• Journal of KIISE:Computer Systems and Theory
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• v.26 no.7
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• pp.827-838
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• 1999

멀티미디어 통신에서 통신 성능에 관련된 요구 사항들은 QoS 매개 변수들로서 서술된다. QoS매개 변수들에서 중요한 매개 변수 중의 하나가 전송 신뢰성이다. QoS 매개 변수로서의 신뢰성은 오류 감지, 보고 그리고 정정 기법으로 정의된다. 하지만 기존의 오류 제어 기법들은 멀티미디어 데이타의 통합된 관점을 고려하지 않았다. 그래서 우리는 MEC(multimedia-oriented error control)라고 명명된 오류 제어 기법을 제안하였다. 1 본 논문에서는 MEC기법의 성능을 조사 하였다. 성능평가 결과는 MEC기법이 기존의 오류 제어 기법보다도 낮은 전송지연(lower delay)과 호손율(blocking probability) 을 갖는다는 것을 볼 수 있었다. 결국 제안된 MEC기법은 수송 프로토콜에게 유연성과 높은 성능을 갖도록 해준다.Abstract Communication performance requirements are described as QoS parameters in multimedia communication. One of the important QoS parameters is the reliability of the transfer. As a QoS parameter, the reliability defines error detection, report and correction mechanisms. Conventional error control mechanisms, however, do not consider the integrated viewpoint of multimedia data. So we have proposed the MEC(multimedia-oriented error control). 1 In this paper, we have investigated the performance evaluation of the MEC. The results show that the MEC mechanism provides lower transfer delay and blocking probability than those of the conventional error recovery mechanism. Therefore, the proposed MEC mechanism makes the transport protocol have the flexibility and high performance.

• ETRI Journal
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• v.20 no.3
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• pp.251-271
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• 1998

It is well known that if usage parameter control/network parameter control (UPC/NPC) functions are used together with a cell loss priority control scheme in ATM networks, the measurement phasing problem can occur. This makes it difficult for a network provider to define and commit the cell loss ratio as a QoS parameter. To solve the problem, we propose a new UPC/NPC algorithm. By using the proposed UPC/NPC algorithm, we can define the cell loss ratios for CLP = 0 and CLP = 0+1 cell streams without the measurement phasing problem under any conditions. We analyzed the performance of the proposed UPC/NPC algorithm. Using a discrete time model for the UPC/NPC architecture with a discrete-time semi-Markov process (DSMP) input model, we obtained the cell discarding probabilities of CLP = 0 and CLP = 0+1 cells streams and showed that more CLP = 0 cells are accepted compared to what was proposed in ITU-T.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.2
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• pp.455-461
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• 2008

Enhanced Distributed Channel Access (EDCA) is a channel access scheme adopted by the IEEE 802.11e draft standard for QoS-enabled wireless local area networks. It classifies traffic into separate Access Categories (ACs) and achieves service differentiation by allowing each AC to have its own values of channel access parameters. This paper proposes a scheme to dynamically adapt the EDCA parameters to traffic environment so that they both maximize the throughput of non real-time traffics and meet the delay and throughput constraints of real-time traffics. For this purpose, we develop a design algorithm for efficient exploration of the EDCA parameter space. Using the algorithm, we derive a Pareto curve fur delay-throughput trade-off in each anticipated traffic environment. The Pareto database can then be used to optimally adjust the parameter according to the traffic environment and administrative policies. Simulation results show that compared with the default parameter configuration, the proposed scheme is better for delay, throughput guarantee and can improve the throughput of non real-time traffics by 12% on average.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.86-89
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• 1998

Multiple types of information in a multimedia service are delivered though multiple virtual connections on ATM network, while each virtual connection may be controlled independently. A multimedia service requires an associative relationship among multiple information streams to provide required harmonization. There may be required additional traffic descriptors to guarantee the required harmonization among multiple information streams in a multimedia service. For buffering of large bandwidth information stream(e.g., video), extremely large buffer size is necessary, but this approach should not be efficient way to compensate a severely delayed cells/blocks experienced at network. The best way to solve this problem will be minimization of relative-delayed-transfer of cells/blocks to application processes through ATM network control. To minimize a delayed transfer the mapping between relative delay parameter(i.e., associative Group QoS parameters) and per-VC traffic descriptor will be necessary. This paper is present additional functions and parameters to interpret the mapping between relative delay parameters(i.e., associative Group QoS parameters) and per-VC traffic descriptors in ATM API for multimedia application services.

• Convergence Security Journal
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• v.8 no.2
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• pp.95-102
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• 2008

In this paper, we proposed topology aggregation method that efficiently aggregating topology information in simple node topology. And, it improved QoS of networks by improving call success rate and call access time. Proposed method can improving performance of network by decreasing aggregation information and aggregating multi-links information between boundary nodes using the line segment scheme within bandwidth and delay parameter. To evaluate performance of the proposed scheme, we compare/analyze the current method with the proposed scheme in respect to call success rate, access time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5B
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• pp.451-463
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• 2011

QoE is defined as a quality perceived by users on a certain service. However, standard method to measure voice QoE(MOS) has been developed, but no standard method to measure video QoE has been defined. In this paper, we propose an efficient algorithm to measure video QoE s automatically for IPTV services. The proposed method selects candidate scenarios that affect the users' MOS directly, and derives weight factors for the selected scenarios. With the weight factors for the scenarios, video QoE value is calculated. For the validation of the proposed algorithm, we made degraded videos reflecting the parameters. With the degraded videos, by comparing the user perceived MOSs with the video QoEs derived by the proposed algorithm, we show that the two values are highly correlated each other.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.11b
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• pp.907-910
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• 2003

