• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.5
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• pp.939-948
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• 1997

In this paper, we propose the new rate-based transmission rates control algorithm that allocates the fair band-width for ABR service in ATM network. In the traditional ABR service, bandwidth is allocated with constant rate increment or decrement, but in the proposed algorithm, it is allocated fairly to the connected calls by the fuzzy inference of the available bandwidth. The fuzzy inference uses buffer state and the buffer variant rate as the input variables, and uses the total transmission rate as a output variable. This inference a bandwidth is fairly distributed over all ABR calls in service. By simmulation, we showed that the proposed method improved 0.17% in link effectiveness when RIF, RDF is 1/4, 38.6% when RIF, RDF 1/16, and 82.4% when RIF, RDF 1/32 than that of the traditional EFPCA.

• Journal of Korea Multimedia Society
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• v.8 no.8
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• pp.1099-1107
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• 2005

The Guaranteed Frame Rate(GFR) service has been designed to accomodate non-real-time applications, such as TCP/IP based traffic in ATM networks. The GFR service not only guarantees a minimum throughput at the frame level, but also supports a fair share of available resources. In this paper, we propose a cell scheduling scheme which can improve the fairness and the goodput through the traffic control in GFR service. For the evaluation of the proposed scheme, we compare the proposed scheme with the existing scheme in the fairness and the goodput. Simulation results show that proposed scheme can improve the fairness and goodput comparing with the existing buffer management scheme.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6A
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• pp.617-626
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• 2006

In this paper, we propose an ASA(Aggressive Subchannel Allocation) algorithm, which is an effective dynamic channel allocation algorithm considering all user's channel state to maximize downlink sector throughput in OFDMA system. We compare an ASA algorithm with Round Robin, ACG(Amplitude Craving Greedy), RCG(Rate Craving Greedy) and GPF(General Proportional Fair) in the 2-tier environment of FRF(Frequency Reuse Factor) 1 and then analyze the performance of each algorithms, through compute simulation. Simulation results show that the proposed ASA algorithm gets 58 %, 190 %, 130 % and 8.5 % better sector throughput compared with the Round Robin, ACG, RCG and GPF respectively.

• Journal of KIISE:Information Networking
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• v.32 no.6
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• pp.732-744
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• 2005

Previous works for the AE(Assured Forwarding) service in the Diffserv network have no sufficient consideration on the fairness of bandwidth share based on the target rate and the effect or RTT and UDP. Also Previous works act like Best-effort service in the UPN(under-Provisioned Network) condition. In this paper, in order to solve these problems, we propose the PFDSA(Proportionally Fair Differentiated Service Architecture) composed of tmTRA3CM(tcp-microflow based Target rate and an Aware Three color Marking), um3CM(udp-microflow based Three color Marker), TRBD(Target Rate Based Dropper), and target rate adjusting function. In the results of comparing the performance among existing mechanisms and the PFDSA, the PFDSA was able to mitigate the RTT and UDP effect better than the former. The PFDSA was shown to provide good performance for transmission rates proportional to various target rates in the UPN condition.

• The Journal of the Korea Contents Association
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• v.8 no.8
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• pp.57-64
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• 2008

In this paper, to resolve the problem caused by a network state information inaccuracy the slow delay time that conclusion of network state of one way delay time which accuracy delay time information, according to network state changes on the based TFRC flow control, and suggest that flow control mechanism to adjust transfer rate fit of real time multimedia data. In simulation, to measure of netowork state information that on the average about 12% difference of compared RTT and $OWD{\times}2$. When used RTT, used fair bandwidth TFRC much better than TCP about 32%, when used OWD, difference about 3% used fair bandwidth. Thus, conclusion of accuracy network state that used fair bandwidth according to network state changes on the based TFRC, users can support service of high quality that flow control mechanism to adjust transfer rate fit of real time data.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.9
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• pp.19-28
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• 2010

The GTS(Guaranteed Time Slot) of the IEEE 802.15.4 standard, which is the contention free access mechanism, is used for low-latency applications or applications requiring specific data bandwidth. But it has some problems such as delay of service due to FIFS(First In First Service) scheduling. In this paper, we proposes a weight based GTS allocation scheme for fair queuing in IEEE 802.15.4 LR-WPAN. The proposed scheme uses a weight that formed by how much more weight we give to the recent history than to the older history for a new GTS allocation. This scheme reduces service delay time and also guarantees transmission simultaneously within a limited time. The results of the performance analysis shows that our approach improves the performance as compared to the native explicit allocation mechanism defined in the IEEE 802.15.4 standard.

• Korean Journal of Digital Imaging in Medicine
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• v.9 no.2
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• pp.39-43
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• 2007

In angiography, the global standard agreements of DICOM is lossless. But it brings on overload and takes too much store space in DICOM sever. Because of all those things we transmit images which is classified in subjective way. But this cause data loss and would be lead doctors to make wrong reading. As a result of that we try to transmit continued image (raw data) to reduce those mistakes. We got angiography images from the equipment(Allura FD20-Philips). And compressed it in two different methods(lossless & lossy fair). and then transmitted them to PACS system. We compared the quality of QC phantom images that are compressed by different compress method and compared spatial resolution of each images after CD copy. Then compared each Image's data volume(lossless & lossy fair). We measured spatial resolution of each image. All of them had indicated 401p/mm. We measured spatial resolution of each image after CD copy. We got also same conclusion (401p/mm). The volume of continued image (raw data) was 127.8MB(360.5 sheets on average) compressed in lossless and 29.5MB(360.5 sheets) compressed in lossy fair. In case of classified image, it was 47.35MB(133.7 sheets) in lossless and 4.5MB(133.7 sheets) in lossy fair. In case of angiography the diagnosis is based on continued image(raw data). But we transmit classified image. Because transmitting continued image causes some problems in PACS system especially transmission and store field. We transmit classified image compressed in lossless But it is subjective and would be different depend on radiologist. therefore it would make doctors do wrong reading when patients transfer another hospital. So we suggest that transmit continued image(raw data) compressed in lossy fair. It reduces about 60% of data volume compared with classified image. And the image quality is same after CD copy.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.1
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• pp.139-150
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• 1997

For high-speed networks, a new ring protocal is proposed in this paper. A ring network combined with destination removal can achieve much higher network throughput than the channel transmission rate. However, such a network exhibits fairness problems. Over a past few years, global fairness algorithms such as ATMR and Metaring have been proposed to solve such problems. But the ring access time delay and fairness in such networks are sensitive to the network parameters such as network size and traffic distribution. In addition to guaranteeing fair ring access to all nodes, there are several other important performance aspects in such networks. The one is that fairness is enforced while node throughputs are kept as high as possible. And another performance measure is access delay and more specifically Head-Of-Line(HOL) delay, i.e., the amount of time the first cell in the transmission buffer of a node has to wait before it accesses the ring. HOL delay is a mijor component in the transmission jitter of the synchronous traffic transmission. A key idea of the proposed ring protocol is to find the nodes that have much more chances to access the ring than any other nodes in the independently distributed node architecture. Since destined by many cells need to share a part of the bandwidths with the next node for the fairness in as much as performance degradation does not become critical. To investigate the performance behavior of the proposed ring protocol for various network condition,s several performance parameters wuch as ring access time delay, and throughput are compared with those of the ATMR and Metaring protocols using simulation package, SIMAN.