• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.1C
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• pp.63-75
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• 2012

This paper researched the operation methods of software video teleconferencing on the tactical communication networks under 1Mbps. The tactical communication networks have limited bandwidths, frequent data losses and transmission delays due to the unstable networks. In addition, the bandwidth for video teleconferencing has to be much smaller since the Army Tactical Command Information System(ATCIS) has priority of using the bandwidth. This paper analyzed such restrictions of tactical communication networks, presented some methods to improve the quality of the software video teleconferencing on the tactical communication networks and their actual experiments as well. It is applied in the first place to re-transmit the lost packets and to reduce the image size for the data traffic. Nothing is better for the video teleconferencing than to provide the bandwidth enough for every user. However, on the tactical communication networks with the limited bandwidth, video teleconferencing can be improved by optimizing the compression rate of image data, the number of image frames, the audio codec and the usage of audio compensation data.

• The Transactions of the Korea Information Processing Society
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• v.2 no.6
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• pp.927-936
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• 1995

ATM Forum has defined a new service class for data applications called Available Bit Rate(ABR) Service, which has highly bursty traffic and unpredictable burst size. It is desirable that we reduce the probability of retrans mission of packets by minimizing the loss of cells because the traffic is much more sensitive to loss than delay. The Forum has also selected the Rate-Based Control for the ABR service and proposed EPRCA as the control mechanism for the service. This paper proposes the combination of EPRCA and the other feedb ack control mechanisms such as BECN and BP. The combined control mechanism control ABR traffic more efficiently and the simulation results show that the network performance can be improved by choosing the appropriate parameters.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7A
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• pp.770-777
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• 2007

Smaller cell size in the micro-cell or pico-cell structure brings about more frequent hand-overs between cells, and higher speed movement of the mobile terminal makes short the permissible time interval for executing these hand-over procedures. In this situation the hand-over failure may occur or some packets may be lost during the hand-over. Moreover a rapid degradation of throughput is triggered by packet re-transmission for compensating such errors. The QoS (Quality of Services) of mobile multimedia applications with higher bits rate requirements and higher speed mobility are severely affected by even shot service interruption. This paper proposes a new hand-over scheme to provide seamless services in the next generation mobile communication systems. Simulation is done to evaluate the performance of the proposed scheme based on its hand-over failure rate and packet loss rate.

• Journal of the Korea Society of Computer and Information
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• v.15 no.11
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• pp.135-142
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• 2010

QoS in the proxy system are under heavy influence from interferences such as congestion, latency, and retransmission. Also, multi-level streaming services affects from temporal synchronization, which lead to degrade the service quality. This paper proposes a new segment-based buffer management mechanism which reduces performance degradation of streaming services and enhances throughput of streaming due to drawbacks of the proxy system. The proposed paper optimizes streaming services by: 1) Use of segment-based buffer management mechanism, 2) Minimization of overhead due to congestion and interference, and 3) Minimization of retransmission due to disconnection and delay. This paper utilizes fuzzy value $\mu$ and cost weight $\omega$ to process the result. The simulation result shows that the proposed mechanism has better performance in buffer cache control rate, average packet loss rate, and delay saving rate with stream relevance metric than the other existing methods of fixed segmentation method, pyramid segmentation method, and skyscraper segmentation method.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• Journal of Korea Society of Industrial Information Systems
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• v.19 no.1
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• pp.13-18
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• 2014

Sensor nodes are used for monitoring and collecting the environmental data via wireless sensor network. The wireless sensor network with various sensor nodes draws attention as a key technology in ubiquitous computing. Sensor nodes has very small memory capacity and limited power resource. Thus, it is essential to have energy efficient strategy for the sensor nodes. Since the sensor nodes are operating on the same frequency bands with ISM frequency bands, the interference by the devices operating on the ISM band degrades the quality of communication integrity. In this paper, the convolutional code is proposed instead of ARQ for the error control for the sensor network. The proposed convolutional code was implemented and the BER performance is measured. For the fixed transmitting powers of -19.2 dBm and -25dBm, the BER with various communication distances are measured. The packet loss rate and the retransmission rate are calculated from the measured BER. It is shown that the porposed method obtained about 9~12% and 12-19% reduction in retransmission rate for -19.2 dBm and -25 dBm respectively.

• Journal of the Korea Society of Computer and Information
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• v.15 no.8
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• pp.79-87
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• 2010

It is widely accepted that the coverage with high user densities in mobile multimedia environments can only be achieved with small cell such as micro- and pico-cell. If handover events occur during the transmission of multimedia, the efficient resource reservation and handover methods are necessary in order to maintain the same QoS of transmitted multimedia traffic because the QoS may be defected by some delay and information loss. In this paper, we propose a resource allocation method in the next generation mobile communication systems, in which the resource allocation process has a tight relation with call admission, call load, and packet scheduling. The simulation results show that our proposed method provides a excellent performance.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.9-12
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• 2000

In this paper, we carry out a performance study related to the Advanced Peer-to-Peer Network(APPN). For this particular network, it has been proposed to use the leaky bucket as a way of controlling congestion within the network. On the top of leaky bucket type rate based congestion control scheme for high speed networks, a user will typically operate an error control scheme for retransmitting lost and erroneous packets. We propose a Perform ance model in order to study the Interaction between a user's error control scheme and the leaky bucket congestion control scheme for high speed networks. Simulation results show that parameters such as the window size and the token generation rate in the leaky bucket are key factors affecting the end-to-end delay.

• Journal of the Korean Institute of Intelligent Systems
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• v.27 no.1
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• pp.1-6
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• 2017

Recent researches on radio frequency energy harvesting networks(RF-EHNs) with limited energy resource like battery have been focusing on the development of a new scheme that can effectively extend the whole lifetime of a network to semipermanent. In order for considerable increase both in the amount of energy obtained from radio frequency energy harvesting and its charging effectiveness, it is very important to design a network that supports energy harvesting and data transfer simultaneously with the full consideration of various characteristics affecting the performance of a RF-EHN. In this paper, we proposes an interference-based charging aware routing protocol(ICARP) that utilizes interference information and charging time to maximize the amount of energy harvesting and to minimize the end-to-end delay from a source to the given destination node. To accomplish the research objectives, this paper gives a design of ICARP adopting new network metrics such as interference information and charging time to minimize end-to-end delay in energy harvesting wireless networks. The proposed method enables a RF-EHN to reduce the number of packet losses and retransmissions significantly for better energy consumption. Finally, simulation results show that the network performance in the aspects of packet transmission rate and end-to-end delay has enhanced with the comparison of existing routing protocols.

• Journal of KIISE:Information Networking
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• v.31 no.6
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• pp.595-607
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• 2004

Networks often evolve to provide a host with multiple access points to the Internet. In this paper, we propose a transport layer load distribution mechanism utilizing the multiple network interfaces simultaneously. We specifically propose an extension of Stream Control Transmission Protoco1 (SCTP) to have load sharing over multiple network interfaces. We named the particular service provided by the Proposed load sharing mechanism to be LS (Load Sharing) mode service. LS mode service is based on the following four key elements: (i) the separation of flow control and congestion control, (ii) congestion window based striping, (iii) redundant packet retransmission for fast packet loss recovery, (iv) a novel mechanism to keep track of the receiver window size with the SACKS even if they arrive out-of-order. Through simulations, it is shown that the proposed LS mode service can aggregate the bandwidth of multiple paths almost ideally despite of the disparity in their bandwidth. When a path with a delay of 100% greater is utilized as the second path, the throughput is enhanced about 20%.