이동통신 기술이 발전함에 따라 이동통신 네트워크를 통한 서비스들이 다양해지고, 사용자들의 수는 점점 늘어가고 있다. 또한 사용자들은 일반적으로 이동통신 서비스에 대해 유선 망과 동등한 수준의 품질을 기대한다. 그러나, 이동통신망은 유무선 통합망으로 구성되어 있으며, 이들 복잡한 구성을 갖는 네트워크에 대한 서비스 품질 보장은 유선망에 비해 훨씬 어렵다. 이의 결과로, 이동통신 서비스 네트워크의 트래픽은 과거에 비해 폭발적으로 증가하였다. 따라서, 네트워크 사업자와 서비스 제공자들은 서비스의 성능 문제에 직면하고 있으며, 네트워크 사업자나 서비스 제공자들은 효과적인 서비스 품질관리 기술을 강력하게 요구하고 있다. QoS 감시는 QoS 제공과 보장을 위한 기본적인 기술로서, 실제 네트워크에서 QoS 감시를 위해서는 네트워크 및 서비스 성능 인자들과 QoS 인자들의 관계를 식별해야 한다. 본 논문에서는 서비스와 네트워크 성능인자 그리고, QoS 인자들간의 관계를 QoS metrics로 정의하며, 각 인자들의 관계는 계층적인 그래프로 나타낸다. QoS metrics의 정의와 이에 따른 계층적 그래프의 구성을 통해 세 가지 이점을 기대 할 수 있다. 첫째, 네트워크 사업자들은 QoS 저하의 주요 원인을 신속하게 식별 할 수 있다. 둘째, 네트워크 사업자들과 서비스 제공자들은 주관적인 QoS 를 수치 적인 성능 지표를 통해 측정이 가능하다. 마지막으로, QoS metrics 는 네트워크 사업자들과 서비스 제공자들이 QoS 감시 활동의 결과에 따라 그들의 네트워크를 재구성하는 데 도움을 주며 E2E QoS 제공에 효율성을 가져다 준다.현을 정형화하기 위해 Oolong 코드의 명령어들을 문법으로 작성하였으며, PGS를 통해 생성된 어휘 정보를 가지고 스캐너를 구성하였으며, 파싱테이블을 가지고 파서를 설계하였다. 파서의 출력으로 AST가 생성되면 번역기는 AST를 탐색하면서 의미적으로 동등한 MSIL 코드를 생성하도록 시스템을 컴파일러 기법을 이용하여 모듈별로 구성하였다.적용하였다.n rate compared with conventional face recognition algorithms. 아니라 실내에서도 발생하고 있었다. 정량한 8개 화합물 각각과 총 휘발성 유기화합물의 스피어만 상관계수는 벤젠을 제외하고는 모두 유의하였다. 이중 톨루엔과 크실렌은 총 휘발성 유기화합물과 좋은 상관성 (톨루엔 0.76, 크실렌, 0.87)을 나타내었다. 이 연구는 톨루엔과 크실렌이 총 휘발성 유기화합물의 좋은 지표를 사용될 있고, 톨루엔, 에틸벤젠, 크실렌 등 많은 휘발성 유기화합물의 발생원은 실외뿐 아니라 실내에도 있음을 나타내고 있다.>10)의 $[^{18}F]F_2$를 얻었다. 결론: $^{18}O(p,n)^{18}F$ 핵반응을 이용하여 친전자성 방사성동위원소 $[^{18}F]F_2$를 생산하였다. 표적 챔버는 알루미늄으로 제작하였으며 본 연구에서 연구된 $[^{18}F]F_2$가스는 친핵성 치환반응으로 방사성동위원소를 도입하기 어려운 다양한 방사성의 약품개발에 유용하게 이용될 수 있을 것이다.었으나 움직임 보정 후 영상을 이용하여 비교한 경우, 결합능 변화가 선조체 영역에서 국한되어 나타나며 그 유

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.2
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• pp.45-53
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• 2000

In wireless environment, due to the limited capacity of radio channels it is not easy to guarantee QoS provisioning to mobile users. Therefore, one of the key problems to support broadband multimedia multi-services in wireless ATM networks is to study an effective call admission control(CAC). The purpose of this paper is to propose a distributed CAC scheme that guarantees multi QoS and multi-class service. Control parameters of the proposed scheme are QoS threshold and channel overload probability. With these parameter control, we show that the scheme can guarantee the requested QoS to both new and handover calls. In the scheme, channels are allocated dynamically, and QoS measurements are made in a distributed manner. We show that by providing variable data rate to calls it can effectively prohibit the QoS degradation even if there are severe fluctuations of network traffic. We compare the proposed CAC scheme to the well-known schemes such as guard band call admission control scheme. Through numerical examples and simulations, the proposed scheme is shown to improve the performance by lowering the probability of handover call dropping

• The Transactions of the Korea Information Processing Society
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• v.4 no.6
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• pp.1433-1442
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• 1997

Algorithm design is needed to optimized bandwidth which satisfy quality of service(QoS) requirements of vary traffic classes for Multimedia service in ATM networks. The diverse flow characteristics e.g., burstiness, bit rate and burst length, have to guarantee the different quality of service(QoS) requirements in Asynchronous Transfer Mode(ATM). The QoS parameter may be measured in terms of cell loss probability and maximum cell dely. In this paper, we consider the ATM networks which the virtual path(vip) concept is implemented by applying the Markov Modulated Deterministic Process method. We develop an efficient algorithm to computer the minimum capacity required to satisfy all the QoS requirements when multiple classes of on-off source are multiplexed on single VP. Using above the result, we propose a simple algorithm to determine the VP combination to achieve the near optimum of total capacity required for satisfying the individual QoS requirements. Numerical results are also presented to demonstrate the performance of the algorithm, when compared to the optimal total capacity required.