• ETRI Journal
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• 제44권6호
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• pp.945-954
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• 2022

The 5G mobile network is promising to handle the dynamic traffic demands of user equipment (UE). Unmanned aerial vehicles (UAVs) equipped with wireless transceivers can act as flying base stations in heterogeneous networks to ensure the quality of service of UE. However, it is challenging to efficiently allocate limited bandwidth to UE due to dynamic traffic demands and low network coverage. In this study, a blockchain-enabled bandwidth allocation framework is proposed for secure bandwidth trading. Furthermore, the proposed framework is based on the Cournot oligopoly game theoretical model to provide the optimal solution; that is, bandwidth is allocated to different UE based on the available bandwidth at UAV-assisted-based stations (UBSs) with optimal profit. The Cournot oligopoly game is performed between UBSs and cellular base stations (CBSs). Utility functions for both UBSs and CBSs are introduced on the basis of the available bandwidth, total demand of CSBs, and cost of providing cellular services. The proposed framework prevents security attacks and maximizes the utility functions of UBSs and CBSs.

• 조명전기설비학회논문지
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• 제24권7호
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• pp.26-40
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• 2010

We propose a new routing protocol, MCQosR, that is based on bandwidth estimation, admission control, and a routing metric, MCCR - suitable for wireless ad-hoc networks with multiple radios and channels. To use the full capacity of a wireless link, we assume a node with multiple radios for full duplex operation, and a radio using multiple channels to exclude route-intra interference. This makes it possible to use the capacity of a wireless link. Then, to provide bandwidth and delay guarantee, we have a radio with a fixed channel for layer-3 data reception at each node, used to estimate the available bandwidth and expected delay of a wireless link. Based on the estimate of available bandwidth and delay, we apply the call admission control to a new call requiring bandwidth and delay guarantee. New calls with traffic that will overflow link or network capacity are rejected so the accepted calls can use the required bandwidth and delay. Finally, we propose a routing metric, MCCR, which considers the channel contentions and collisions of a wireless link operating in CSMA/CA. MCCR is useful for finding a route with less traffic and distributing traffic over the network to prevent network congestion as much as possible. The simulation of the MCQosR protocol and the MCCR metric shows traffic is distributed and guaranteed service is provided for accepted calls.

• 디지털콘텐츠학회 논문지
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• 제15권2호
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• pp.147-154
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• 2014

높은 오류율을 가지는 무선 TCP(Transmission Control Protocol) 환경을 위한 TCP-Westwood 방법에 대한 연구가 많이 진행되었다. 이러한 TCP-Westwood 방법은 가용대역폭측정(Available Bandwidth Estimation)에 의해서 혼잡윈도우를 조절하는 방법이기 때문에 오류율이 높아질수록 이 가용대역폭측정값의 정확도가 떨어지는 문제점이 있다. 이러한 문제점을 해결하기 위해 본 논문에서는 ACK의 도착간격이 주어진 시간 간격을 벗어나는 경우에 ABE값을 무시하는 기존의 패킷 패턴 기반 가용대역폭필터링 방법과 가용대역폭측정값이 허용 가능한 범위를 벗어나는 경우에 주어진 허용 가능한 가용대역폭측정값으로 할당하는 새로운 방법을 모두 사용하였다. 제안한 방법의 실험결과는 기존의 다른 알고리즘들보다 가용대역폭측정값이 설정한 시뮬레이션 대역폭에 가장 유사함을 확인하였다.

• 대한전자공학회논문지TC
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• 제41권4호
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• pp.35-44
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• 2004

최근에 제안된 TCP Vegas는 네트워크의 상황을 예측하고, 상황에 맞는 전송율의 변화를 통해 흐름제어와 혼잡제어를 함으로써 기존의 TCP Reno에 비해 많은 성능향상을 이루었다. 그러나 TCP Vegas는 네트워크에 대한 모델이 없기 때문에 네트워크의 가용대역폭을 충분히 활용할 정확한 전송율을 계산하지 못하고, 제한된 윈도우의 변화만을 적용시킴으로 인해, 급변하는 가용 대역폭의 변화에 적응을 하지 못하고 손실이 발생할 가능성이 여전히 존재한다. 본 논문에서는 이러한 단점을 극복하고자 TCP의 단대단 연결을 큐잉 시스템으로 모델링하여 적절한 전송율을 결정함으로써, 안정되고 빠르게 가용 대역폭에 수렴하는 알고리듬을 제안한다. 시뮬레이션을 통한 TCP Vegas와의 성능 비교 결과는 제안된 알고리듬이 가용 대역폭이 급변하는 네트워크 환경에서 TCP Vegas 보다 안정적이고 빠르게 반응하기 때문에 외부 트래픽의 변화에 더 잘 적응하고 처리율을 향상시키는 결과를 보여준다.

