• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.260-269
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• 2003

The traditional distributed object middleware platform is not suitable for the transmission of stream data, because RPC(Remote Procedure Call)-based message transmission have a great overhead. Therefore, the OMG(Object Management Group) proposes the AV(Audio and Video) stream reference model for streaming on the distributed object middleware platform. But, this reference model has not a detail of implementation. Particularly it also has not congestion control scheme for improvement of network efficiency on the real network environment. It is a very important and difficult technical issue to provide the stream transmission platform with advanced congestion control scheme. In this paper, we propose an architecture of a distributed stream transport platform and deal with the design and implementation concept of our proposed platform. Also, we present a mechanism to improve streaming utilization by SRTP(Smart RTP). SRTP is our proposed TCP-Friendly scheme.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.

• Journal of KIISE:Information Networking
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• v.36 no.2
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• pp.143-152
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• 2009

IEEE 802.11 WLANs provide multiple transmission rates to improve the system throughput by adapting the transmission rate to the current wireless channel conditions. Many rate adaptation schemes have been proposed because IEEE 802.11 standard does not contain any specifications for the rate adaptation scheme. In this paper, in order to overcome limitations of the previous research, we propose a new rate adaptation scheme called SARA(Sender-oriented Automatic Rate Adaptation). The SARA scheme, a proposed rate adaptation scheme, appropriately adjusts the data transmission rate based on the estimated wireless channel conditions, specifically the measured RSSI at the sender-side. Moreover it continuously updates the thresholds for selecting the transmission rate and selectively enforces the RTS/CTS exchanges to adapt the changes in the wireless channel conditions. Through the performance evaluations, we prove that the SARA scheme overcomes the limitations of the previous research and improves the wireless link utilization.

• KIISE Transactions on Computing Practices
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• v.21 no.12
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• pp.780-785
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• 2015

Recently HTTP (Hypertext Transfer Protocol) adaptive streaming services have been the subject of much attention. The video quality control scheme of conventional HTTP adaptive streaming services estimates bandwidth using segment throughput and smooths out the sample of segment throughput. However, the conventional scheme has the problem of QoE (Quality of experience) degradation occurring with buffer underflow and frequent quality change due to the fixed number of samples. In order to solve this problem, we propose a video quality control scheme based on segment throughput and buffer occupancy. The proposed scheme determines the number of samples according to the variation of segment throughput. The proposed scheme also controls video quality based on the threshold of bitrate to keep stable buffer occupancy. The simulation results show that proposed scheme improves QoE by preventing buffer underflow and decreasing quality change when compared with the conventional scheme.

• Journal of KIISE:Information Networking
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• v.34 no.3
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• pp.143-155
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• 2007

In video streaming applications over the Internet, TCP-friendly rate control schemes are useful for improving network stability and inter-protocol fairness. However it does not always guarantee a smooth quality for video streaming. To simultaneously satisfy both the network and application requirements, video streaming applications should be quality-adaptive. In this paper, we propose a new quality adaptation mechanism to adjust the quality of congestion controlled video stream by controlling the frame rate. Based on the current network condition, it controls the frame rate and sending rate of video stream. Through the simulation, we prove that our adaptation mechanism appropriately adjusts the quality of video stream while improving network stability.

• Journal of KIISE:Information Networking
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• v.34 no.2
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• pp.100-109
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• 2007

Streaming protocol with the Available Bandwidth measurement scheme has problems that are to measure a Available Bandwidth uncorrectly and slowly. In this basis, in order to overcome limitations of the previous streaming protocols, we propose the NABO that is a New Available Bandwidth measurement scheme used by OWD(One-Way Delay). Proposed NABO(New Available Bandwidth measurement based on OWD) measures the constant transmission delay occurred by bottleneck link capacity and the variable delay. Competing traffic contribute to the variable delay. Through the measurement of the constant transmission delay and the competing traffic, a NABO can measure the Available Bandwidth correctly and fast in network. The simulation result proves that the proposed NABO has a performance that satisfies both accuracy viewpoint and measurement speed viewpoint.

• Journal of KIISE:Information Networking
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• v.37 no.1
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• pp.42-49
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• 2010

Because sensor nodes operate with the limited power based on battery which cannot be easily replaced, energy efficiency is a fundamental issue pervading the design of communication protocols developed for wireless sensor networks. In wireless networks, energy efficient MAC protocols can usually be described as being either a contention-based protocol or a schedule-based protocol. It is suitable to use combination of both contention-based protocol and schedule-based protocol, because the strengths and weaknesses of these protocols are contrary to each other. In this paper, in order to minimize energy consumption of sensor nodes and maximize network lifetime, we propose a new MAC protocol called "Tiered-MAC" The Tiered-MAC uses a schedule-based TDMA inside maximum transmission range of sink node and a contention-based CSMA otherwise. Therefore, by efficiently managing the congested traffic area, the Tiered-MAC reduces the unnecessary energy consumption. Based on the ns-2 simulation result, we prove that the Tiered-MAC improves the energy-efficiency of sensor network nodes.

• Journal of KIISE:Databases
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• v.35 no.2
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• pp.125-131
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• 2008

It is very useful to predict orthologs for new genome annotation and research on genome evolution. We showed that the previous work can be extended to construct OCs(Ortholog Clusters) automatically from multiple complete-genomes. The proposed method also has the quality of production of InParanoid, which produces orthologs from just two genomes. On the other hand, in order to predict more exactly the function of a newly sequenced gene it can be an important issue to prevent unwanted inclusion of paralogs into the OCs. We have, here, investigated how well it is possible to construct a functionally purer OCs with score cut-offs. Our OCs were generated from the datasets of 20 procaryotes. The similarity with both COG(Clusters of Orthologous Group) and KO(Kegg Orthology) against our OCs has about 90% and inclines to increase with the growth of score cut-offs.

• Journal of KIISE:Information Networking
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• v.32 no.5
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• pp.583-592
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• 2005

An important factor influencing the robustness of the Internet is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes the network congestion worse. Brakmo and Peterson proposed a congestion control algorithm, TCP Vegas, by modifying the congestion avoidance scheme of TCP Reno. Many studies indicate that Vegas is able to achieve better throughput and higher stability than Reno. But there are three unfairness problems in Vegas. These problems hinder the spread of Vegas in the current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP NewVegas. The proposed NewVegas is able to solve these unfairness problems effectively by using the variation of the number of queued packets in a bottleneck router. To evaluate the proposed approach, we compare the performance among NewVegas, Reno and Vegas. Through the simulation, NewVegas is shown to be able to achieve throughput and better fairness than Vegas.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.524-532
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• 2002

With the recent rapid progress of Internet, the higher speed network is needed to support the exploration of ambient information from text-based to multimedia-based information. Also, demands for additional Layer 3 routing technique, such as Network Address Translator (NAT) and Firewall, are required to solve a limitation of a current Internet address space and to protect the interior network from the exterior network. However, current router-based algorithms do not provide mechanisms to solve the congestion and fairness problems, while supporting the multimedia services and satisfying the user requirements. In this paper, to solve these problems, a new active queue management, called MFRED (Multiple Fairness RED) algorithm, is proposed. This algorithm can efficiently reduce the congestion in a router or gateway based on the Layer 3 routing technique, such as NAT. This algorithm can improve the fairness among TCP-like flows and unresponsive flows. It also works well in fairly protecting congestion-sensitive flows, i.e. fragile TCP, from congestion-insensitive or congestion-causing flows, i.e. robust TCP.