• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.8
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• pp.2796-2813
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• 2014

This paper presents a new algorithm called the adaptive logarithmic increase and adaptive decrease algorithm (A-LIAD), which mainly addresses the Round-Trip Time (RTT) fairness problem in satellite networks with a very high propagation delay as an alternative to the current TCP congestion control algorithm. We defined a new increasing function in the fashion of a logarithm depending on the increasing factor ${\alpha}$, which is different from the other logarithmic increase algorithm adopting a fixed value of ${\alpha}$ = 2 leading to a binary increase. In A-LIAD, the ${\alpha}$ value is derived in the RTT function through the analysis. With the modification of the increasing function applied for the congestion avoidance phase, a hybrid scheme is also presented for the slow start phase. From this hybrid scheme, we can avoid an overshooting problem during a slow start phase even without a SACK option. To verify the feasibility of the algorithm for deployment in a high-speed and long-distance network, several aspects are evaluated through an NS-2 simulation. We performed simulations for intra- and interfairness as well as utilization in different conditions of varying RTT, bandwidth, and PER. From these simulations, we showed that although A-LIAD is not the best in all aspects, it provides a competitive performance in almost all aspects, especially in the start-up and packet loss impact, and thus can be an alternative TCP congestion control algorithm for high BDP networks including a satellite network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12A
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• pp.975-982
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• 2003

A connection admission control scheme is proposed that can seamlessly support QoS in face of handoff in mobile packet networks. The proposed scheme is based on the adaptive time-selective bandwidth reservation with the reduced signaling and computational overhead. The reservation parameters are adjusted dynamically to cope with user mobility. Throughout the computer simulations, the performance of the proposed scheme is evaluated. As a result, it is shown that the proposed scheme reduces handoff call blocking probability remarkably, while other parameters (new call blocking probability and bandwidth utilization efficiency) are slightly deteriorated.

• 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.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1222-1227
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• 2005

In this paper, an efficient scheme of the adaptive notch filter is presented for rejecting the narrow bandwidth interferences(NBI) in GPS receiver. Designed is the lattice IIR double notch filter for more efficient suppression of the NBI with less computational complexity. The algorithm is of recursive prediction error form and uses a special constrained model of IIR with a minimal number of parameters. This paper chooses seven different jamming scenarios including one without jamming for evaluating the proposed filter algorithm. The simulation results to the jamming scenarios show that the proposed algorithm adjusts the double notch filter effectively for the given JSR, and provides better SNR than the conventional algorithms. Finally, it is shown that the advantages of the proposed filter algorithm can range as high as JSR 79dB in time domain processing. Also, the ADNF(adaptive double notch filter) guarantees that more than SNR 10dB of GPS receiver can be always maintained. In conclusion, there is enough evidence to believe that the proposed algorithm will perform quite well for removing interference signals.

• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.783-788
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• 2002

Adaptive Cyclic Packet Dropping algorithm (ACPD), and Non-adaptive Cyclic Packet Dropping algorithm (NCPD) are applying stricter drop precedence than that of RIO algorithm. Especially, the ACPD algorithm drops adaptively packets for the congestion control, as predicting traffic pattern between each cycle. Therefore the ACPD algorithm makes up for the drawback of RIO algorithm and minimizes the wastes of the bandwidth being capable of predicting in the NCPD algorithm. And we executed a simulation and analyzed the throughput and packet drop rate based on Sending Priority changing dynamically depending on network traffic. In this algorithm, applying strict drop precedence policy, we get better performance on priority levels. The results show that the proposed algorithms may provide more efficient and stricter drop precedence policy as compared to RIO independent of traffic load. The ACPD algorithm can provide better performance on priority levels and keep stricter drop policy than RIO and the NCPD algorithm.

• The KIPS Transactions:PartA
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• v.10A no.6
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• pp.691-700
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• 2003

Due to the fast development of wire&wireless internet and computer hardware, more and more internet services are being developed, such as Internet broadcast, VoD (Video On Demand), etc. So QoS (Qualify of Service) is essentially needed to guarantee the quality of these services. Traditional Internet is Best-Effort service in which all packets are transported in FIFO (First In First Out) style. However, FIFO is not suitable to guarantee the quality of some services, so more research in QoS router and QoS protocol are needed. Researched QoS router and protocol are high cost and inefficient because the existing infra is not used. To solve this problem, a new QoS control method, named Network Adaptive QoS, is introduced and applied to client/server-based streaming systems. Based on network bandwidth monitoring mechanism, network adaptive QoS control method can be used in wire&wireless networks to support QoS in real-time streaming system. In order to reduce application cost, the existing streaming service is used in NAQoS. A new module is integrated into the existing server and client. So the router and network line are not changed. By simulation in heavy traffic network conditions, we proved that stream cannot be seamless without network adaptive QoS method.