• 산업경영시스템학회지
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• 제20권41호
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• pp.51-60
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• 1997

In this paper, A Connection Admission Control(CAC) algorithm by Dynamic Bandwidth Allocation is proposed. The B-ISDN will be required to carry traffic generated by a various traffic sources. These source will have all traffic flow characteristics and quality of services requirements. When a connectionless message with burstiness arrives at a node during a renewal period, CAC will be dynamically determined connection request by using the predetermined connection admission bounds in estimating the number of input arriving cells. The proposed algorithm estimates the available bandwidth by observing cell streams of the aggregate traffic. A connection request is accepted if the available bandwidth is larger than declared peak rate, and otherwise rejected. The proposed CAC scheme is shown to be promising and practically feasible in obtaining efficient transmission of burst traffic with guaranteed quality of services.

• 한국멀티미디어학회논문지
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• 제7권8호
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• pp.1078-1085
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• 2004

SARLM(Source Adaptive Receiver-Driven Multi-Layered Multicast)은 레이어 병합과 분할을 통해 수신자의 이질적인 대역폭 문제를 개선하고자 하였다. 그러나 이 기법은 레이어 병합 시에 상위 레이어의 수신자가 하위 레이어로 드롭되어 가용 대역폭의 이용률이 저하되는 문제가 있다. 본 논문에서는 수신자의 손실률 정도를 고려한 멀티캐스트 재그룹 기법을 제안한다. 제안 기법은 손실률에 따라 가중치를 부여하여, 상위 레이어와 하위 레이어 중 여유 대역폭이 많은 수신자를 그룹화하고, 하위 레이어와 상위 레이어 중 손실률이 높은 수신자를 그룹화한다. 실험 결과, 상위 레이어 수신자가 높은 전송률을 유지함으로써 수신자의 가용 대역폭 이용률이 향상됨을 알 수 있다.

• Communications for Statistical Applications and Methods
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• 제31권3호
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• pp.337-347
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• 2024

Kernel density estimation is a prevalent technique employed for nonparametric density estimation, enabling direct estimation from the data itself. This estimation involves two crucial elements: selection of the kernel function and the determination of the appropriate bandwidth. The selection of the bandwidth plays an important role in kernel density estimation, which has been developed over the past decade. A range of methods is available for selecting the bandwidth, including the plug-in bandwidth. In this article, the proposed plug-in bandwidth is introduced, which leverages shifted Chebyshev series-based approximation to determine the optimal bandwidth. Through a simulation study, the performance of the suggested bandwidth is analyzed to reveal its favorable performance across a wide range of distributions and sample sizes compared to alternative bandwidths. The proposed bandwidth is also applied for kernel density estimation on real dataset. The outcomes obtained from the proposed bandwidth indicate a favorable selection. Hence, this article serves as motivation to explore additional plug-in bandwidths that rely on function approximations utilizing alternative series expansions.

• 한국통신학회논문지
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• 제22권4호
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• pp.618-636
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• 1997

