaluate the performance enhancement in each scheme. The results show that our scheme has the highest accuracy among the LDAs and it improves the most highly TCP performance in multi-hop wireless networks.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.8
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• pp.1132-1137
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• 2013

This paper considers the $H_{\infty}$ control problem for networked control systems(NCSs). In order to solve the problem which comes from discontinuous control signal in NCSs, an approach that discontinuous control signals treat time-varying delayed continuous signals is applied to achieve $H_{\infty}$ stability of NCSs. In addition, randomly occurring packet losses and disturbances are considered by introducing stochastic variables with Bernoulli distribution. Based on Lyapunov stability theory, a new stability condition is obtained via linear matrix inequality formulation to find the $H_{\infty}$ controller which achieves the mean square stability of NCSs. Finally, the proposed method is applied to a numerical example in order to show the effectiveness of our results.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.11
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• pp.810-819
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• 2006

This paper proposes an Adaptive Double Buffer Model. As a new FIFO buffer model, this technique minimizes packet losses from network congestion by logically managing buffers. It allocates the spare spaces of non-congested buffers to congested buffers by allowing receive/send buffers to share two queues, and hence it minimizes packet losses. In contrast to the buffer model utilizing a free list, this buffer model can prevent the bubble phenomenon caused by a memory leak and thereby apply to a network buffer in a restricted environment. Also, compared with the model using an way, this model brings maximum 100 percent improvement in accepting packets and compared with the model utilizing a free list, this model has the similar efficiency Results of the performance test on Adaptive Double Buffer Model, shows that this proposed model decreases packet losses and enhances memory efficiency.

• 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.