• Journal of Digital Contents Society
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• v.16 no.3
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.53-60
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• 2011

In Wireless Sensor Networks, IEEE 802.11 happen unnecessary energy consume because of packet transmission using maximum power between sensor node. The BASIC scheme is to use maximum transmission power for RTS-CTS and minimum required transmission power so as to high energy efficiency for DATA-ACK. However BASIC scheme may degrade network throughput with collision of ACK packet by node in carrier sensing zone and may result in higher energy consumption than when using IEEE 802.11 without power control. Existing PCM(Power Control MAC) scheme is to use DATA packet transmission method by periodically maximum power level so as to sensing DATA packet transmit in carrier sensing zone of transmission node, and this method can avoid collision of ACK packet. This paper present problem by energy efficiency of PCM scheme, and design some more improved PCM scheme.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2012.05a
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• pp.110-113
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• 2012

DDS middleware transmits data packets through unicast or multicast method for discovering endpoints on a distributed network and exchanging information between endpoints. Depending on the communicating situation between endpoints, the transmission method which is used to exchange packets is already defined by the standard protocol. However, the predefined usage of the transmission method has a problem that the middleware does not deal with increasing usage of the system resources properly when the configuration of endpoint is changed dynamically. To solve this problem, we propose the scheme of selecting the packet transmission method in an adaptive manner.

• Proceedings of the Korea Society for Simulation Conference
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• 2005.05a
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• pp.1-4
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• 2005

Information communication environment integrates communication, broadcasting and internet, and Digital Convergence service emerges in result. Thus, the effective routers are needed so that they can transmit a huge number of data to core internet through appropriate base center. Therefore, the network guaranteeing QoS in transport layer supports interoperability with different wireless networks. So as to users receive necessary information anywhere seamlessly, the network architecture focuses on packet transmission and it is efficient for the control layer switches and controls packets between different networks. Since individual users take advantage of different services and data, the effective router architecture must be designed. Hence in this paper we design monitoring technique to solve security problem and to support premium service to ultimate users. Thereafter, we run opnet simulation and show the improvement of proposed router architecture.

• Journal of Advanced Navigation Technology
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• v.13 no.4
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• pp.523-531
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• 2009

In this study, we propose a joint source/channel rate control algorithm for video encoder targeting packet erasure channel. Based on the buffer constraints of video communication systems, encoding rate constraint is presented. After defining source distortion models for coded and skipped video frames and a channel distortion model for packet errors and their propagation, an average distortion model of received video is proposed for a given encoding window. Finally, we define an optimization problem to minimize the average distortion for given channel rates and packet loss rates by controlling spatio-temporal parameters of source video and FEC block sizes. Then, we propose a window-based algorithm to solve the problem in real-time.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• The Journal of the Korea Contents Association
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• v.6 no.12
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• pp.96-104
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• 2006

In this study, data transmission ratio and power consumption issues of Zigbee based sensor module and personal digital assistant(PDA) were addressed to develop ECG telemetry device. PDA processes the data transmitted through serial port using non-blocking method. The transmission rate was dependent on the packet structure. It was 300 ECG samples/sec, when each packet was composed of 2 ECG data and 3-axial acceleration vector. Using two AAA batteries in series, operating time of the wireless sensor module was above 28 hours in average. Power consumption of PDA was dependent on screen ON/OFF condition and serial port usage. In this application, operating time of PDA was 5 hours in average. In conclusion, there was no problem in the power consumption of wireless sensor module and transmission rate, when the developed device was used as 24 hour Holter device. But, PDA has the problem of power consumption, which should be solved.

• Journal of the Korea Society of Computer and Information
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• v.21 no.6
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• pp.47-53
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• 2016

This paper deals with the capacity constrained time slot assignment problem(CTSAP) that a satellite switches to traffic between $m{\times}n$ ground stations using on-board $k{\leq}_{min}\{m,n\}$ k-transponders switching modes in SS/TDMA time-division technology. There was no polynomial time algorithm to solve the optimal solution thus this problem classified by NP-hard. This paper suggests a heuristic algorithm with O(mn) time complexity to solve the optimal solution for this problem. Firstly, the proposed algorithm selects maximum packet lengths of $\({mn \atop c}\)$ combination and transmits the cut of minimum packet length in each switching mode(MSMC). In the case of last switching mode with inefficient transmission, we applies a compensation strategy to obtain the minimum number of switching modes and the minimum makespan. The proposed algorithm finds optimal solution in polynomial time for all of the experimental data.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.12
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• pp.1054-1060
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• 2013

This paper proposes a vehicle ad-hoc network (VANET) routing algorithm to reduce the probability of routing path discontinuity in urban intersections. In urban intersections, the vehicles may frequently change their moving directions. It can cause the disconnection of routing path and the increase of a packet transmission delay. In order to resolve this problem, the proposed routing algorithm exploits the information of moving directions in urban intersections. In this way, the proposed algorithm can reduce the probability of the local maximum which causes the increase of the number of routing hops and packet transmission delay. Simulation results show that the proposed algorithm can reduce the local maximum probability by 10% and increase the successful packet transmission ratio by 5% compared to the conventional VANET routing algorithms.