• The KIPS Transactions:PartC
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• v.10C no.3
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• pp.305-310
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• 2003

The performance of various GFR implementations has been recently studied due to the interest to provide bandwidth guarantees with a simpler implementation than ABR in ATM networks. One of the important factors is buffer management for guaranteeing QoS in GFR service. An efficient buffer management algorithm is necessary to guarantee MCR for untagged cell in ATM switch. In this paper, we propose and evaluate a buffer management scheme to provide the GFR service guarantees. The proposed scheme can control the cell discarding for fairness in each VC, and compared with Double-EPD and DFBA. Our results show that the proposed buffer management with per-VC queuing achieves significant enhancement on goodputs and fairness index than those of existing methods.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.4
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• pp.246-251
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• 2013

In this paper, it is proposed a bandwidth allocation Scheme based on Bankruptcy theory in Distributed Mobile Multimedia Network. The proposed scheme is guaranteed a minimum allocation. So, the minimum quality of each service are guaranteed. Therefore efficient and fairness network can be configured. The performance evaluation results indicate that the proposed scheme has good performance than other existing schemes by the fairness index and the Erlang blocking formular calculation. The minimum bandwidth of the proposed scheme can be applied to other techniques of a priority based bandwidth allocation scheme and dynamic bandwidth allocation scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.12
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• pp.15-21
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• 2004

The Guaranteed Bandwidth Rate scheme has been designed to accomodate non-real-time applications, such as Internet based traffic in Pm networks. The Guaranteed Bandwidth Rate scheme not only guarantees a minimum throughput at the frame level, but also supports a fair share of available resources. In this paper, we propose a cell control algorithm which can improve the fairness and the throughput through the traffic control in PM networks. For the evaluation of the proposed scheme, we compare proposed scheme with the existing scheme in the fairness and the throughput. Simulation results show that proposed scheme can improve the fairness and throughput than the existing scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.9-16
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• 2005

A new online routing algerian is proposed in this paper, which use the interference-prediction to solve the network congestion originated from extension of Internet scope and increasing amount of traffic. The end-to-end QoS has to be guaranteed in order to satisfy service level agreements (SLAs) in the integrated networks of next generation. For this purpose, bandwidth is allocated dynamically and effectively, moreover the path selection algorithm is required while considering the network performance. The proposed algorithm predicts the level of how much the amount of current demand interferes the future potential traffic, and then minimizes it. The proposed algorithm considers the bandwidth on demand, link state, and the information about ingress-egress pairs to maximize the network performance and to prevent the waste of the limited resources. In addition, the interference-prediction supports the bandwidth guarantee in dynamic network to accept more requests. In the result, the proposed algorithm performs the effective admission control and QoS routing. In this paper, we analyze the required conditions of routing algorithms, the aspect of recent research, and the representative algorithms to propose the optimized path selection algorithm adequate to Internet franc engineering. Based on these results, we analyze the problems of existing algorithms and propose our algorithm. The simulation shows improved performance by comparing with other algorithms and analyzing them.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.441-449
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• 2002

For multimedia traffics to be supported successfully in wireless network environment, it is necessary to provide Qualify-of-Service(QoS) guarantees among mobile hosts(clients). In order to guarantee the QoS, we have to keep the call blocking probability below a target value during hand-off session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic of a new cell, since on service mobile clients should be able to continue their sessions. In this paper, we propose an efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. Proposed algorithm predicts the direction of clients in a cell and adjusts the amount of the channel to be reserved according to the load status of the cell. This method is used to reserve a part of bandwidths of a cell for hand-off calls to its adjacent cells and this reserved bandwidth can be used for hand-off call prior to new connection requests. If the number of free channels is also under a low threshold value, our scheme use a load-balancing algorithm with an adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure metrics such as the blocking probability of new calls and dropping probability of hand-off calls, and compare with those of existing schemes.

• Journal of information and communication convergence engineering
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• v.1 no.3
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• pp.109-114
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• 2003

There is an emergence of Internet applications that have real-time requirements. These applications require IP to support guaranteed capacity, higher priority and lower packet loss rate. To address this, the Internet Engineering Task Force (IETF) is developing a set of protocols and standards for Integrated Services on the Internet. Using RSVP and policies to manage the allocation of network resources in order to provide different levels of service is a topic of great interest to service providers. Currently, it is not possible to dynamically reallocate resources during an application's session. This paper discusses how policies in conjunction with new service class can provide a more enhanced network resource allocation by allowing for this dynamic reallocation.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.25-28
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• 2000

As Internet users and network traffics are explosively increased, many researches have been done to provide fast and guaranteed Internet services. The results of these efforts are Multiprotocol Lab el Switching(MPLS) and differentiated service(Diff serv) emerging as a next generation backbone net work. In this paper, we make a research on gearing (or interworking) two technology needed to support end-to-end QoS and explain sequential procedure. And we simulate the L-LSP/E-LSP proposed by Working Group of IETF to be deployed to the DiffServ module which is implemented based on the MPLS network simulator and analyze the performance of the simulation result.

• Proceedings of the Society of Korea Industrial and System Engineering Conference
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• 2002.05a
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• pp.105-111
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• 2002

Presented is a Genetic Algorithm(GA) for dynamic partitioning an ATM LANE(LAN Emulation) network. LANE proves to be one of the best solutions to provide guaranteed Quality of Service(QoS) for mid-size campus or enterprise networks with a little modification of legacy LAN facilities. However, there are few researches on the efficient LANE network operations to deal with scalability issues arising from broadcast traffic delivery. To cope with this scalability issue, proposed is a decision model named LANE Partitioning Problem(LPP) which aims at partitioning the entire LANE network into multiple Emulated LANs(ELANs), each of which works as an independent virtual LAN.

• Journal of Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1465-1468
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• 2003

본 연구는 실시간 서비스 트래픽, 즉 보장형 서비스를 위한 스케줄링 알고리즘들에 대한 성능 분석에 대한 연구이다. 특히 실시간 데이터 전송의 경우, 작은 지연 시간을 요구하면서 안정된 QoS를 요구하고 있다. 기존에 알려진 FQ, WFQ, WF2Q, Virtual Clock 스케줄링 알고리즘들을 사용해서 대기류의 수학적 모델이 아닌 시뮬레이션 도구를 사용하여, 지연에 민감한 보장형 서비스 트래픽에 대한 시간 복잡도, 공정성, 처리율 측면으로 성능을 분석하였다.