DQDB(Distributed Queue Dual Bus) protocol, the IEEE 802.6 standard protocol for metropolitan area networks, does not fully take advantage of the capabilities of dual bus architecture. Although fairness in bandwidth distribution among nodes is improved when using so called the bandwidth balancing mechanism, the protocol requires a considerable amount of time to adjust to changes in the network load. Additionally, the bandwidth balancing mechanism leaves a portion of the available bandwidth unused. In a high-speed backbone network, each node may act as a bridge/ router which connects several LANs as well as hosts. However, Because the existence of high speed LANs becomes commonplace, the congestionmay occur on a node because of the limitation on access rate to the backbone network and on available buffer spaces. to release the congestion, it is desirable to install some congestion control algorithm in the node. In this paper, we propose an efficient congestion control mechanism and fair and waster-free MAC protocol for dual bus network. In this protocol, all the buffers in the network can be shared in such a way that the transmission rate of each node can be set proportional to its load. In other words, a heavily loaded node obtains a larger bandwidth to send the sements so tht the congestion can be avoided while the uncongested nodes slow down their transmission rate and store the incoming segments into thier buffers. this implies that the buffers on the network can be shared dynamically. Simulation results show that the proposed probotol significantly reduces the segment queueing delay of a heavily loaded node and segment loss rate when compared with original DQDB. And it enables an attractive high throughput in the backbone network. Because in the proposed protocol, each node does not send a requet by the segment but send a request one time in the meaning of having segments, the frequency of sending requests is very low in the proposed protocol. so the proposed protocol signigificantly reduces the segment queuing dely. and In the proposed protocol, each node uses bandwidth in proportion to its load. so In case of limitation of available buffer spaces, the proposed protocol reduces segment loss rate of a heavily loaded node. Bandwidth balancing DQDB requires the wastage of bandwidth to be fair bandwidth allocation. But the proposed DQDB MAC protocol enables fair bandwidth without wasting bandwidth by using bandwidth one after another among active nodes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제10권10호
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• pp.4787-4807
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• 2016

Although the dense interconnection datacenter networks (DCNs) (e.g. FatTree) provide multiple paths and high bisection bandwidth for each server pair, the single-path TCP (SPT) and ECMP which are widely used currently neither achieve high bandwidth utilization nor have good load balancing. Due to only one available transmission path, SPT cannot make full use of all available bandwidth, while ECMP's random hashing results in many collisions. In this paper, we present OFPT, an OpenFlow based Parallel Transport framework, which integrates precise routing and scheduling for better load balancing and higher network throughput. By adopting OpenFlow based centralized control mechanism, OFPT computes the optimal path and bandwidth provision for each flow according to the global network view. To guarantee high throughput, OFPT dynamically schedules flows with Seamless Flow Migration Mechanism (SFMM), which can avoid packet loss in flow rerouting. Finally, we test OFPT on Mininet and implement it in a real testbed. The experimental results show that the average network throughput in OFPT is up to 97.5% of bisection bandwidth, which is higher than ECMP by 36%. Besides, OFPT decreases the average flow completion time (AFCT) and achieves better scalability.

• 대한전자공학회논문지TC
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• 제43권10호
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• pp.8-17
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• 2006

현재 초고속 인터넷 사용자가 급증하고 있고, 광대역 네트워크 인프라의 구축이 늘어나고 있다. 하지만, 지금 사용하고 있는 TCP 혼잡 제어 알고리즘은 협대역 네트워크 환경에 적합하여, 광대역 네트워크의 트래픽 전송에 있어서 효율성이 낯은 상태이다. 이런 문제점을 개선하기 위해 광대역 네트워크에 적합하도록 개선된 혼잡 제어 알고리즘을 적용한 TCP가 요구되고 있다. 본 논문에서는 광대역 네트워크를 충분히 활용할 수 있도록 개선된 TCP 혼잡 제어 알고리즘을 제안한다. 제안된 알고리즘은 수신측에서 송신측에 보내는 ACK 정보 및 RTT 변화와 특성을 이용하여 가용한 대역폭을 예측함으로써 가용한 혼잡 윈도우 크기를 조정하고 대량의 패킷 손실을 최소화한다. 또한, 혼잡 회피 단계에서의 기존 TCP 알고리즘을 개선함으로써 가용한 대역폭을 빠르게 활용할 수 있도록 한다. 본 논문에서는 제안된 알고리즘 성능을 평가하기 위하여 ns-2를 이용하여 시뮬레이션을 수행하였으며, 시뮬레이션 결과 광대역 네트워크 환경에서 제안된 알고리즘이 기존 $HSTCP^{[2]}$보다 전송률이 향상되어 가용 대역폭 활용률을 높였을 뿐만 아니라 공정성과 RTT 공정성도 개선하였음을 보였다.