• Journal of Korea Multimedia Society
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• v.20 no.12
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• pp.1849-1855
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• 2017

The main purpose of cooperative communications is improvement of communication quality and efficient use of transmission power. In this paper, a cooperative retransmission method is proposed, where neighbor nodes that receive messages correctly between transmit and receive nodes will participate in retransmission of the erroneous packet of the direct link. When the proposed retransmission method is used, the performance of the PER can be greatly improved. In case that the limited information is included in the NACK message, the transmit power of the retransmission packet can be reduced using the proper power control method.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.385-393
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• 2005

In this paper, we propose a simple algorithm to adaptively adjust the value of dupthresh, the duplicate acknowledgement threshold that triggers the transmission control protocol (TCP) fast retransmission algorithm, to improve the TCP performance in a network environment with persistent packet reordering. Our algorithm uses an exponentially weighted moving average (EWMA) and the mean deviation of the lengths of the reordering events reported by a TCP receiver with the duplicate selective acknowledgement (DSACK) extension to estimate the value of dupthresh. We also apply an adaptive upper bound on dupthresh to avoid the retransmission timeout events. In addition, our algorithm includes a mechanism to exponentially reduce dupthresh when the retransmission timer expires. With these mechanisms, our algorithm is capable of converging to and staying at a near-optimal interval of dupthresh. The simulation results show that our algorithm improves the protocol performance significantly with minimal overheads, achieving a greater throughput and fewer false fast retransmissions.

• Proceedings of the Korea Contents Association Conference
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• 2007.11a
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• pp.60-64
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• 2007

Increasing using multimedia services as VoIP, Video conference, DMB, IPTV, etc, it is necessary to increase network traffics and develop the mechanism about a flow control for real time traffic transmission. In order to transfer realtime multimedia data, the transfer rate can be control on network state data measuring packet losses of a receiver and delay time of packets through getting periodical feedback RTP/RTCP packet. This paper describes using efficiant flow control on multicast that can reduce errors according to getting feedback tranfer delay and proposes the mechanism that can adapt dynamic change of network. In simulation, the transfer rate can efficiently be control on dynamic change of network and it makes the maximum of the use of a bandwidth and the minimum of packet losses.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.44 no.5
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• pp.18-29
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• 2007

The main objective of this paper is to propose a new stability analysis method for a networked control system with multiple packet transmission. The new scheduling method that can guarantee the maximum time delay and discrete switch state equation model which represent a network data loss is proposed. The equivalent model of a MIMO(multi-input multi-output) networked control system is derived from a state space model of linear time invariant interconnected systems in the form of asynchronous dynamical system. Using this model, this paper presents new stability theorems that can determine stability of the networked control system with regard to time delay, data loss, and the number of transmission packets. Simulation results verify the effectiveness of proposed stability analysis method.

• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.23-34
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• 2004

This paper presents architecture and control method of packet scheduler to guarantee QoS of high quality streaming services in high-speed packet-switched networks. Since streaming services need far more stringent QoS requirements than the typical sort of burst data applications, they should be guaranteed minimum bandwidth and end-to-end delay bound to each flow, regardless of the behavior of other flows. To meet these requirements, a packet scheduler isolate a flow from the undesirable effects of other flows and provides end-to-end delay guarantees for individual flow and divides stringently the available link bandwidth among flows sharing the link. Until now, many vendors are developing traffic management chips running at 10Gbps, but most of chips have drawbacks to support high quality streaming services. In this paper, we investigate the drawbacks of commercial TM chips and traffic characteristic of streaming services and present implementation frameworks of the proposed packet scheduler. Finally, we analyze the simulation results of the proposed scheduler.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.111-114
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• 2004

Sensor networks are deployed with a limited energy source. Thus, energy efficient design can be challenging. This paper has been studied optimal packet length with energy efficiency for sensor networks. And using Power Management can not improve energy efficiency. Power Management is turning off transceiver when transceiver is idle statue. We show that BCH code for error control can improve energy efficiency better than Convolutional code.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.782-787
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• 2004

Internet Engineering Task Force (IETF) has been considering the deployment of the Random Early Detection (RED) in order to avoid the increasing of packet loss rates which caused by an exponential increase in network traffic and buffer overflow. Although RED mechanism can prevent buffer overflow and hence reduce an average values of packet loss rates, but this technique is ineffective in preventing the consecutive drop in the high traffic condition. Moreover, it increases a probability and average number of consecutive dropped packet in the low traffic condition (named as "uncritical condition"). RED mechanism effects to TCP congestion control that build up the consecutive of the unnecessary transmission rate reducing; lead to low utilization on the link and consequently degrade the network performance. To overcome these problems, we have proposed a new mechanism, named as Extended Drop slope RED (ExRED) mechanism, by modifying the traditional RED. The numerical and simulation results show that our proposed mechanism reduces a drop probability in the uncritical condition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9B
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• pp.589-597
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• 2007

In this paper, a WDM metro ring consisting of access nodes with $FT-FR^n$ (Fixed Transmitter - n Fixed Receivers) is considered. A trade-off exists between node throughput and transmission fairness because the access nodes share wavelength channels. In order to eliminate the transmission unfairness and to increase throughput, the p-persistent medium access control (MAC) protocol is proposed: each node uses an empty optical slot to transmit a packet and make it available with the extraction of a transferred packet at the source access node, called source-stripping. The local empty slot can be used to transfer a head-of-line packet in the local buffer with probability p or it is used for the next downstream nodes with 1-p. The proposed MAC protocol provides better node throughput than the non-persistent protocol and exhibits better fairness index than the 1-persistent protocol in WDM ring networks. In addition, numerical analysis shows that the proposed MAC protocol maximizes the node throughput under uniform traffic conditions. For more detailed results, we use the network simulation under Poisson and self-similar traffic. Furthermore, unpredictable traffic constructed by the combination of the former and the latter is also considered. The reasonable probability of the p-persistent protocol for a given architecture can be determined through simulation.