• Title/Summary/Keyword: 시간 지연을 이용한 제어

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A Playing Module under UPnP framework for Various Audio/Video Service (다양한 오디오/비디오 서비스를 위한 UPnP 프레임워크의 재생 모듈 연구)

  • Yun H.;Lee H.;Kim S.;Lee K.;Kim S.;Mah P.;Cho C.;Bae S.
    • Proceedings of the Korean Information Science Society Conference
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    • 2005.11a
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    • pp.247-249
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    • 2005
  • 최근 디지털 TV, DVD 플레이어, 홈시어터와 같은 디지털 A/V 기기들을 통합하여 멀티미디어 컨텐츠를 서비스 받는 홈 네트워크 기술에 대한 관심이 높아지고 있다. 이러한 환경에서 UpnP를 이용한 디지털 기기들의 상태를 모니터링하고, 사용자의 요청과 제어 명령을 전달하여, 멀티미디어 컨텐츠를 서비스 받을 수 있도록 하는 연구는 매우 중요하다. 그러나 UPnP는 TCP/IP 기반의 프로토콜을 사용하기 때문에 사용자 인터랙션에 의한 전송 지연 시간이 발생한다. 그러므로 본 논문에서는 셋탑 박스에 최적화된 UPnP AV 시스템을 구현하고 AV 컨텐츠의 효율적인 재생을 위한 UPnP프레임워크의 재생 기법을 제안한다. UPnP AV 시스템은 미디어 서버가 있고, 미디어 서버에 저장된 비디오/오디오 파일 등을 전송받아 재생하기 위한 셋탑 박스에는 컨트롤 포인트와 UPnP 미디어 렌더러의 통합 모델이 탑재된다. 이러한 기법은 다양한 기기를 제어하고 호환성을 제공하는 UPnP프레임워크가 될 수 있다.

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TCP Congestion Control of Transfer Rate-based in End-to-End Network Systems (종단간 네트워크 시스템에서 전송율 기반 TCP 혼잡제어)

  • Bae, Young-Geun;Yoon, Chan-Ho;Kim, Gwang-Jun
    • The Journal of the Korea institute of electronic communication sciences
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    • v.1 no.2
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    • pp.102-109
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    • 2006
  • In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.

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Designing Distributed Real-Time Systems with Decomposition of End-to-End Timing Donstraints (양극단 지연시간의 분할을 이용한 분산 실시간 시스템의 설계)

  • Hong, Seong-Soo
    • Journal of Institute of Control, Robotics and Systems
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    • v.3 no.5
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    • pp.542-554
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    • 1997
  • In this paper, we present a resource conscious approach to designing distributed real-time systems as an extension of our original approach [8][9] which was limited to single processor systems. Starting from a given task graph and a set of end-to-end constraints, we automatically generate task attributes (e.g., periods and deadlines) such that (i) the task set is schedulable, and (ii) the end-to-end timing constraints are satisfied. The method works by first transforming the end-to-end timing constraints into a set of intermediate constraints on task attributes, and then solving the intermediate constraints. The complexity of constraint solving is tackled by reducing the problem into relatively tractable parts, and then solving each sub-problem using heuristics to enhance schedulability. In this paper, we build on our single processor solution and show how it can be extended for distributed systems. The extension to distributed systems reveals many interesting sub-problems, solutions to which are presented in this paper. The main challenges arise from end-to-end propagation delay constraints, and therefore this paper focuses on our solutions for such constraints. We begin with extending our communication scheme to provide tight delay bounds across a network, while hiding the low-level details of network communication. We also develop an algorithm to decompose end-to-end bounds into local bounds on each processor of making extensive use of relative load on each processor. This results in significant decoupling of constraints on each processor, without losing its capability to find a schedulable solution. Finally, we show, how each of these parts fit into our overall methodology, using our previous results for single processor systems.

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Automated Maintenance Unmanned Monitoring System Using Intelligent Power Control System (지능형 전원제어장치를 이용한 자동화 유지보수 무인감시시스템)

  • Cha, Min-Uk;Lee, Choong Ho
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2021.05a
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    • pp.237-239
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    • 2021
  • Failure and malfunction of the unmanned surveillance facility cost can lead to delays occurring until the person in charge arrives at the unmanned surveillance facility, and theft, damage, and information leakage damage caused by intruders. In addition, due to equipment failure and malfunction, additional costs are incurred due to constant inspection by the manager. In this paper, in order to compensate for the malfunction of unmanned facility costs, we propose a system that diagnoses the monitoring facility in real time, displays the contents of the problem, automatically restores the facility power, and informs the person in charge of the situation by text message. The proposed system is a surveillance facility consisting of main facilities such as video equipment (CCTV), sound equipment, floodlights, etc. And SMS server that can send text messages in real time. Through experiments, the effectiveness of the proposed system was verified.

