• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.4
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• pp.720-740
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• 2011

Wi-Fi-enabled portable devices have recently been introduced into the consumer electronics market. These devices download or upload content, from or to a host machine, such as a personal computer, a laptop, a home gateway, or a media server. This paper investigates the fairness among multiple Wi-Fi-enabled portable devices in a home network when they are simultaneously communicated with the host machine. First, we present that, a simple IEEE 802.11-based home network suffers from unfairness, and the fairness is exaggerated by the wireless link errors. This unfairness is due to the asymmetric response of the TCP to data-packet loss and to acknowledgment-packet loss, and the wireless link errors that occur in the proximity of any node; the errors affect other wireless devices through the interaction at the interface queue of the home gateway. We propose a QoS-provisioning framework in order to achieve per-device fairness and service differentiation. For this purpose, we introduce the medium access price, which denotes an aggregate value of network-wide traffic load, per-device link usage, and per-device link error rate. We implemented the proposed framework in the ns-2 simulator, and carried out a simulation study to evaluate its performance with respect to fairness, service differentiation, loss and delay. The simulation results indicate that the proposed method enforces the per-device fairness, regardless of the number of devices present and regardless of the level of wireless link errors; furthermore it achieves high link utilization with only a small amount of frame losses.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1609-1617
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• 2013

TFRC(TCP-Friendly Rate Control) has a performance degradation in wireless networks because it performs congestion control by judging all the losses occurred in wireless networks as a congestion indicator. It is also degraded by the increased Round Trip Time(RTT) due to packet retransmission and contention overhead in the link layer. In this paper, we propose an adaptive rate control scheme based on cross-layer to improve the quality of streaming services in the wireless networks. It provides new RTT estimation and loss discrimination methods to improve transmission rate of TFRC. The simulation results show that the proposed scheme can improve the performance of TFRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.1-8
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• 2006

Currently, there are many efforts underway to provide Internet service on integrated wireless and wired networks. Supporting IP mobility is one of the major issues to construct IP based wireless network. Mobile IP has been proposed to solve the IP Mobility problem. But, in processing frequent handoffs in cellular based wireless access network, Micro mobility protocols have been proposed to solve these problems. Micro mobility protocols proposed the Cellular IP, HAWII, and Hierarchical Mobile IP. Cellular IP attracts special attention for it's seamless mobility support in limited geographical areas. New BS must be known to occur begging of handoff in Cellular IP indirect handoff. Therefore during perceiving of hanoff, packet loss or packet duplication still can occur in Cellular IP indirect handoff, which results in the degradation of UDP and TCP performance. In this paper, we propose a enhanced indirect handoff scheme for Cellular IP. Proposed handoff scheme is using a crossover node to minimize the signalling procedure and using a buffering to minimize the packet loss or packet duplication.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.210-216
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• 2014

In this paper, it was proposed to develop front and rear image monitoring system for vehicle that help a driver to cope with urgent situation about a dangerous element. When parking a vehicle, the risk factors to be formed by the dead zone can be resolved by using anterior and posterior cameras of the vehicle. In embedded system environment, a SoC(System on Chip) and two high-resolution CMOS (Complementary metal-oxide-semiconductor) image sensors were used to transfer two high-resolution image data through he TCP/ IP-based network. To transfer image data through he TCP/ IP-based network, the images received by two cameras were compressed by using H.264 and they were transmitted with wireless method(Wi-Fi) by using real-time transport protocol (Real-time Transport Protocol). Transmission loss, transmission delay and transmission limit were solved in wireless (Wi-Fi) environment and the bit-rate of two image data compressed by H.264 was adjusted. And the system for the optimal transmission in wireless (Wi-Fi) environment was materialized and experimented.

• Journal of Korea Multimedia Society
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• v.13 no.5
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• pp.740-753
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• 2010

Performance of TCP can be severely degraded in WLP-based Mobile IP wireless networks where packet loss not related to network congestion occurs frequently during WLP-based inter-subnetwork handoff by user mobility. To resolve such a problem in the networks using WLP-based Mobile IP, the packet buffering method recovering seamlessly the packets dropped due to user mobility has been proposed. The packet buffering method at a bridge station recovers those packets dropped during handoff by forwarding buffered packets at the old bridge station to the WLP device. But, when the WLP device moves to a congested bridge station in a new WLP foreign subnetwork, those buffered packets forwarded by the old bridge station are dropped and TCP transmission performance of a WLP device in the congested bridge station degrades due to increased congestion by those forwarded burst packets. In this paper, a PBM(Packet Bridge Marker) is proposed for preventing buffered out-of-profile(OUT) packets from reducing the throughput of in-profile(IN) packets of an Assured Service WLP device. From this operation, the packet losses of buffered OUT packets are avoided and the throughput of IN and Total packets of an AS WLP device are increased.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.466-474
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• 2007

Asymmetric networks that the downlink bandwidth is larger than the uplink bandwidth may cause the degradation of the TCP performance due to the uplink congestion. In order to solve this problem, this paper designs and implements the Dynamic Segment Size Control mechanism which offers a suitable segment size for current networks. The proposed mechanism does not require any changes in customer premises but suppress the number of ACKs using segment reassembly technique to avoid the uplink congestion. The gateway which adapted the Dynamic Segment Size Control mechanism, detects the uplink congestion condition and dynamically measures the bandwidth asymmetric ratio and the packet loss ratio. The gateway reassembles some of segments received from the server into a large segment and transmits it to the client. This reduces the number of corresponding ACKs. In this mechanism, the SACK option is used when occurs the bit error during the transmission. Based on the simulation in the GEO satellite network environment, we analyzed the performance of the Dynamic Segment Size Control mechanism.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Journal of Communications and Networks
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• v.8 no.1
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• pp.93-105
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• 2006

The domination of the Internet by TCP-based services has spawned many efforts to provide high network utilization with low loss and delay in a simple and scalable manner. Active queue management (AQM) algorithms attempt to achieve these goals by regulating queues at bottleneck links to provide useful feedback to TCP sources. While many AQM algorithms have been proposed, most suffer from instability, require careful configuration of nonintuitive control parameters, or are not practical because of slow response to dynamic traffic changes. In this paper, we propose a new AQM algorithm, hybrid random early detection (HRED), that combines the more effective elements of recent algorithms with a random early detection (RED) core. HRED maps instantaneous queue length to a drop probability, automatically adjusting the slope and intercept of the mapping function to account for changes in traffic load and to keep queue length within the desired operating range. We demonstrate that straightforward selection of HRED parameters results in stable operation under steady load and rapid adaptation to changes in load. Simulation and implementation tests confirm this stability, and indicate that overall performances of HRED are substantially better than those of earlier AQM algorithms. Finally, HRED control parameters provide several intuitive approaches to trading between required memory, queue stability, and response time.

• The Journal of the Korea Contents Association
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• v.5 no.2
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• pp.115-121
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• 2005

In the present internet environment, various traffic flooding attacks and worm attacks cause economical loss. If IPv4 is substituted by IPv6 because of the lack of IP address, it will be more serious. Therefore, we design and implement the traffic analysis tool which can detect attacks by expecting them encountered in the IPv6 environment. Proposed tool is composed of packet generation module, packet gathering module, discrimination module, and display module in X-windows. As a simulation result, it is proved that it can effectively detect DAD-NA message attack, TCP SYN flooding attack, UDP flooding attack and ICMP flooding attack in the IPv6 environment.