• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• 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 KIISE:Information Networking
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• v.29 no.4
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• pp.447-458
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• 2002

In this paper, we design and implement Wireless Transaction Protocol(WTP) and evaluate it for wireless transaction processing in mobile computing environments. The design and implementation of WTP are based on the coroutine model that might be suitable for light-weight portable devices. We test the compatibility between our product and the other products such as Nokia, Kannel and WinWAP For the evaluation of WTP, we use an Internet simulator that can arbitrary generate random wireless errors based on the Gilbert model. In our experiment, the performance of WTP is measured and compared to those of Transmission Control Protocol(TCP) and TCP for Transactions. The experiment shows that WTP outperforms the other two protocols for wireless transaction processing in terms of throughput and delay. Especially, WTP shows much higher performance In ease of high error rate and high probability of burst errors. This comes from the fact that WTP uses a small number of packets to process a transaction compared to the other two protocols and introduces a fixed time interval for retransmission instead of the exponential backoff algorithm. The experiment also shows that the WTP performance is optimized when the retransmission counter is set to 5 or 6 in case of high burst error rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12B
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• pp.1183-1191
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• 2010

In this paper, I propose an adaptive OFDM CP length algorithm for in PLC systems for smart grid. The proposed scheme calculates the channel delay information at the CP controller of the receiver by taking correlation between a received data frame and the following delayed one. The CP controller, immediately, feeds back the channel delay information to the transmitter. Then, the transmitter adapts CP length for next data frame. As an impulsive noise model, Middleton Class A interference model was employed. The performance is evaluated in terms of packet data rate, cumulative packet data rate, and bit error rate (BER). The simulation results showed data gain (which is the amount of the reduced bits) gets larger as the number of packets increase, but the amount of data gain reduced as the number of branches ($N_{br}$) increase. In respects of BER for the cases $N_{br}$ is 3, 4, and 5, performance of the adaptive CP length algorithm and the fixed CP scheme are similar. Therefore, it is confirmed the proposed scheme achieved data rate increment without BER performance reduction compared to the conventional fixed CP length scheme.

• Journal of Internet Computing and Services
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• v.10 no.2
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• pp.15-28
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• 2009

This paper proposes HWbF(Hit and WLC based Firewall) design which consists of an PFS(Packet Filter Station) and APS(Application Proxy Station). PFS is designed to reduce bottleneck and to prevent the transmission delay of them by distributing packets with PLB(Packet Load Balancing) module, and APS is designed to manage a proxy cash server by using PCSLB(Proxy Cash Server Load Balancing) module and to detect a DoS attack with packet traffic quantity. Therefore, the proposed HWbF in this paper prevents packet transmission delay that was a drawback in an existing Firewall, diminishes bottleneck, and then increases the processing speed of the packet. Also, as HWbF reduce the 50% and 25% of the respective DoS attack error detection rate(TCP) about average value and the fixed critical value to 38% and 17%. with the proposed expression by manipulating the critical value according to the packet traffic quantity, it not only improve the detection of DoS attack traffic but also diminishes the overload of a proxy cash server.

• Journal of KIISE:Information Networking
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• v.37 no.3
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• pp.204-216
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• 2010

Since Sensor nodes around a fixed sink have huge concentrated network traffic, the battery consumption of them is increased extremely. Therefore the lifetime of sensor networks is limited because of huge battery consumption. To address this problem, a mobile sink has been studied for load distribution among sensor nodes. Since a mobile sink changes its location in sensor networks continuously, the mobile sink has time limits to communicate with each sensor node and unstable signal strength from each sensor node. Therefore, a fair and stable data collection method between a mobile sink and sensor nodes is necessary in this environment. When some sensor nodes are not able to send data to the mobile sink, a real-time application in sensor networks cannot be provided. In this paper, the new scheduling method, FQMS (Fair Queuing for Mobile Sink), is proposed for fair and stable data collection for mobile sinks in sensor networks. The FQMS guarantees balanced data collecting between sensor nodes for a mobile sink. In out experiments, the FQMS receives more packets from sensor nodes than legacy scheduling methods and provides fair data collection, because moving speed of a mobile sink, distance between a mobile sink and sensor nodes and the number of sensor nodes are considered.