• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.227-238
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• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.

• ETRI Journal
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• v.39 no.2
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• pp.213-221
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• 2017

The advent of wireless access in vehicular environments (WAVE) technology has improved the intelligence of transportation systems and enabled generic traffic problems to be solved automatically. Based on the IEEE 802.11p standard for vehicle-to-anything (V2X) communications, WAVE provides wireless links with latencies less than 100 ms to vehicles operating at speeds up to 200 km/h. To date, most research has been based on field test results. In contrast, this paper presents a numerical analysis of the V2X broadcast throughput limit using a path loss model. First, the maximum throughput and minimum delay limit were obtained from the MAC frame format of IEEE 802.11p. Second, the packet error probability was derived for additive white Gaussian noise and fading channel conditions. Finally, the maximum throughput limit of the system was derived from the packet error rate using a two-ray path loss model for a typical highway topology. The throughput was analyzed for each data rate, which allowed the performance at the different data rates to be compared. The analysis method can be easily applied to different topologies by substituting an appropriate target path loss model.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.9
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• pp.30-38
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• 2008

The traditional TCP was designed assuming wired networks. Thus, if it is used networks consisting of both wired and wireless networks, all packet losses including random losses in wireless links are regarded as network congestion losses. Misclassification of packet losses causes unnecessary reduction of transmission rate, and results in waste of bandwidth. In this paper, we present WTCP(wireless TCP) congestion control algorithm that differentiates the random losses more accurately, and adopts improved congestion control which results in better network throughput. To evaluate the performance of proposed scheme, we compared the proposed algorithm with TCP Westwood and TCP Veno via simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12B
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• pp.1106-1114
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• 2003

Several analytic models describe transmission control protocol (TCP) performance such as steady-state throughput as an averaged ratio of number of transmissions to latency. For more detailed analysis of TCP latency, the latency during packet losses are recovered should be considered. In this paper, we derive the expected duration of loss recovery latency considering the number of packet losses recovered by retransmissions. Based on the numerical results verified by simulations, TCP using selective acknowledgement (SACK) option is more effective than TCP NewReno from the aspect of loss recovery latency.

• The Journal of the Korea Contents Association
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• v.4 no.4
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• pp.189-196
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• 2004

Most of the recent researches for ATM switches have been based on multistage interconnection network known as regularity and self-routing property. These networks can switch packets simultaneously and in parallel. However, they are blocking networks in the sense that packet is capable of collision with each other Mainly Banyan network have been used for structure. There are several ways to reduce the blocking or to increase the throughput of banyan-type switches: increasing the internal link speeds, placing buffers in each switching node, using multiple path, distributing the load evenly in front of the banyan network and so on. Therefore, this paper proposes the use of recirculating shuffle-exchange network to reduce the blocking and to improve hardware complexity. This structures are recirculating shuffle-exchange network as simplified in hardware complexity and Rank network with tree structure which send only a packet with highest priority to the next network, and recirculate the others to the previous network. after it decides priority number on the Packets transferred to the same destination, The transferred Packets into banyan network use the function of self routing through decomposition and composition algorithm and all they arrive at final destinations. To analyze throughput, waiting time and packet loss ratio according to the size of buffer, the probabilities are modeled by a binomial distribution of packet arrival. If it is 50 percentage of load, the size of buffer is more than 15. It means the acceptable packet loss ratio. Therefore, this paper simplify the hardware complexity as use of recirculating shuffle-exchange network instead of bitonic sorter.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.4
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• pp.211-219
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• 2012

In this paper, we described the software development for verifying the integrity of output data of TCP/IP for VLBI Correlation Subsystem (VCS) correlation block and proposed the performance improvement method in order to prevent the data loss of correlation output. The VCS correlation results are saved at the Data Archive system through TCP/IP packet transmission. In this paper, the integrity verification software is developed so as to confirm the integrity of correlation result saved at the data archive system using TCP/IP packet information of VCS. The 3-step integrity verification process is proposed by using the developed software, its effectiveness was confirmed in consequence of correlation experiments. In addition, TCP/IP packet transmission must be completed within minimum integration period. However, there is not only TCP/IP packet loss occurred but also the problem of correlation result integrity affected in account of a large quantity of packets and data during short integration time. In this paper, the reason of TCP/IP packet loss is analyzed and the modified methods for FPGA(Field Programmable Gate Array) of VCS are proposed, the integrity problem of correlation results will be solved.

• The Journal of the Acoustical Society of Korea
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• v.32 no.1
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• pp.86-94
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• 2013

In this paper, we propose a network coding scheme that is one of the most promising techniques for overcoming transmission errors in underwater acoustic communications. It is assumed that the proposed scheme operates in a Code Division Multiple Access (CDMA) network where multiple sensor nodes share the underwater acoustic channel in both the frequency and the time domains by means of orthogonal codes. The network topology deploys multi-hop transmission with relaying between multiple source nodes and one destination node via multiple relay nodes. The proposed scheme is evaluated in terms of the successful packet delivery ratio of end-to-end transactions under varying packet loss rates. A computer simulation shows that the successful delivery ratio is maintained at over 95% even when the packet loss rate reaches 50%.

• 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.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.307-310
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• 2000

ATM is the switching and multiplexing technology chosen by the ITU-T for the operation of B-lSDN. Basically, ATM technology is designed to combine the reliability of circuit switching with the efficiency and flexibility of packet switching technology. For servicing QoS in IPOA(IP over ATM) when the larger effort is given, it will be the good method that the original QoS benefits having ATM switching have in ATM layer underlying layer. The IETF has recently proposed Differentiated Services framework for provision of QoS. In this paper we analyse performance of two Diffserv mechanism. Threshold Dropping and Priority Scheduling. Threshold Dropping and Priority Scheduling can be regarded as basic mechanisms from which the other mechanisms have been derived. Hence comparative performance of these two mechanisms in providing required QoS is an important issue. In this Paper we carry out a performance comparison of the TD and PS mechanisms with the aim of providing the same level of packet loss to the preferred flow. Our comparison of the TD and PS allows us to determine resultant packet loss for the non-preferred flows as a function of various parameters of the two mechanisms.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.7 s.337
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• pp.19-26
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• 2005

IETF proposed the Fast Handover mechanism to reduce the latency during which the mobile node is effectively disconnected from the Internet. However, the Fast Handover mechanism did not resolve packet loss and packet disordering problem. In this paper, we propose buffering mechanisms to resolve above problems in the Fast Handover mechanism. Though the simulation, we showed that packet loss and disordering problem have been absolutely resolved. In addition, our proposal reduced about 27$\%$ of the delay time by the buffering mechanism.