• Journal of Electrical Engineering and Technology
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• v.12 no.4
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• pp.1627-1633
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• 2017

This paper proposes a voice enhancing ear-protection system which is based on feedback active noise control(FBANC). The proposed system selectively suppresses the background noise and preserves the talking voice by controlling the adaptive algorithm with the voice activity period detection module. The noise reduction performance of the proposed noise canceling algorithm is analytically derived for the two key performance affecting parameters, i.e., electro-acoustic coupling distance and noise bandwidth. The proposed system is also implemented with a floating-point DSP system and its performance is experimentally tested to compare with the analytically derived results. The achieved levels of noise reduction for the three different noise bandwidths cases, i.e., 10Hz, 50Hz, and 90Hz, are high to show 17.05dB, 10.54dB and 8.99dB, respectively. The feasibility of the proposed system is also shown by the peak noise reduction achieved more than 25dB while preserving the voice component in the frequency range between 200-800Hz.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2064-2067
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• 2003

본 논문에서는 비트율 제어 (rate-control)와 혼잡 제어(congestion-control) 사이의 관계를 고려하여 비디오데이터의 전송량 변화에 따른 화질 증감의 관계를 조사하고, 전송 대역폭의 변화에 대한 화질 열화를 최소화하기 위한 비디오 데이터 전송량의 변화 조건을 제안하고자 한다. 비디오 데이터를 전송하는 경우에 혼잡 제어는 현 채널 상에서 가용 전송 대역폭(Available Transmission Bandwidth)을 예측하고, 비트율 제어는 예측한 전송 대역폭을 비디오 부호화기의 전송 비트율에 적응시킨다. AIMD 기반의 혼잡 제어 기법은 비디오 데이터의 목표 비트율 (target bit rate)을 변화시키게 되고 이러한 결과는 비디오 화질의 잦은 변동으로 인해 나쁜 영향을 준다. 이러한 문제를 해결하기 위해 QP, SNR, 비트량 사이의 관계를 이용한 적응적인 메커니즘을 제안하고자 한다

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.11
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• pp.1-8
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• 2001

Adaptive and Strict Packet Dropping mechanism we proposed in this paper drops adaptively packets for the congestion control, as predicting traffic pattern between each cycle. Therefore the proposed mechanism makes up for the drawback of RIO mechanism and minimizes the wastes of the bandwidth being capable of predicting in Dynamic and Strict Packet Dropping (DSPD) mechanism. And we executed a simulation and analyzed the throughput and packet drop rate based on the Sending Drop Precedence changing dynamically depending on the network traffic and compared RIO and the DSPD. The results show that the proposed mechanism provides better performance on drop precedence levels and stricter drop precedence policy for AF class than RIO and the DSPD mechanism.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.8
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• pp.1805-1824
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• 2013

Recently, the use of wireless body area networks (WBAN) has been increasing rapidly in medical healthcare applications. WBANs consist of smart nodes that can be used to sense and transmit vital data such as heart rate, temperature and ECG from a human body to a medical centre. WBANs depend on limited resources such as energy and bandwidth. In order to utilise these resources efficiently, a very well organized medium access control (MAC) protocol must be considered. In this paper, a new, adaptive and energy-efficient MAC protocol, entitled isMAC, is proposed for WBANs. The proposed MAC is based on multi-channel communication and aims to prolong the network lifetime by effectively employing (i) a collision prevention mechanism, (ii) a coordinator node (WCN) selection algorithm and (iii) a transmission power adjustment approach. The isMAC protocol has been developed and modelled, by using OPNET Modeler simulation software. It is based on a networking scenario that requires especially high data rates such as ECG, for performance evaluation purposes. Packet delay, network throughput and energy consumption have been chosen as performance metrics. The comparison between the simulation results of isMAC and classical IEEE 802.15.4 (ZigBee) protocol shows that isMAC significantly outperforms IEEE 802.15.4 in terms of packet delay, throughput and energy consumption.