• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.117-120
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• 2000

As the use of real-time multimedia applications increases, a considerable amount of “greedy” UDP traffic would easily dominate network bandwidth and packet loss. As a result, bandwidth available to TCP connections is oppressed and their performance extremely deteriorates. In or-der that both TCP and UDP sessions fairly co-exist in the Internet, it is vital that we consider the fairness among both protocols. In this work, we implement a “TCP-friendly” rate control mechanism suitable to video applications and con-sider its applicability to a real system through observation of the video quality at the receiver and the connection state. It is shown that we can achieve high-quality and stable video transfer fairly sharing the network bandwidth with TCP by applying our rate control at a control interval of 32 times as long as RTT.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.3809-3822
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• 2017

Wireless Sensor Networks hold the limelight because of significant potential for distributed sensing of large geographical areas. The radio duty cycling mechanism that turns off the radio periodically is necessary for the energy conservation, but it deteriorates the network congestion when the traffic load is high, which increases the packet loss and the delay too. Although many papers for WSNs have tried to mitigate network congestion, none of them has mentioned the congestion problem caused by the radio duty cycling of MAC protocols. In this paper, we present a simple and efficient congestion control technique that operates on the radio duty cycling MAC protocol. It detects the congestion by checking the current queue size. If it detects the congestion, it extends the network capacity by adding supplementary wakeup times. Simulation results show that our proposed scheme highly reduces the packet loss and the delay.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.6 s.348
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• pp.30-38
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• 2006

In this paper, an algerian that provisions absolute and proportional differentiation of packet delays is proposed with an objective for enhancing quality of service (QoS) in future packet networks. It features a scheme that compensates the deviation for prediction on the traffic to be arrived continuously It predicts the traffic to be arrived at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot and derives the difference between them. The deviation is utilized to the delay control operation for the next time slot to offset it. As it compensates the prediction error continuously, it shows superior adaptability to the bursty traffic as well as the exponential traffic. It is demonstrated through simulation that the algorithm meets the quantitative delay bounds and shows superiority to the traffic fluctuation in comparison with the conventional non-adaptive mechanism.

• Journal of the Korea Society for Simulation
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• v.15 no.4
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• pp.79-86
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• 2006

Data transmission in poor environment like underwater has considerably serious delay rate and ewer rate. Like this environment usually has heavy fluctuation of error rate and limited wireless communication state. Therefore, mechanism using in such environment has to be efficient and simple. This paper suggests a new block ack mechanism, called the Pervasive Block ACK (PBA), which transmits aggregated ACKs. This mechanism takes effect on reducing number of traffic, decreasing overhead and delay rate in poor environment networks like underwater. Additionally, we can expect energy consumption. We verify propriety and efficiency of PBA through describing numerical result based analytical formula in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.29-40
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• 2004

In this paper, we propose a new early congestion detection and notification technique called QR-AQM. Unlike RED and it's variation, QR-AQM measures the total traffic rate from TCP sessions, predicts future network congestion, and determine the packet marking probability based on the measured traffic rate. By incorporating the traffic rate in the decision process of the packet marking probability, QR-AQM is capable of foreseeing future network congestion as well as terminating congestion resolution procedure in much more timely fashion than RED. As a result, simulation results show that QR-AQM maintains the buffer level within a fairly narrow range around a target buffer level that may be selected arbitrarily as a control parameter. Consequently, compared to RED and its variations, QR-AQM is expected to significantly reduce the jitter and delay variance of packets traveling through the buffer while achieving nearly identical link utilization.

• Journal of the Korean Data and Information Science Society
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• v.21 no.4
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• pp.783-793
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• 2010

Several multi-hop applications developed for vehicular ad hoc networks use broadcast as a means to either discover nearby neighbors or propagate useful traffic information to other vehicles located within a certain geographical area. However, the conventional broadcast mechanism may lead to the so-called broadcast storm problem, a scenario in which there is a high level of contention and collisions at the link layer due to an excessive number of broadcast packets. We present a fuzzy alert message dissemination algorithm to improve performance for road safety alert application in Vehicular Ad-hoc Network (VANET). In the proposed algorithm, when a vehicle receives an alert message for the first time, the vehicle rebroadcasts the alert message according to the fuzzy control rules for rebroadcast degree, where the rebroadcast degree depends on the current traffic density of the road and the distance between source vehicle and destination vehicle. Also, the proposed algorithm is the hybrid algorithm that uses broadcast protocol together with token protocol according to traffic density. The performance of the proposed algorithm is evaluated through simulation and compared with that of other alert message dissemination algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.11
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• pp.2828-2835
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• 1996

In this paper we propose a new shaping algorithm which can control the shaping delay and the output buffer size based on the leaky bucket counter with a threshold value. This paper assumes that input traffic of the proposed shaping algorithm is the worst case traffic tolerated by the continuous leaky bucket algorithm and claracterizes traffic patterns that can depart from our shaping algorithm. We also compare shaping delay and output buffer size of the proposed algorithm with the existing shaping algorithm without a threshold value. Our results show that the proposed shaping algorithm can easily manage the shaping delay and output buffer size than any other mechanism.

• Journal of Communications and Networks
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• v.6 no.1
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• pp.68-77
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• 2004

In the near future packet networks should support applications which can not predict their traffic requirements in advance, but still have tight quality of service requirements, e.g., guaranteed bandwidth, jitter, and packet loss. These dynamic characteristics mean that the sources can be made to modify their data transfer rates according to network conditions. Depending on the customer&; needs, network operator can differentiate incoming connections and handle those in the buffers and the interfaces in different ways. In this paper, dynamic QoS-aware scheduling algorithm is presented and investigated in the single node case. The purpose of the algorithm is in addition to fair resource sharing to different types of traffic classes with different priorities ?to maximize revenue of the service provider. It is derived from the linear type of revenue target function, and closed form globally optimal formula is presented. The method is computationally inexpensive, while still producing maximal revenue. Due to the simplicity of the algorithm, it can operate in the highly nonstationary environments. In addition, it is nonparametric and deterministic in the sense that it uses only the information about the number of users and their traffic classes, not about call density functions or duration distributions. Also, Call Admission Control (CAC) mechanism is used by hypothesis testing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1B
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• pp.8-19
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• 2010

In this paper, we propose the NCP (Network Control Platform)-based defense mechanism against DDoS (Distributed Denial of Service) attacks in order to guarantee the transmission of normal traffic and prevent the flood of abnormal traffic. We also define defense modules, the threshold and packet drop-rate used for the response against DDoS attacks. NCP analyzes whether DDoS attacks are occurred or not based on the flow and queue information collected from SR (Source Router) and VR (Victim Router). Attack packets are dopped according to drop rate decided from NCP. The performance is simulated using OPNET and evaluated in terms of the queue size of both SR and VR, the transmitted volumes of legitimate and attack packets at SR.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1647-1653
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• 2004

In a contention-based wireless LAN protocol, maximization of channel utilization and fair bandwidth allocations are main topics to deal with. But it is very difficult to achieve these two goals simultaneously. Many studies have been done to achieve these goals. In this paper we propose a control mechanism to support fair transmissions among traffic classes in IEEE 802.11e Wireless LAN. The proposed algorithm uses short-term and long-term transmission times of each traffic classes to control their $CW_{min}$ for fairness. The proposed algorithm don't need to know the exact number of nodes in the networks to support fairness as other studies do. Furthermore any modifications in AP and mobile hosts are not required.