• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.3
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• pp.537-548
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• 2000

ATM is the effective information transmission method which multiply statistical and can accept effective the traffic of the various transmission rate. However, it can happen excessive cell loss probability and cell delay when it have temporarily overload. Therefore, it is required the effective traffic control and network resource management for which guarantee QoS(Quality of Service) in terms of users and bandwidth utilization maximization in terms of networks. In this paper, we proposed POBP(PushOut BP) scheme which mixed pushout scheme with BP(Back Pressure), reactive control scheme recommended at the ATM Forum, to guarantee QoS in two stages ATM switch networks. In proposed method, we have to understand exactly using bandwidth information in real-time and become traffic shaping. Thorough these processes, we can not only use effectively unused bandwidth, but also guarantee the fair bandwidth utilization and then can improve cell-loss possibility happened by congestion states in two stage ATM networks.

• Asia-pacific Journal of Multimedia Services Convergent with Art, Humanities, and Sociology
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• v.7 no.1
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• pp.287-295
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• 2017

Recently, MTC (Machine Type Communication) is known as an important part to support IoT (Internet of Things) applications. MTC provides network connectivities between MTC devices without human intervention. In MTC, a large number of devices try to access over communication resource with a short period of time. Due to the limited communication resource, resource contention becomes severe and it brings about access failures of devices. To solve the problem, it needs to regulate device accesses. In this paper, we present an efficient access management scheme. We measure the number of devices which try to access in a certain time period and predict the change of the number of devices in the next time period. Using the predicted change, we control the number of devices which try to access. To verify our scheme, we conduct experiments in terms of success probability, failure probability, collision probability and access delay.

• 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.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.19 no.3
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• pp.38-51
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• 2020

A bottleneck and congestion occur when a freeway is closed due to maintenance and construction activities on the freeway. Although various traffic managements have been developed to improve the traffic efficiency at freeway work zones, such as merge control, there is a limit to those controls with human drivers. On the other hand, the wireless communication of connected and automated vehicles (CAVs) enables the operation of advanced traffic management. This study developed a traffic control strategy that integrates Dynamic Merge Control (DMC) and Lane Change Control (LCC) in a CAV environment. DMC operates as an either early or late merge based on the occupancy rate of upstream of the work zone. The LCC algorithm determines the number of vehicles that need to change their lane to balance the traffic volume on open lanes. The simulation results showed that integrated control improves the cumulative vehicle count, average speed upstream, and average network travel time.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.409-414
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• 2008

In mobile ad hoc networks, most of on demand routing protocols such as DSR and AODV do not deal with traffic load during the route discovery procedure. To solve the congestion and achieve load balancing, many protocols have been proposed. However, the existing load balancing schemes has only considered avoiding the congested route in the route discovery procedure or finding an alternative route path during a communication session. To mitigate this problem, we have proposed a new scheme which considers the packet bursting mechanism in congested nodes. The proposed packet bursting scheme, which is originally introduced in IEEE 802.11e QoS specification, is to transmit multiple packets right after channel acquisition. Thus, congested nodes can forward buffered packets promptly and minimize bottleneck situation. Each node begins to transmit packets in normal mode whenever its congested status is dissolved. We also propose two threshold values to define exact overloaded status adaptively; one is interface queue length and the other is buffer occupancy time. Through an experimental simulation study, we have compared and contrasted our protocol with normal on demand routing protocols and showed that the proposed scheme is more efficient and effective especially when network traffic is heavily loaded.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.

• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.233-243
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• 2003

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In this paper, we propose three schemes that improve the ability to conceal the packet losses in the wireless network while limiting the degree of violating TCP end-to-end semantics to a temporary incidents. If there happens a packet loss at the wireless link and there is a chance that the loss is noticed by the sending TCP, the proposed schemes send an indirect acknowledgement. Each of the proposed schemes uses different criteria to decide whether there is a chance that the packet loss occurred in the wireless part is noticed by the sender. In order to limit the buffer overhead in the base, the indirect acknowledgements are issued only when the length of buffer is less than a certain threshold. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compared to Snoop with a limited amount of buffer at the base station.

• Journal of Korea Multimedia Society
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• v.7 no.4
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• pp.518-531
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• 2004

This paper discusses a testbed architecture for implementing scalable service differentiation in the Internet. The differentiated services (DiffServ) testbed architecture is based on a model in which a bandwidth broker (BB) can control network resources, and the ALTQ can reserve resources in a router to guarantee a Quality of Service (QoS) for incoming traffic to the testbed. The reservation and releasemessage for the ALTQ is contingent upon a decision message in the BE. The BB has all the information in advance, which is required for a decision message, in the form of PIB. A signaling protocol between the BB and the routers is the COPS protocol proposed at the IETF. In terms of service differentiation, a user should make an SLA in advance, and reserve required bandwidth through an RAR procedure. The SLA and RAR message between a user and the BB has implemented with the COPS extension which was used between a router and the BB. We evaluates the service differentiation for the video streaming in that the EF class traffic shows superb performance than the BE class traffic where is a network congestion. We also present the differentiated service showing a better packet receiving rate, low packet loss, and low delay for the EF class video service.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.9
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• pp.73-80
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• 2004

In this paper, we present the cell resequencing buffer to solve the cell sequence integrity problem of the Cyclic banyan network that is a high-performance fault-tolerant cell switch. By offering multiple paths between input ports and output ports, using the deflection self-routing, the Cyclic banyan switch offer high reliability, and it also solves congestion problem for the internal links of the switch. By the way, these multiple paths can be different lengths for each other. Therefore, the cells departing from an identical source port and arriving at an identical destination port can reach to the output port as the order that is different from the order arriving at input port. The proposed cell resequencing buffer is a hardware sliding window mechanism. to solve such cell sequence integrity problem. To calculate the size of sliding window that cause the prime cost of the presented device, we analyzed the distribution of the cell delay through the simulation analyses under traffic load that have a nonuniform address distribution that express tile Property of traffic of the Internet. Through these analyses, we found out that we can make a cell resequencing buffer by which the cell sequence integrity is to be secured, by using a, few of ordinary memory and control logic. The cell resequencing buffer presented in this paper can be used for other multiple paths switching networks.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.3
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• pp.259-269
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• 2018

Multipath TCP (MPTCP) can be expected to provide improved network performance because it transmits data through multiple paths. However, Bufferbloat, which unexpectedly occurs in the transmission path, degrades not only the performance of the corresponding path but also the performance of other paths, so that the performance is worse than that of a single TCP. In this paper, we propose the transmission status monitoring method at the sender's MPTCP level and also HoL packets retransmission algorithm in order to solve the Bufferbloat problem. The proposed algorithm enables Bufferbloat detection by the sender side independently, and it can resolve the HoL blocking problem by identifying the HoL packet in the proposed transmission status monitoring buffer and retransmitting it to the normal path. Simulation results based on NS-3 show that the proposed algorithm achieves the improved throughput performance up to 22.8% compared to the existing MPTCP, and the average number of queued packets in the sender and receiver's buffers is decreased to 44.3% and 9.2%, respectively.