• Journal of Communications and Networks
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• v.17 no.3
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• pp.286-295
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• 2015

Avionic databuses fulfill a critical function in the connection and communication of aircraft components and functions such as flight-control, navigation, and monitoring. Ethernet-based avionic databuses have become the mainstream for large aircraft owning to their advantages of full-duplex communication with high bandwidth, low latency, low packet-loss, and low cost. As a new generation aviation network communication standard, avionics full-duplex switched ethernet (AFDX) adopted concepts from the telecom standard, asynchronous transfer mode (ATM). In this technology, the switches are the key devices influencing the overall performance. This paper reviews the avionic databus with emphasis on the switch architecture classifications. Based on a comparison, analysis, and discussion of the different switch architectures, we propose a new avionic switch design based on a time-division switch fabric for high flexibility and scalability. This also merges the design concept of space-partition switch fabric to achieve reliability and predictability. The new switch architecture, called space partitioned shared memory switch (SPSMS), isolates the memory space for each output port. This can reduce the competition for resources and avoid conflicts, decrease the packet forwarding latency through the switch, and reduce the packet loss rate. A simulation of the architecture with optimized network engineering tools (OPNET) confirms the efficiency and significant performance improvement over a classic shared memory switch, in terms of overall packet latency, queuing delay, and queue size.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.45-50
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• 2008

Many applications of sensor network require connection to the Internet. The transmission protocol of traditional sensor network was designed within the sensor network itself. However, based on 6LoWPAN which can be accessed using IPv6, direct connection is possible between the sensor network and the TCP/IP network outside. Transmission of data in applications of sensor network falls into two main categories. One is a small packet that is periodically produced such as packet related to temperature and humidity. The other is a relatively large packet that brings about network overheads such as images. We investigated the conformance test and pros and cons of application data over the transmission protocol of Zigbee and 6LoWPAN. As a result, both Zigbee and 6LoWPAN have shown low rate of loss for periodic data and have in creased reliability of data transfer. When transmitting streaming image data, both ACK, non ACK mode of Zigbee and UDP of 6LoWPAN minimized transmission time but suffered the consequences of high packet loss. Even though TCP of 6LoWPAN required a long transmission time, we were able to confirm that no loss has occurred.

• Journal of the Korea Society of Computer and Information
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• v.7 no.1
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• pp.112-119
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• 2002

We propose an enhanced wavelet packet image coder algorithm which is based on the coefficients partition. The proposed wavelet packet image coder uses the first-order entropy to reduce the total compression time, and achieves low bit rates and rate-distortion performance by the zero-tree based coding using correlations between coefficients partition. This new algorithm represents new parent-children relationships for reducing image reconstruction error using the correlations between each frequency subbands and then the wavelet packet coefficients are Partitioned by a new order. The computer simulations demonstrate higher PSNR under the same bit rate and improved image compression time and enhanced rate control compare with conventional algorithms. From the simulation results, it is shown that the encoding and decoding process of proposed coder are much simple and accurate than present method against texture images , which include many mid-frequency elements.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.165-167
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• 2014

In this paper, an packet drop technique is proposed to outperform the previous RIO (RED In and Out) drop mechanism for DiffServ in ship area networks. the proposed packet drop technique does not manage the individual flows and divides them into several flow groups according to a criterion. And it guarantees the fairness between individual flows in the same QoS class through the group-based control. In simulation results of the proposed packet drop technique, the link utilization decreases than RIO. But it guarantees more data rates to DiffServ flows passing multiple bottleneck links.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.358-366
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• 2010

In the wireless network the congestion and delay occurs mainly when there are too many packets for the network to process or the sender transmits more packets than the receiver can accept. The congestion and delay is the reason of packet loss which degrades the performance of multimedia streaming. This paper proposes a novel transmission rate monitoring-based optimization mechanism to optimize packet loss and to improve QoS. The proposed scheme is based on the trade-off relationship between transmission rate monitoring and overhead monitoring. For this purpose this paper processes a source rate control-based optimization which optimizes congestion and delay. Performance evaluated RED, TFRC, and the proposed mechanism. The simulation results show that the proposed mechanism is more efficient than REC(Random Early Detection) mechanism and TFRC(TCP-friendly Rate Control) mechanism in packet loss rate, throughput rate, and average response rate.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.91-100
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• 2004

This paper presents a receiver-driven TCP flow control mechanism, which is adaptive to the wireless condition, for memory constrained mobile receiver. A receiver-driven TCP flow control is, in general, achieved by adjusting the size of advertised window at the receiver. The proposed method constantly measures at the receiver both the available wireless bandwidth and the packet round-trip time. Depending on the measured values, the receiver adjusts appropriately the size of advertised window. Constrained by the adjusted window which reflects the current state of the wireless network, the sender achieves an improved TCP throughput as well as the reduced round-trip packet delay. Its implementation only affects the protocol stack at the receiver and hence neither the sender nor the router are required to be modified. The mechanism has been implemented in real environments. The experimental results show that in CDMA2000 1x networks the TCP throughput of the proposed method has improved about 5 times over the conventional method when the receiver's buffer size is limited to 2896 bytes. Also, with 64Kbytes of buffer site, the packet round-trip time of the proposed method has been reduced in half, compared the case with the conventional method.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.1-10
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• 2004

This study discusses Application-layer FEC using erasure codes. Because of the simple decoding process, erasure codes are used effectively in Application-layer FEC to deal with Packet-level errors. The large number of parity packets makes the loss rate to be small, but causes the network congestion to be worse. Thus, a redundancy control algorithm that can adjust the number of parity packets depending on network conditions is necessary. In addition, it is natural that high-priority frames such as I frames should produce more parity packets than low-priority frames such as P and B frames. In this paper, we propose a redundancy control algorithm that can adjust the amount of redundancy depending on the network conditions and depending on data priority, and test the performance in simple links and congestion links.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6C
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• pp.581-593
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• 2002

In this paper, we proposed an Adaptive RIO scheme to solve the problem of RIO scheme that occurs when admission control is performed for QoS guarantee of Assured Service in Differentiated Services. To prevent an early random drop of the admitted In-profile packet, proposed Adaptive RIO scheme updates parameters of RIO scheme every time interval according to the estimated numbers of maximum packet arrivals of In-profile traffic and total traffic during the next time interval. The numbers of maximum packet arrivals during the next time interval are estimated based on the buffer size determined by the network topology and the ratio of bandwidth allocated to each subclass. We found from simulation results that, compared with RIO scheme, proposed Adaptive RIO scheme can improve performance of the throughput for In-profile traffic when admission control is performed or congestion occurs.

• Journal of Advanced Navigation Technology
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• v.17 no.2
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• pp.218-224
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• 2013

Uplink overflow in WMSN (Wireless Multimedia Sensor Networks) aggravates the resource consumption, delay, and traffic congestion. This paper proposes a new traffic congestion control mechanism using popularity. The proposed mechanism controls congestions by dispersing the media traffic, and it control fairly packets according to priority. This paper proposes PQS (Packet Queue Scheduler) to control fairly packets, and the proposed mechanism provides a fair opportunity to all sensor nodes without a specific location. The simulation results show that the proposed mechanism achieves improved performance in throughput, delay ratio, link quality, and buffer queue control ratio compared with those of other existing methods.