• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1045-1054
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• 2020

In this paper, we designed a TCP/IP based ring buffer system that can stably transfer bulk data streams in the unstable network environments. In the scheme we proposed, The observation data stream generated and output by each radio observatory's backend system as a UDP frame is stored as a UDP packet in a large capacity ring buffer via a socket buffer in the client system. Thereafter, for stable transmission to the remote destination, the packets are processed in TCP and transmitted to the socket buffer of server system in the correlation center, which packets are stored in a large capacity ring buffer if there is no problem with the packets. In case of errors such as loss, duplication, and out of order delivery, the packets are retransmitted through TCP flow control, and we guaranteed that the reliability of data arriving at the correlation center. When congestion avoidance occurs due to network performance instability, we also suggest that performance degradation can be minimized by applying parallel streams.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.4
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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.

• Journal of Korean Society of Transportation
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• v.29 no.5
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• pp.121-138
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• 2011

More affordable and available cutting-edge technologies (e.g., wireless vehicle communication) are regarded as a possible alternative to the fixed infrastructure-based traffic information system requiring the expensive infrastructure investments and mostly implemented in the uninterrupted freeway network with limited spatial system expansion. This paper develops an advanced decentralized traveler information System (ATIS) using vehicle-to-vehicle (V2V) communication system whose performance (drivers' travel time savings) are enhanced by three complementary functions (autonomous automatic incident detection algorithm, reliable sample size function, and driver behavior model) and evaluates it in the typical $6{\times}6$ urban grid network with non-recurrent traffic state (traffic incident) with the varying key parameters (traffic flow, communication radio range, and penetration ratio), employing the off-the-shelf microscopic simulation model (VISSIM) under the ideal vehicle communication environment. Simulation outputs indicate that as the three key parameters are increased more participating vehicles are involved for traffic data propagation in the less communication groups at the faster data dissemination speed. Also, participating vehicles saved their travel time by dynamically updating the up-to-date traffic states and searching for the new route. Focusing on the travel time difference of (instant) re-routing vehicles, lower traffic flow cases saved more time than higher traffic flow ones. This is because a relatively small number of vehicles in 300vph case re-route during the most system-efficient time period (the early time of the traffic incident) but more vehicles in 514vph case re-route during less system-efficient time period, even after the incident is resolved. Also, normally re-routings on the network-entering links saved more travel time than any other places inside the network except the case where the direct effect of traffic incident triggers vehicle re-routings during the effective incident time period and the location and direction of the incident link determines the spatial distribution of re-routing vehicles.

• Journal of KIISE:Information Networking
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• v.35 no.2
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• pp.99-111
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• 2008

In recent years, the needs for WLANs(Wireless Local Area Networks) technology which can access to Internet anywhere have been dramatically increased particularly in SOHO(Small Office Home Office) and Hot Spot. However, unlike wired networks, there are some unique characteristics of wireless networks. These characteristics include the burst packet losses due to unreliable wireless channel. Note that burst packet losses, which occur when the distance between the wireless station and the AP(Access Point) increase or when obstacles move temporarily between the station and AP, are very frequent in 802.11 networks. Conversely, due to burst packet losses, the performance of 802.11 networks are not always as sufficient as the current application require, particularly when they use TCP at the transport layer. The high packet loss rate over wireless links can trigger unnecessary execution of TCP congestion control algorithm, resulting in performance degradation. In order to overcome the limitations of WLANs environment, MAC-layer LDA(Loss Differentiation Algorithm)has been proposed. MAC-layer LDA prevents TCP's timeout by increasing CRD(Consecutive Retry Duration) higher than burst packet loss duration. However, in the wireless channel with high packet loss rate, MAC-layer LDA does not work well because of two reason: (a) If the CRD is lower than burst packet loss duration due to the limited increase of retry limit, end-to-end performance is degraded. (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily by Reducing the drainage speed of the network buffer due to the increase of CRD. In this paper, we propose a new retransmission module based on Cross-layer approach, called BLD(Burst Loss Detection) module, to solve the limitation of previous link layer retransmission schemes. BLD module's algorithm is retransmission mechanism at IEEE 802.11 networks and performs retransmission based on the interaction between retransmission mechanisms of the MAC layer and TCP. From the simulation by using ns-2(Network Simulator), we could see more improved TCP throughput and energy efficiency with the proposed scheme than previous mechanisms.