• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.215-217
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• 2004

In wireless sensor networks, fair allocation of bandwidth among different nodes is one of the critical problems that effects the serviceability of the entire system. Fair bandwidth allocation mechanisms, like fair queuing, usually need to maintain state, manage buffers, and perform packet scheduling on a per flow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. It is a very important and difficult technical issue to provide packet scheduling architecture for fairness in wireless sensor networks. In this paper, we propose an packet scheduling architecture for sensor node, called FISN (Fairness Improvement Sensor Network), that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. Sensor node for sensing estimate the incoming rate of each sensor device and insert a label into each transmission packet header based on this estimate. Sensor node for forwarding maintain no per flow state; they use FIFO packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the gathering node. We present the detailed design, implementation, and evaluation of FISN using simulation. We discuss the fairness improvement and practical engineering challenges of implementing FISN in an experimental sensor network test bed based on ns-2.

• The KIPS Transactions:PartA
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• v.13A no.7 s.104
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• pp.573-580
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• 2006

The conventional transport layer protocol TCP is designed to work under condition of packet loss is due to the network congestion, so that it's suitable in the traditional wired network with fixed hosts but it's inefficient on the wireless network where the environment of fading, noise, and transmission error comes from interference. This result from the needless transmission control of the bit error is due to treats the packet loss as a packet congestion control in the wireless network. In this paper, we propose the advanced SNOOP protocol with the consecutive packet loss and TCP window control to avoid the needless congestion management algorithm in wireless network for the wireless TCP packet transmission enhancement. We verify the performance of the advanced module from the simulation experiment result.

• International Journal of Computer Science & Network Security
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• v.22 no.4
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• pp.394-400
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• 2022

The radio resources available in a wireless network system are limited. Therefor, job of managing resources is not easy task. Because the resources are shared among the UEs that are connected, the process of assigning resources must be carefully controlled. The packet scheduler in an LTE network is in charge of allocating resources to the user equipment (UE). Femtocells networks are being considered as a promising solution for poor channel performance for mulitple environments. The implementation of femtocells into a macrocell (traditional base station) would boost the capacities of the cellular network. To increase femtocells network capacity, a reliable Packet Scheduler mechanism should be implemented. The Packet Scheduler technique is introduced in this paper to maximize capacity of the network while maintaining fairness among UEs. The proposed solution operates in a manner consistent with this principle. An analysis of the proposed scheme's performance is conducted using a computer simulation. The results reveal that it outperforms the well-known PF scheduler in terms of cell throughput and average throughput of UEs.

• 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.

• Journal of Broadcast Engineering
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• v.12 no.2
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• pp.80-92
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• 2007

IPTV Services that is representative union service of broadcasting and telecommunication need guarantee of QoS, efficiency of multicasting, and hish bandwidth on the network. Because typical TDM based metro transport network was designed by transporting fixed voice traffic with stable and recovering method, it has a defect of bottleneck and a waste of bandwidth for acceptance of data traffic with burst feature and then all of data are treated equally at the transport network because it cannot classify between advanced high end service and best effort low end service. for completely resolving this kind of problem about increasing burst traffic and QoS issues, firstly we need to new design for transport network. This paper presents transformation method from TDM based metro transport network to packet based transport network and advantage and effectiveness of packet based transport network and also indicates technical factor and characters about method of packet transport system. As a result of research, the Packet Transport System, which is a transmission network for packet delivery, take in not only a specific character of legacy TDM but QoS, Multicast and high bandwidth, then, it is able to keep an effective bandwidth and a stabilized performance of packet transmissions. Additionally, if a fault be occurred on an optical link, the system is able to guarantee a differential QoS by an each service class using an algorithm to make certain of a traffic existence and contain a protective mechanism.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.12
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• pp.5948-5971
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• 2019