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Sample size determination based on placements for non-inferiority trials (비열등성 시험에서 위치 방법에 기초한 표본 수 결정)

  • Kim, Jiyeon;Kim, Dongjae
    • Journal of the Korean Data and Information Science Society
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    • v.24 no.6
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    • pp.1349-1357
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    • 2013
  • In clinical research, sample size determination is one of the most important things. There are parametric method using t-test and non-parametric method suggested by Kim and Kim (2007) based on Wilcoxon's rank sum test for determining sample size in non-inferiority trials. In this paper, we propose sample size calculation method based on placements method suggested by Orban and Wolfe (1982) and using the power calculated by Kim (1994) in non-inferiority trials. We also compare proposed sample size with that using Kim and Kim (2007)'s formula and that of t-test for parametric methods. As the result, sample size calculated by proposed method based on placements is the smallest. Therefore, proposed method based on placements is better than parametric methods in case that it's hard to assume specific distribution function for population and also more efficient in terms of time and cost than method based on Wilcoxon's rank sum test.

A Traffic Management Scheme for the Scalability of IP QoS (IP QoS의 확장성을 위한 트래픽 관리 방안)

  • Min, An-Gi;Suk, Jung-Bong
    • Journal of KIISE:Information Networking
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    • v.29 no.4
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    • pp.375-385
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    • 2002
  • The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

A Steady State Analysis of TCP Rate Control Mechanism on Packet loss Environment (전송 에러를 고려한 TCP 트래픽 폭주제어 해석)

  • Kim, Dong-Whee
    • Journal of Korea Society of Industrial Information Systems
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    • v.22 no.1
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    • pp.33-40
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    • 2017
  • In this Paper, Analyse the Steady State Behavior of TCP and TFRC with Packet Error when both TCP and TFRC Flows Co-exist in the Network. First, Model the Network with TCP and TFRC Connections as a Discrete Time System. Second, Calculate Average Round Trip Time of the Packet Between Source and Destination on Packet Loss Environment. Then Derive the Steady State Performance i.e. Throughput of TCP and TFRC, and Average Buffer Size of RED Router Based on the Analytic Network Model. The Throughput of TCP and TFRC Connection Decrease Rapidly with the Growth of Sending Window Size and Their Transmission Rate but Their Declines become Smoothly when the Number of Sending Window Arrives on Threshold Value. The Average Queue Length of RED Router Increases Slowly on Low Transmission Rate but Increases Rapidly on High Transmission Rate.

An Efficient MAC Protocol for Supporting Multimedia Services in APON (APON에서 멀티미디어 전송을 위한 효율적인 MAC 프로토콜)

  • 은지숙;이호숙;윤현정;소원호;김영천
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.25 no.1A
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    • pp.132-141
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    • 2000
  • In this paper, we proposed the MAC protocol of APON supporting multi-class traffic such as CBBUVBR, ABR, UBR, to guarantee the required QoS of each service. For this, we analyze the performance of variousrequest mechanisms and employee the different request mechanism for each traffic classes. Upstream anddownstream frame structures to minimize transmission overhead are proposed based on our request mechanism.The proposed MAC protocol applies the different priority to permit distribution process. CBBWBR traffic, withthe stringent requirements on CDV or delay, is allocated prior to any other class. ABR traffic, which hasnon-strict CDV or delay criteria, uses flexibly the available bandwidth but ensures a minimum cell rate (MCR).UBR traffic is allocated with lowest priority for the remaining capacity. The performance of proposed protocol isevaluated in terms of transfer delay and 1-point CDV with various offered load. The result of simulation showsthat the proposed protocol guarantees the required QoS of the corresponding category, while making use of theavailable resources in both an efficient and dynamical way.

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A Distributed Real-Time Concurrency Control Scheme using Transaction the Rise of Priority (트랜잭션 우선 순위 상승을 이용한 분산 실시간 병행수행제어 기법)

  • Lee, Jong-Sul;Shin, Jae-Ryong;Cho, Ki-Hyung;Yoo, Jae-Soo
    • Journal of KIISE:Databases
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    • v.28 no.3
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    • pp.484-493
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    • 2001
  • As real-time database systems are extended to the distributed computing environment, the need to apply the existing real-time concurrency control schemes to the distributed computing environment has been made. In this paper we propose an efficient concurrency control scheme for distributed real-time database system. Our proposed scheme guarantees a transaction to commit at its maximum, reduces the restart of a transaction that is on the prepared commit phase, and minimizes the time of the lock holding. This is because it raises the priority of the transaction that is on the prepared commit phase in the distributed real-time computing environment. In addition, it reduces the waiting time of a transaction that owns borrowed data and improves the performance of the system, as a result of lending the data that the transaction with the raised priority holds. We compare the proposed scheme with DO2PL_PA(Distributed Optimistic Two-Phase Locking) and MIRROR(Managing Isolation in Replicated Real-time Object Repositories) protocol in terms of the arrival rate of transactions, the size of transactions, the write probability of transactions, and the replication degree of data in a firm-deadline real-time database system based on two-phase commit protocol. It is shown through the performance evaluation that our scheme outperforms the existing schemes.

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Design and Performance Analysis of Dynamic QoS Control for RTP-based Multimedia Data Transmission (RTP 기반 멀티미디어 데이터 전송을 위한 동적 QoS 제공방안의 설계 및 성능 분석)

  • Moon, Young-Jun;Ryoo, In-Tae;Park, Gwang-Hoon
    • The KIPS Transactions:PartC
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    • v.10C no.7
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    • pp.891-898
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    • 2003
  • This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.