• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.505-509
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• 2006

Recently, wireless Internet service has become generalized, and required the user authentication of a mobile hosts and QoS. However, time required for user authentication during handoff Processes between a wireless LAN system and mobile hosts is long, and is unsuitable for real-time communication and multimedia applications. In this paper. we improved the existing system in order to reduce a lead time of user authentication as they extended a Fast Inter-AP handoff. We use a Mechanism to assume a confidence degree with bases by the time that stayed at individual wireless LAN systems in order to reduce the handoffs. The experiment network shows that it decreases communication load, and improves communication QoS.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.571-573
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• 2014

In today's technology streaming service's QoS technologiey is an issue. On quality video streaming service there are some technical issues exist, such as buffering. This submission is "Adaptive dual-streaming system design" which is for the integrity of the data streaming that is sent to TCP and UDP for faster transmission of data to the stream. This system provides real-time incoming video encoding in bitrate of h.264-based H through a process based on the video footage of several server and client-to-TCP and UDP via Adaptive providing streaming services in a network environment. This is an unspecified number of buffers in a network environment and continued through the minimization of various streaming for playback of videos and multimedia will be utilized in the field.

• Journal of KIISE:Information Networking
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• v.29 no.1
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• pp.1-14
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• 2002

The most important issue in providing multimedia traffic on a mobile computing environments is to guarantee the mobile host(client) with consistent QoS(Quality of Service). However, the QoS negotiated between the client and network in one cell may not be honored due to client mobility, causing hand-offs between cells. In this paper, a call admission control mechanism is proposed to provide consistent QoS guarantees for multimedia traffics in a mobile computing environment. Each cell can reserve fractional bandwidths for hand-off calls to its adjacent cells. It is important to determine the right amount of reserved bandwidth for hand-off calls because the blocking probability of new calls may increase if the amount of reserved bandwidth is more than necessary. An adaptive bandwidth reservation based on an MPP(Mobility Pattern Profile) and a 2-tier cell structure has been proposed to determine the amount of bandwidth to be reserved in the cell and to control dynamically its amount based on its network condition. We also propose a call admission control based on this bandwidth reservation and "next-cell prediction" scheme using an MPP. In order to evaluate the performance of our call admission control mechanism, we measure the metrics such as the blocking probability of our call admission control mechanism, we measure the metrics such as the blocking probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The simulation results show that the performance of our mechanism is superior to that of the existing mechanisms such as NR-CAT1, FR-CAT1, and AR-CAT1.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.126-129
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• 1998

In order to control the flow of traffics in ATM networks and optimize the usage of network resources, an efficient control mechanism is necessary to cope with congestion and prevent the degradation of network performance caused by congestion. This paper proposes a new UPC(Usage Parameter Control) mechanism that varies the token generation rate and the buffer threshold of leaky bucket by using a Neural Network controller observing input buffers and token pools, thus achieving the improvement of performance. Simulation results show that the proposed adaptive algorithm uses of network resources efficiently and satisfies QoS for the various kinds of traffics.

• Journal of Communications and Networks
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• v.10 no.1
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• pp.38-43
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• 2008

In this paper, adaptive channel borrowing approach fuzzy neural networks for load balancing (ACB-FNN) is presented to maximized the number of served calls and the depending on asymmetries traffic load problem. In a wireless network, the call's arrival rate, the call duration and the communication overhead between the base station and the mobile switch center are vague and uncertain. A new load balancing algorithm with cell involved negotiation is also presented in this paper. The ACB-FNN exhibits better learning abilities, optimization abilities, robustness, and fault-tolerant capability thus yielding better performance compared with other algorithms. It aims to efficiently satisfy their diverse quality-of-service (QoS) requirements. The results show that our algorithm has lower blocking rate, lower dropping rate, less update overhead, and shorter channel acquisition delay than previous methods.

• Journal of Information Processing Systems
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• v.10 no.2
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• pp.176-192
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• 2014

The realization of Wireless Multimedia Sensor Networks (WMSNs) has been fostered by the availability of low cost and low power CMOS devices. However, the transmission of bulk video data requires adequate bandwidth, which cannot be promised by single path communication on an intrinsically low resourced sensor network. Moreover, the distortion or artifacts in the video data and the adherence to delay threshold adds to the challenge. In this paper, we propose a two stage Quality of Service (QoS) guaranteeing scheme called Prioritized Multipath WMSN (PMW) for transmitting H.264 encoded video. Multipath selection based on QoS metrics is done in the first stage, while the second stage further prioritizes the paths for sending H.264 encoded video frames on the best available path. PMW uses two composite metrics that are comprised of hop-count, path energy, BER, and end-to-end delay. A color-coded assisted network maintenance and failure recovery scheme has also been proposed using (a) smart greedy mode, (b) walking back mode, and (c) path switchover. Moreover, feedback controlled adaptive video encoding can smartly tune the encoding parameters based on the perceived video quality. Computer simulation using OPNET validates that the proposed scheme significantly outperforms the conventional approaches on human eye perception and delay.

