• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• Communications of the Korean Mathematical Society
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• v.14 no.2
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• pp.405-423
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• 1999

We consider a statistical multiplexer model with finite buffer capacity and finite number of independent identical 3-state bursty voice sources. The burstiness of the sources is modeled by describing both two different active periods (at the rate of one packet perslot) and the passive periods during which no packets are generated. Assuming a mixture of two geometric distributions for active period and a geometric distribution for passive period and geometric distribution for passive period, we derive the recursive algorithm for the probability mass function of the buffer contents (in packets). We also obtain loss probability and the distribution of packet delay. Numerical results show that the system performance deteriorates considerably as the variance of the active period increases. Also, we see that the loss probability of 2-state Markov models is less than that of 3-state Markov models.

• Journal of applied mathematics & informatics
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• v.8 no.1
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• pp.81-95
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• 2001

This paper models and evaluates the AAL multiplexer to analyze AAL protocol in ATM networks. We consider an AAL multiplexer in which a single periodically determinsitic CBR traffic stream and several variable size bursty background traffic streams are multiplexed and one ATM cell stream goes out. We model the AAL multiplexer as a B/sup X/ + D/D/1/K queue and analyze this queueing system. We represent various performance measures such as loss probability and waiting time in the basis of cell and packet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1824-1831
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• 1999

In this paper, the data recovery performance of systematic erasure codes in burst loss environments is analyzed and the estimation method of redundant data according to loss characteristics is suggested. The burstness of packet loss is modeled by Gilbert model, and the performance of proposed packet loss recovery method in the case of using systematic erasure code is analyzed based on previous study on the loss recovery in the case of using erasure code. The required redundancy data fitting method for systematic erasure code in the condition of given loss property is suggested in the consideration of packet loss characteristics such as average packet loss rate and average loss length.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10A
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• pp.1756-1763
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• 2001

In this paper, we propose an ATM-LAN IWU(interworking unit) with threshold based dynamic bandwidth allocation scheme. We analyze a discrete-time based finite queueing model with deterministic service times in order to investigate the performance of the proposed scheme. It is known that the arrival process of IP packets is bursty. So we use an MMPP(Markov Modulated Poisson Process) to model the bursty input traffic. As performance measures, we obtain the packet loss probability and the mean packet delay. We present some numerical results to show the effects of the thresholds on the performance of the DBAS(dynamic bandwidth allocation scheme).

• Journal of Korean Institute of Industrial Engineers
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• v.28 no.2
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• pp.139-146
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• 2002

The third generation mobile communication needs to provide multimedia service with increased data rates. Thus an efficient allocation of radio and network resources is very important. This paper models the 'burst switching' as an efficient radio resource allocation scheme and the performance is compared to the circuit and packet switching. In burst switching, radio resource is allocated to a call for the duration of data bursts rather than an entire session or a single packet as in the case of circuit and packet switching. After a stream of data burst, if a packet does not arrive during timer2 value ($\tau_{2}$), the channel of physical layer is released and the call stays in suspended state. Again if a packet does not arrive for timerl value ($\tau_{1}$) in the suspended state, the upper layer is also released. Thus the two timer values to minimize the sum of access delay and queuing delay need to be determined. In this paper, we focus on the decision of $\tau_{2}$ which minimizes the access and queueing delay with the assumption that traffic arrivals follow Poison process. The simulation, however, is performed with Pareto distribution which well describes the bursty traffic. The computational results show that the delay and the packet loss probability by the burst switching is dramatically reduced compared to the packet switching.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.707-715
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• 2003

In this paper, we propose a new MAC(Medium Access Control) protocol using Movable-boundary, which tries to guarantee Qos for multimedia traffic in the slotted CDMA system. In this scheme, the traffic scheduler assigns channel resource according to the packet priority per service class and adapts the Movable-boundary concept in which the minimum resource is assigned to each traffic class; the remaining resource if it is available can be assigned dynamically according to the temporal demand of other traffic classes. For performance analysis, we performed computer simulations to obtain throughput and packet loss rate and compared the results with Fixed-boundary system. We observed that the error rate of voice traffic could be maintained below a prescribed value while bursty traffic such as video source shares the same channel. In comparison with Fixed-boundary scheme, our protocol exhibits better throughput and packet loss rate performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.396-405
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• 2005

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem; it cannot perform local retransmission efficiently under the bursty-error prone wireless link. In this paper, we propose Enhanced Snoop(E-Snoop) protocol to solve this problem of Snoop protocol. With E-Snoop protocol, packet losses can be noticed by receiving new ACK packets as well as by receiving duplicate ACK packets or local retransmission timeout. Therefore, TCP throughput can be enhanced by fast recognition of bursty packet losses and fast local retransmissions. From the simulation results, E-Snoop protocol can improve TCP throughput more efficiently than Snoop protocol and can yield more TCP improvement especially in the channel with high packet loss rates.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.5
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• pp.21-31
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• 2002

As the MPEG streaming traffics are transmitted over the Internet, MPEG packet losses cause error propagation in both spatial and temporal domains which in turn leads to severe degradation in image quality. This image degradation is become worse by the bursty packet loss at the network. In this paper, we propose a new error spreading method to reduce the effects of bursty packet losses over the Internet. The proposed error spreading method spreads out the not utilize packet losses widely, and can be used for videocasting and video-broadcasting that do not utilize the ARQ method. The proposed method can not only reduce the consecutive losses of image blocks, but also improve the performance of error concealment. Experimental results show that the deterioration of MPEG video quality can be reduced by spreading out the MPEG packet losses.