• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10B
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• pp.613-620
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• 2007

Recently, due to the developing of broadband convergence networks, the high quality multimedia services which are converged with communication and broadcasting can be provided. These high quality communication and broadcasting convergence service needs reliable multicasting mechanism in IP networks. However, multicasting mechanism recently deployed is difficult to guarantee the end-to-end QoS. For the reliable high quality If multimedia IPTV service, the QoS guaranteed multicasting mechanism should be provided. Therefore, we proposed a probing packet based multicasting admission control mechanism using centralized multicasting control management block in this paper. And for the performance evaluation of the proposed mechanism, we used ns-2 simulator.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1683-1690
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• 2005

This study examines dynamic flow control method in UDP, analyzes packet loss which is frequently used element in measuring existing dynamic flow control and characteristics of round trip time, and proposes a new method of measurement, RSST. The algorithm that uses the proposed RSST enables accurate measurement of network status by considering both the currently measured network status and the past history of network status in controlling the transmission rate. For comparison study, a network status measurement software program that displays detailed information about volume of transmission generation of network status, and flow pattern of network was developed. The performance test shows that the proposed algorithm can better adjust to network condition in terms of low pack loss rate over existing algorithms.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.6
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• pp.1402-1409
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• 2007

In this paper, we propose various types of AGC algorithms for implementing the OFDM communication systems. For the high-speed packet transmission, in this paper, we assume the OFDM system with relatively long and repeated preambles. We propose the maximum sample value counter for counting the number of maximum sample. In the maximum sample value counter, we use the buffer for the digital signal buffering. Finally, the counting value of the maximum sample value counter controls the gain control signal generator by using gain control table automatically.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.3
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• pp.117-124
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• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.284-287
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• 2000

Recently, developing medical devices have a tendency becoming the module for satisfying user's mutual complex needs. Because the most effective method for the observation of patients condition a diagnosis and a treatment is collecting data from various devices and controling operation following it. Module tendency is more popular due to manage easily totally many individual systems. This study implemented communication protocol to control by one control system connecting modular medical devices. Implemented system consist of one master module controlling all module and managing communication and many Slave modules. Communication between each modules introduced SPI(Serial Peripheral Interface) among many synchronous serial communication methods for the exact transmission and receipt of data. All communication executes by packet format. This can detect error. And, this protocol introduced PNP(Plug And Play) function that auto-detect connecting or removing module during running. This protocol exactly transmitted and received in faster speed more than 1Mbps. And in practical application to the ventilator this confirmed to give and take real-time data. And various functions by th central control system is implemented in this protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.595-604
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• 2009

The HCCA reserves the channel resources based on the mean data rate in IEEE 802.11e. It may cause either the waste of channel resource or the increase of transmission delay at MAC layer if the frame size is rapidly varied when a compressed mode video codec such as MPEG video is used. To solve these problems, it is developed that the packet scheduler allocates the wireless resource adaptation by according to the packet size. However, it is difficult to perform the admission control because of the difficulty with calculating the available resources. In this paper, we propose a CAC mechanism to solve the problem that may not satisfy the QoS by increasing traffic load in case of using EDCA. Especially, the proposed CAC mechanism calculates the EB of TSs using the traffic information transmitted by the application layer and the number of average transmission according to the wireless channel environment, and then determines the admission of the TS based on the EB. According to the simulation results of the proposed CAC mechanism, it admitted the TSs under the loads which are satisfied within the delay bound. Therefore, the proposed mechanism guarantees QoS of streaming services effectively.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.891-898
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• 2003

This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.

• The Journal of the Korea Contents Association
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• v.11 no.2
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• pp.69-78
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• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.11
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• pp.5338-5359
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• 2017

In-band full-duplex (IBFD) wireless communication supports symmetric dual transmission between two nodes and asymmetric dual transmission among three nodes, which allows improved throughput for distributed IBFD wireless networks. However, inter-node interference (INI) can affect desired packet reception in the downlink of three-node topology. The current Half-duplex (HD) medium access control (MAC) mechanism RTS/CTS is unable to establish an asymmetric dual link and consequently to suppress INI. In this paper, we propose a medium access control mechanism for use in distributed IBFD wireless networks, FD-DMAC (Full-Duplex Distributed MAC). In this approach, communication nodes only require single channel access to establish symmetric or asymmetric dual link, and we fully consider the two transmission modes of asymmetric dual link. Through FD-DMAC medium access, the neighbors of communication nodes can clearly know network transmission status, which will provide other opportunities of asymmetric IBFD dual communication and solve hidden node problem. Additionally, we leverage FD-DMAC to transmit received power information. This approach can assist communication nodes to adjust transmit powers and suppress INI. Finally, we give a theoretical analysis of network performance using a discrete-time Markov model. The numerical results show that FD-DMAC achieves a significant improvement over RTS/CTS in terms of throughput and delay.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.51-54
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• 2008

PON(Passive Optical Network) is the promising technology applicable to TPS(Triple Play Service). To improve process the packet, MAC(Multiple Access Control) is the most important technology in the PON. The core of MAC is DBA(Dynamic Bandwidth Assignment), it classifies SR-DBA(Status Report DBA) and NSR-DBA(Non Status Report DBA). But GPON DBA is using BPON's DBA, so it's bad in network efficiency. This study develop BR(Borrow-Refund)-DBA for improve network efficiency and prompt process. For take the gauge of performance evaluation, estimate about throughput, fairness and queue delay in SR-DBA and NSR-DBA environment