Estimating loss rates with the packet traces captured from some point in the middle of the network has received much attention within the research community. Meanwhile, existing intermediate-point methods like [1] require the capturing system to capture all the TCP traffic that crosses the border of an access network (typically Gigabit network) destined to or coming from the Internet. However, limited to the performance of current hardware and software, capturing network traffic in a Gigabit environment is still a challenging task. The uncaptured packets will affect the total number of captured packets and the estimated number of packet losses, which eventually affects the accuracy of the estimated loss rate. Therefore, to obtain more accurate loss rate, a method of strengthening packet loss measurement from the network intermediate point is proposed in this paper. Through constructing a series of heuristic rules and leveraging the binomial distribution principle, the proposed method realizes the compensation for the estimated loss rate. Also, experiment results show that although there is no increase in the proportion of accurate estimates, the compensation makes the majority of estimates closer to the accurate ones.

• Journal of Korea Multimedia Society
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• v.14 no.12
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• pp.1572-1582
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• 2011

The rapid growth of UnderWater Acoustic Sensor Networks (UW-ASNs) has led researchers to enhance underwater MAC protocols against limitations existing in underwater environment. We propose the customized robust real-time packet inspection mechanism with addressing the problem of the search for the data packet loss and network performance quality analysis in UW-ASNs, and describe our experiences using this approach. The goal of this work is to provide a framework to assess the network real-time performance quality. We propose a customized and adaptive mechanism to detect, monitor and analyze the data packets according to the MAC protocol standards in UW-ASNs. The packet analyzing method and software we propose is easy to implement, maintain, update and enhance. We take input stream as real data packets from sniffer node in capture mode and perform fully analysis. We were interested in developing software and hardware designed tool with the same capabilities which almost all terrestrial network packet sniffers have. Experimental results confirm that the best way to achieve maximum performance requires the most adaptive algorithm. In this paper, we present and offer the proposed packet analyzer, which can be effectively used for implementing underwater MAC protocols.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.4
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• pp.2382-2392
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• 2014

In this paper, a new data frame structure for wavelength routing in optical ring access network(ORAN) is designed and the functions of each field in the frame are defined. In addition, the IP packet transmission to the network layer corresponding to the newly proposed structure of the protocol stack is newly presented. Tha data transmission protocol design techniques of ORAN was proposed. IP packet and broadcasting packet transmission within the sub-network, and the IP packet transmission to other sub-network was shown through the process illustration. In the process, the encapsulation and framing process of the wavelength information to routing has been described in detail. And each step takes place in a packet transfer process is demonstrated. A ring type optical access network protocol is not yet research field. ORAN data transfer protocol to send/receive module structure of a subscriber node and a control node of the two rings which analyzed the results are shown. The high-speed Internet solution is proposed that by using wavelength routing the packet transport protocol for ORAN is designed.

• Journal of Korea Society of Digital Industry and Information Management
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• v.8 no.2
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• pp.81-94
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• 2012

This paper suggests PARED algorithm, a modified RED algorithm, that actively reacts to dynamic changes in network to apply packet drop probability flexibly. The main idea of PARED algorithm is that it compares the target queue length to the average queue length which is the criterion of changes in packet drop probability and feeds the gap into packet drop probability. That is, when the difference between the average queue length and the target queue length is great, it reflects as much as the difference in packet drop probability, and reflects little when the difference is little. By doing so, packet drop probability could be actively controled and effectively dealt with in the network traffic situation. To evaluate the performance of the suggested algorithm, we conducted simulations by changing network traffic into a dynamic stat. At the experiments, the suggested algorithm was compared to the existing RED one and then to ARED one that provided the basic idea for this algorithm. The results proved that the suggested PARED algorithm is superior to the existing algorithms.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.289-296
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• 2015

Wireless multimedia sensor network senses and transfers mass multimedia data. Also, it is sensitive to latency. This thesis proposes a routing technique based on traffic priority in order to improve the network efficiency by minimizing latency. In addition, it proposes a congestion control mechanism that uses packet service time, packet inter-arrival time, buffer usage, etc. In this thesis, we verified the reduction of packet latency in accordance with the quality level of packet as a result of the performance analysis through the simulation method. Also, we verified that the proposed mechanism maintained a reliable network state by preventing packet loss due to network overload.