• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.94-97
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• 1999

In this paper, a new real-time neural network connection admission controller is proposed. The proposed controller measures traffic flows, cell loss rate and cell delay periodically each classes. The Neural network learns the relation between those measured information and service quality by real-time. Also the proposed controller uses the DWRR multiplexer with buffer dedicated to every traffic source in order to measure the delay that cells experience in buffer. Experimental result shows that the proposed method can control effectively heterogeneous traffic sources with diverse QoS requirement.

• Information and Communications Magazine
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• v.26 no.6
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• pp.23-29
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• 2009

꾸준한 성장세를 바탕으로 최근에는 타 매체가 쉽게 접근하기 어려운 HD다채널 방송이라는 전략적 이점으로 제2도약의 동력을 마련한 국내위성방송은 그러나, 미디어의 큰 흐름 중 Interactivity즉 양방향성에 대해서는 기본적인 제한성을 갖고 있다. 또한 초기진입 단계인 IPTV의 경우, 전국적 동일 서비스를 위한 망의 QoS(Quality of Service)나 콘텐츠 수급 비용 측면에서 기존 매체가 겪어온 과정을 겪고 있다. 본 고에서는 이러한 양 매체의 기본적 속성을 바탕으로 상호 장단점을 보완할 수 있는 전략적 방안의 하나로 하이브리드형 서비스에 대해 사례 중심으로 알아보고자 한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.338-348
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• 2005

Currently, IEEE 802.11 Wireless LAN (WLAN) is rising as the most popular means for the broadband wireless access network. In this thesis, we propose a QoS(Quality of Service)-Oriented mechanism using handshaking method of scanning phase in IEEE 802.11 handoff. In conventional process for handoff, the major criterion to select the best AP(Access Point) among candidates is normally based on the RSS(Received Signal Strength), which does not always make the selected network guarantee the maximum achievable performance. Even though the link quality with a neighboring AP is excellent, the AP may not be a good candidate to handoff to simply because MAC(Medium Access Control) protocol of IEEE 802.11 standard is contention-based such as CSMA/CA. Therefore, if we apply network level information to AP selection criteria, we can achieve better handoff efficiency rather than before. The analysis and simulation results applied to our new mechanism show clearly better performance than AP selection based on traditional handoff method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.3
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• pp.229-235
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• 2010

The Orthogonal Frequency Division Multiplexing(OFDM) has been strongly recommended as a transmission technique in order to satisfy requests of high speed and high quality multimedia information. This paper considers resource allocation algorithm which supports the user Quality of Service(QoS) for multi-user OFDM system with simple CSI (Channel State Information) structure. After users eligible for services and the number of subcarrier are determined by minimum request data rate and average channel gain, subcarriers are allocated to increase total transmission rates. Although ideal CSI is not reported to base station, compared to conventional algorithm, we have obtained better result with simple CSI structure.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8A
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• pp.1195-1207
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• 1999

Wireless channel has high error rate and limited bandwidth. In order to guarantee the quality of service for multimedia through wireless channel, wireless ATM network requires the cell structure with error control schemes and an efficient medium access control protocol. In this paprt, an efficient medium access control protocol, which is considered of cell structure and the QOS, is proposed for wireless ATM networks. Many kinds of ATM cell structures are analyzed in terms of cell error rate and cell loss rate to choose the proper cell structures for this proposed MAC protocol. And a bandwidth allocation algorithm is also designed to satisfy the priority of services. The performance of proposed MAC protocol are evaluated in terms of channel utilization, call blocking and packet discard probability accotding to different kinds of cell structure through simulation.

• Journal of the Korea Society for Simulation
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• v.22 no.2
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• pp.53-62
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• 2013

Cloud environment is one in the field of distributed computing and it consists of physical nodes and virtual nodes. In distributed cloud environment, an optimal path search is that each node to perform a search for an optimal path. Synchronization of each node is required for the optimal path search via fast data transmission because of real-time environment. Therefore, a quantization technique is required in order to guarantee QoS(Quality of Service) and search an optimal path. The quantization technique speeds search data transmission of each node. So a main server can transfer data of real-time environment to each node quickly and the nodes can perform to search optimal paths smoothly. In this paper, we propose the quantization technique to solve the search problem. The quantization technique can reduce the total data transmission. In order to experiment the optimal path search system which applied the quantized data transmission, we construct a simulation of cloud environment. Quantization applied cloud environment reduces the amount of data that transferred, and then QoS of an application for the optimal path search problem is guaranteed.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.5
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• pp.605-614
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• 2005

In this paper, we present the design and implementation of an agent knowledge and QoS tuning methodology to improve the flexibility of agent-based flexible video-conference system. In order to improve the flexibility during video-conferencing, we propose a new T-INTER(Tuning-INTER) architecture of knowledge part in video-conference manager (VCM) agent in which an automatic QoS parameter tuning method is imbedded. The flexible video-conference system structured based on the proposed architecture can cope with the changes in service quality required by users. The VCM agent cooperates with other agents by protocols and executes the automatic QoS parameter tuning task whenever needed. By the tuned parameters, the system is able to flexibly cope with the internal or external changes and the burden of users can be decreased. In the experimental section, it is shown that our proposed system outperforms the existing system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8A
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• pp.656-663
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• 2005

This paper presents a multi-channel enhancement scheme for the MAC protocol of IEEE 802.15.3 High-rate WPAN (Wireless Personal Area Network). The current MAC protocol of the IEEE 802.15.3 High-rate WPAN is designed for sharing a single channel among DEVs of a piconet; that is, within a single piconet, PNC prevents interference through MAC layer assignment of time slots to other DEVs using time-division multiple access. When the number of DEVs that communicate with each other frequently, is increased in a single WPAN, the size of the superframe becomes inevitably large, and this may result in a significant throughput drop or a failure to provide QoS guarantee. A multi-channel enhancement scheme for the MAC protocol of IEEE 802.15.3 High-rate WPAN is proposed to significantly increase the aguegate throughput and more reliably provide the QoS guarantees in a piconet

• The KIPS Transactions:PartA
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• v.8A no.4
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• pp.345-352
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• 2001

In this paper, a VoIP (Voice on Internet Protocol) terminal system, which can process voice in real time based on Linux, is designed and implemented. The hardware of it is designed using a i486 processor and a DSP codec chip which encodes and decodes voice data in real time. As an operating system, RTLinux, which is a real-time operating system based on Linux, is ported to manage real-time voice processing. The voice processing module of the system uses G.723.1 voice codec of ITU-T standard. It transfers voice data within 30ms to assure good voice quality. In order to satisfy the real time requirements and QoS (Quality-of-Service) for the voice data, the real-time voice processing device driver is designed and implemented. To verify the system, the chatting application program is developed and tested for QoS of the system.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.136-142
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• 2006

Existing buffer management schemes for shared-memory output queueing switches can be classified into two types: In the first type, some constant amount of memory space is guaranteed to each virtual queue using static queue thresholds. The static threshold method (ST) belongs to this type. On the other hand, the second type of approach tries to maximize the buffer utilization in 머 locating buffer memories. The complete sharing (CS) method is classified into this type. In the case of CS, it is very hard to protect regular traffic from mis-behaving traffic flows while in the case of ST the thresholds can not be adjusted according to varying traffic conditions. In this paper, we propose a new buffer management method called weighted dynamic thresholds (WDT) which can process packet flows based on loss priorities for quality-of-service (QoS) functionalities with fairly high memory utilization factors. We verified the performance of the proposed scheme through computer simulations.