• Journal of Communications and Networks
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• v.17 no.4
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• pp.428-439
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• 2015

In this study, we present an approach for reducing the information inaccuracy of existing star aggregation based on bypass links when there are multi-constraint QoS parameters in asymmetric networks. In our approach, bypass links with low similarity are selected. Links that are not chosen as bypass links are included in each group depending on the star's link characteristics. Moreover, each link group is aggregated differently according to the similarity of the links that make up the group. The selection of a bypass link by using link similarity reduces the existing time complexity of O($N^3$) to O(N) by virtue of the simplification of the selection process. In addition, the adaptive integration according to the characteristics of the links in each group is designed to reduce the information inaccuracy caused by static aggregation. Simulation results show that the proposed method maintains low information distortion; specifically, it is 3.8 times lower than that of the existing method, even when the number of nodes in a network increases.

• Journal of Communications and Networks
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• v.12 no.6
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• pp.568-576
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• 2010

The paper presents a control architecture aimed at implementing bandwidth optimization combined with call admission control (CAC) over a digital video broadcasting (DVB) return channel satellite terminal (RCST) under quality of service (QoS) constraints. The approach can be applied in all cases where traffic flows, coming from a terrestrial portion of the network, are merged together within a single DVB flow, which is then forwarded over the satellite channel. The paper introduces the architecture of data and control plane of the RCST at layer 2. The data plane is composed of a set of traffic buffers served with a given bandwidth. The control plane proposed in this paper includes a layer 2 resource manager (L2RM), which is structured into decision makers (DM), one for each traffic buffer of the data plane. Each DM contains a virtual queue, which exactly duplicates the corresponding traffic buffer and performs the actions to compute the minimum bandwidth need to assure the QoS constraints. After computing the minimum bandwidth through a given algorithm (in this view the paper reports some schemes taken in the literature which may be applied), each DM communicates this bandwidth value to the L2RM, which allocates bandwidth to traffic buffers at the data plane. Real bandwidth allocations are driven by the information provided by the DMs. Bandwidth control is linked to a CAC scheme, which uses current bandwidth allocations and peak bandwidth of the call entering the network to decide admission. The performance evaluation is dedicated to show the efficiency of the proposed combined bandwidth allocation and CAC.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.5
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• pp.37-43
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• 2009

Streaming video service over wireless and mobile communication networks has received significant interests from both academia and industry recently. Specifically, mobile WiMAX (IEEE 802.16e) is capable of providing high data rate and flexible Quality of Service (QoS) mechanisms, supporting mobile streaming very attractive. However, we need to note that streaming videos can be partially deteriorated in their macroblocks and/or slices owing to errors on OFDMA subcarriers, as we consider that compressed video sequence is generally sensitive to the error-prone channel status of the wireless and mobile network. In this paper, we introduce an OFDMA subcarrier-adaptive mobile streaming server based on cross-layer design. This streaming server system is substantially efficient to reduce the deterioration of streaming video transferred on the subcarriers of low power strength without any modifications of the existing schedulers, packet ordering/reassembly, and subcarrier allocation strategies in the base station.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.6 s.312
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• pp.60-70
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• 2006

Currently, Scalable Video Coding (SVC) is being standardized. By using scalability of SVC, QoS managed video streaming service is enabled in heterogeneous networks even with only one original bitstream. But current SVC is insufficient to dynamic video conversion for the scalability, thereby the adaptation of bitrate to meet a fluctuating network condition is limited. In this paper, we propose dynamic full-scalability conversion method for QoS adaptive video streaming in H.264/AVC SVC. To accomplish full scalability dynamic conversion, we propose corresponding bitstream extraction, encoding and decoding schemes. On the encoder, we newly insert the IDR NAL to solve the problems of spatial scalability conversion. On the extractor, we analyze the SVC bitstream to get the information which enable dynamic extraction. By using this information, real time extraction is achieved. Finally, we develop the decoder so that it can manage changing bitrate to support real time full-scalability. The experimental results showed that dynamic full-scalability conversion was verified and it was necessary for time varying network condition.