• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.57-60
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• 2002

본 논문은 VoIP 시스템에서 적용 가능한 QoS 구조를 연구하였다. 현재 망 구조에서 실시간 트래픽은 최선형 트래픽과 구별이 되지 않아 보장받지 못한다. 따라서 본 논문에서는 QoS 구조를 위해 IETF에서 제안된 차등화서비스와 Intserv over Diffserv를 개관하였고, 이런 환경에서 VoIP 트래픽에 대한 품질을 보장하기 위한 QoS 구조를 연구하였다 또한, 측정기반 수락제어에 의해 기존망에 비해. 보다 좋은 품질을 제공할 수 있는 VoIP 시스템 구조도 함께 연구하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04a
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• pp.524-526
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• 2003

지문은 사람에 있어서 유일한 특징과 변하지 않는 특성으로 인하여 개인의 인증이나 식별에 많이 사용되고 있다. 이러한 특성을 가진 지문을 스마트카드의 인증과정에 이용함으로써 보안성을 높이고 사용에 편리함을 추가할 수 있다. 본 논문에서는 스마트카드와 서비스시스템 간에 지문을 이용한 인증시스템을 제안하였다. 제안된 시스템은 카드와 서비스시스템이 각각 특징점 정보와 방향성정보를 매칭 함으로서 스마트카드의 계산부하를 줄일 수 있고, 본인 거부율과 타인 수락율의 오류를 줄일 수 있다.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1788-1796
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• 1997

In this paper, we propose a new connection admission control(CAC) algorithm for traffic control in ATM network in which traffic estimation is performed based on user-specified parameters at every moment of connection request or connection release by recursive formula which makes real-time calculation possible. And traffic estimation using cell flow measurement is carried out when the number of connectioned calls does not change during a measurement reflection period. Performance analysis of the proposed method is carried out using several aspects for homogeneous and heterogeneous bursty traffic. The results showed that the proposed CAC method revealed better performance, than conventional CAC method for burst model in both utilization and QoS point of view.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8B
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• pp.681-688
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• 2008

In this paper, we propose a new price-based call admission control algorithm for multi-class communication networks. When a call arrives at the network, it informs the network of the number of requested circuits and the minimum amount of time that it will require. The network provides the optimal price for the arrived call with which it tries to maximize its expected revenue. The optimal price is dynamically adjusted based on the information of the arrived call, and the present and the estimated future congestion level of the network during the reservation time of the call. If the call accepts the price, it is admitted. Otherwise, it is rejected. We compare the performance of our dynamic pricing algorithm with that of the static pricing algorithm by Courcoubetis and Reiman [1], and Paschalidis and Tsitsiklis [2]. By the comparison, we show that our dynamic pricing algorithm has better performance aspects such as higher call admission ratio and lower price than the static pricing algorithm, although these two algorithms result in almost the same revenue as shown in [2]. This implies that, in the competitive situation, the dynamic pricing algorithm can attract more users than the static pricing algorithm, generating more revenue. Moreover, we show that if a certain fixed connection fee is introduced to the price for a call, our dynamic pricing algorithm yields more revenue.

• The Journal of the Korea Contents Association
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• v.6 no.9
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• pp.1-8
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• 2006

In a CDMA system, minimizing handoff calls is an important factor for user satisfaction. In this paper, we divide the cell area into two regions: normal region and soft handoff region. And we show that differentiating the guard channel numbers in those two regions make handoff calls minimized. We develop an SRN based cell model consists of normal and soft handoff region and analyze the performance of the model.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.1
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• pp.1-13
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• 2001

In this paper, we propose a new dynamic bandwidth allocation and call admission control for VBR video sources with QoS constraints to support an efficient resource management and at the same Lime to satisfy the user's quality or service requirements. For the dynamic bandwidth allocation, first the next amount of traffic is predicted using a modified adaptive linear prediction method that considers abrupt scene change effects. And then, we dynamically allocate the necessary bandwidth to each connection based on the currently provided quality degree by the network with respect to the user's QoS requirements in terms of average delay and loss ratio. For the admission control, we determine the acceptance or rejection or a new connection based on the quality satisfaction degrees of the existing connections. Simulation results show that our proposed dynamic schemes are able to provide a stable service, which well meets the user's quality requirements.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.20-33
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• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.

• Journal of KIISE:Information Networking
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• v.34 no.5
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• pp.396-404
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• 2007

In this paper, an Ad-hoc Routing Protocol which works in wireless Ad-hoc communication networks with multiple radios and multiple channels, and controls call admission based on bandwidth measurement is proposed. Unlike the conventional Ad-hoc node with a single radio using a single channel, an Ad-hoc node of the protocol proposed, MCQosR(Multiple Channel Quality of Service Routing), has multiple radios and uses multiple channels, which makes full duplex transmission between wireless Ad-hoc nodes, and reduces the intra interference on a route. Also, a fixed channel only for reception at each node enables the measurement of the available bandwidth, which is used to control the call admission for QoS provision. The performance of MCQosR is verified by simulation.

• KSCE Journal of Civil and Environmental Engineering Research
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• v.14 no.6
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• pp.1273-1279
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• 1994

There are 3 cases that only permissive left-turn(PLT) vehicles use the possible lane for PLT. In these cases, left turn and through movements can not be included in the same lane group, hence saturation flow rate and left turn adjustment factor of PLT are obtained separately from through movement. In capacity analysis procedures at signalized intersection with PLT phasing, PLT capacity should be known to discriminate among 3 cases stated above. The capacity is directly used not only to get saturation flow rate and left turn adjustment factor, but as a threshold for the feasibility of PLT control. This study calculated through field data the critical gap and minimum headway of left turn which affect the PLT capacity. The capacity was obtained by using theoretical models, which consequently could be used to calculate the saturation flow rate and left turn adjustment factor.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.270-277
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• 2006

In the wireless adaptive modulation and coding(AMC) systems, the modulation type of user's connection can be changed dynamically. and the ongoing connection might fail due to the change of modulation. In this paper, we approach the AMC-induced CAC problem by focusing on the guaranteed connection. Three kinds of calls, new, handoff, and modulation-changed calls, are considered. We propose a modified guard channel CAC scheme that allows the modulation-changed and handoff calls to use the guard channel. Then we analyze a Markov model for the CAC scheme with long-term AMC in mind. According to the simulation results, the proposed approach reduces the call dropping probability for modulation-changed calls, which suggests the threshold of guard channels can be determined based on the proposed approach.