• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9B
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• pp.610-618
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• 2005

As developed Internet and Computer Capability, Many Users take the many information through the network. So requirement of User that use to network was rapidly increased and become various. But it spend much time to accept user requirement on current network, so studied such as Active network for solved it. This Active node on Active network have the capability that stored and processed execution code aside from capability of forwarding packet on current network. So required execution code for executed packet arrived in active node, if execution code should not be in active node, have to take by request previous Action node and Code Server to it. But if this execution code take from previous active node and Code Server, bring to time delay by transport execution code and increased traffic of network and execution time. So, As used execution code stored in cache on active node, it need to increase execution time and decreased number of request. So, our paper suggest ANC caching technique that able to decrease number of execution code request and time of execution code by efficiently store execution code to active node. ANC caching technique may decrease the network traffic and execution time of code, to decrease request of execution code from previous active node.

• Journal of KIISE:Information Networking
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• v.29 no.6
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• pp.595-612
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• 2002

One of the most important issues in guaranteeing the high degree of QoS on mobile computing is how to reduce hand-off drops caused by lack of available bandwidth in a new cell. Each cell can request bandwidth reservation to its adjacent cells for hand-off calls. This reserved bandwidth can be used only for hand-offs, not for new calls. It is also important to determine how much of bandwidth should be reserved for hand-off calls because reserving too much would increase the probability of a new call being blocked. Therefore, it is essential to develop a new mechanism to provide QoS guarantee on a mobile computing environment by reserving an appropriate amount of bandwidth and call admission control. In this paper. bandwidth reservation and call admission control mechanisms are proposed to guarantee a consistent QoS for multimedia traffics on a mobile computing environment. For an appropriate bandwidth reservation, we propose an adaptive bandwidth reservation mechanism based on an MPP and a 2-tier cell structure. The former is used to predict a next move of the client while the latter to apply our mechanism only to the client with a high hand-off probability. We also propose a call admission control that performs call admission test only on PNC(Predicted Next Cell) of a client and its current cell. In order to minimize a waste of bandwidth caused by an erroneous prediction of client's location, we utilize a common pool and QoS adaptation scheme. In order evaluate the performance 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-CAT2, FR-CAT2, and AR-CAT2.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.4
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• pp.171-180
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• 2004

CDP(handoff Call Dropping Probability) and CBP(new Call Blocking Probability) have been used as two important call level QoS parameters in cellular mobile communications. But, many methods to reduce CDP without considering CBP have been studied, and hand-off call priority scheme has been introduced. But the use of hand-off call priority scheme increases CBP and decreases channel utilization rate depending on the number of reserved channel for priority. In this paper, we propose a CAC(Call Admission Control) algorithm with overflow and preemption to solve the problem caused by considering CDP and CBP in calculation of the number of channel reserved. The problem is the increase of CDP as the traffic load increases. In our CAC algorithm, hand-off call is permitted to use(overflow) unreserved and unused channel if there is no reserved and unused channel, and new call is permitted to use(preemption) the channel overflowed by hand-off call if there is no unreserved and unused channel. This mechanism of calculation of the number of reserved channel and CAC algorithm is expected to increase channel utilization rate, and can be applied to media-based QoS provision in cellular mobile communications.

• The KIPS Transactions:PartC
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• v.12C no.4 s.100
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• pp.597-602
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• 2005

In this study, file size for measurement and the service system's billing data were submitted to a comparative analysis by performing a verification test on the billing system of three major mobile communication services providers, based on the wireless Internet service packet. As shown in the result of the verification test, there were some differences in the billing data due to transmission overhead, according to the network quality that is affected by the wireless environment of mobile operators. Consequently, the packet analysis system was proposed as a means of applying consistent packet billing to all service providers being compared. If the packet analysis system is added to supplement the current billing system various user requirements can be met. Billing by Packet among mobile operators and differentiated billing based on the content value are available, since the packet data can be extracted through protocol analysis by service, and it can be classified by content tape through traffic data analysis. Furthermore, customer's needs can be satisfied who request more information on the detailed usage, and more flexible and diverse billing policies can be supported like application of charging conditions to the non-charging packet handling. All these services are expected to contribute to the popularization of the wireless Internet service, since user complaints about the service charge could be reduced.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.975-983
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• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11B
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• pp.946-955
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• 2004

It is shown that the do Bruijn graph (dBG) can be used as an architecture for interconnection network and a suitable structure for parallel computation. Recent works have classified dBG based routing algorithms into shortest path routing and fault tolerant routing but investigation into fault free shortest path (FFSP) on dBG has been non-existent. In addition, as the size of the network increase, more faults are to be expected and therefore shortest path dBG algorithms in fault free mode may not be suitable routing algorithms for real interconnection networks, which contain several failures. Furthermore, long fault free path may lead to high traffic, high delay time and low throughput. In this paper we investigate routing algorithms in the condition of existing failure, based on the Bidirectional do Bruijn graph (BdBG). Two FFSP routing algorithms are proposed. Then, the performances of the two algorithms are analyzed in terms of mean path lengths and discrete set mean sizes. Our study shows that the proposed algorithms can be one of the candidates for routing in real interconnection networks based on dBG.

• The KIPS Transactions:PartC
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• v.11C no.7 s.96
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• pp.1033-1042
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• 2004

According to the arbitration mechanism among the peers in the network, the P2P networking environments can be classified into hybrid P2P networking environments and pure P2P networking environments. In hybrid P2P networking environments, each peer gets continual services from the servers that arc operational most of the time, and so, network isolation does not occur because every peer can always keep connection to the server. In pure P2P networking environments, however, every peer directly connects to another peer without server intervention, and so, network isolation can occur when the per mediating the connection is terminated. In this paper, we propose a scheme for each peer to keep connection information of other peers by maintaining IDs of its neighbor peers, to reconnect to another peers when the mediating peer fails to work. and, for efficiency. to balance the number of connections that should be maintained by each peer. With our mechanism, each pier in the network can continuously maintain connection to the network and get seamless services from other peers. Through the simulation, we ascer-tained that network isolation does not occur in the pure P2P network adopting our mechanism and that our mechanism distributes and balances the connections that are maintained by each peer. We also analyzed the total network traffic and the mean number of hops for the connections made by each peer according to the recommended number of connections that is established at system setup time.

• 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 the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.29-36
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• 2012

In Wireless Sensor Network(WSN)s, the detection of precise location of each node is essential for utilizing sensing data acquired from sensor nodes effectively. Among various location methods, the received signal strength(RSS) based localization scheme is mostly preferable in many applications because it can be easily implemented without any additional hardware cost. Since a RSS-based localization scheme is mainly affected by radio channel or obstacles such as building and mountain between two nodes, the localization error can be inevitable. To enhance the accuracy of localization in RSS-based localization scheme, a number of RSS measurements are needed, which results in the energy consumption. In this paper, a RSS based localization using Bayesian Compressive Sensing(BSS) with path-loss exponent estimation is proposed to improve the accuracy of localization in the energy-efficient way. In the propose scheme, we can increase the adaptative, reliability and accuracy of localization by estimating the path-loss exponents between nodes, and further we can enhance the energy efficiency by the compressive sensing. Through the simulation, it is shown that the proposed scheme can enhance the location accuracy of multiple unknown nodes with fewer RSS measurements and is robust against the channel variation.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.