• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.1
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• pp.35-46
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• 1994

In this paper, the performance of DS/SS acquisition system over frequency nonselective Rician fading channel is analyzed by means of packet loss probability. The power ratio of the fading component to the specular compnent. seccessive constant fadong chips k. and correlation coefficient among k chipe are taken for channel parameters. The false alarm probabilities and detection probabilities are derived, and packet loss probability is evaluated in terms of these probablities in the stats transition diagram. From the results of the performance test, these exists the region of packet loss probability in crease because of autocorrelation sidelobe. As k increases, the packet loss probabolotoes decrease.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.5
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• pp.957-962
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• 2013

This paper proposes the new enhanced 2-D(2-Dimension) FEC scheme. It analyzes the probability of entire packet loss rate of the existing 2-D FEC by mathematical modeling, finds the problem of the existing 2-D FEC, and deduces the new enhanced 2-D FEC scheme that reduces the entire packet loss probability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4A
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• pp.259-270
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• 2005

This paper proposes a mathematical model to analyze call-and packet-level performance of the TD-CDMA/TDD system which could serve a flexible radio resource management against multi-type heterogeneous and asymmetrical traffic conditions. On call-level analysis, the mathematical model based on queueing theory performs multi-dimensional operations using random vectors or matrices to consider multiple types of traffic and also deal with asymmetrical up- and down-direction transmissions separately. Employing the mathematical model, we obtain rail blocking probability for each type of traffic and also the optimum switching-point with the smallest call flocking probability. And on packet-level analysis, employing a non-prioritized queueing scheme between circuit and packet calls, we solve 2-dimensional random vector problem composed of the queue length for packets and the number of circuit calls being served. Finally, packet-level performance is analyzed in terms of the packet loss probability and the buffer size required under mixed-traffic conditions of multiple types of circuit and packet calls.

• 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.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.307-310
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• 2007

In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed to improve the quality of decoded speech under a burst packet loss condition. The proposed algorithm is based on the recovery of voiced excitation using an estimate of the voicing probability and the generation of random excitation by permutating the previously decoded excitation. The voicing probability is estimated from the correlation using the previous correctly decoded excitation and pitch. The proposed algorithm is implemented as a PLC algorithm for G.729 and its performance is compared with PLC employed in G.729 by means of perceptual evaluation of speech quality (PESQ) and an A-B preference test under the random and burst packet losses with rates of 3% and 5%. It is shown that the proposed algorithm provides better speech quality than the PLC of G.729, especially under burst pack losses.

• Journal of Information Processing Systems
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• v.2 no.1
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• pp.34-38
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• 2006

In the wireless network, TCP performs poorly because it was originally designed for wired networks and does not take into consideration wireless characteristics such as mobility, high-loss probability, and hidden-terminal problems. In particular, in the wireless multi-hop networks, a large congestion window increases the probability of contention and packet losses, and TCP performance is degraded severely as a result. So, it is necessary to limit the TCP congestion window size in order keep the probability of contention loss in the system to a minimum. In this paper, we propose a new scheme for determining the maximum congestion window size based on the measured bandwidth and Round-Trip-Time (RTT). Using ns-2 simulation, we show that the proposed scheme reduces the probability of packet contention and improves TCP performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4B
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• pp.288-297
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• 2003

The fast retransmit and fast recovery algorithm of TCP Reno, when multiple packets in the same window are lost, cannot recover them without RTO (Retransmission Timeout). TCP New-Reno can recover multiple lost packets by extending fast recovery using partial acknowledgement. If the retransmitted packet is lost again during fast recovery, however, RTO cannot be avoided. In this paper, we propose an algorithm called "Duplicate Acknowledgement Counting(DAC)" to alleviate this problem. DAC can detect the retransmitted packet loss by counting duplicate ACKs. Conditions that a lost packet can be recovered by loss recovery of TCP Reno, TCP New-Reno and TCP New-Reno using DAC are derived by modeling loss recovery behavior of each TCP. We calculate the loss recovery probability for random packet loss probability numerically, and show that DAC can improve loss recovery behavior of TCP New-Reno.

• Annual Conference of KIPS
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• 2008.11a
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• pp.1332-1335
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• 2008

In order to various service types of real time and non-real time traffic with varying requirements are transmitted over the IEEE 802.16 standard is expected to provide quality of service(QoS) researchers have explored to provide a queue management scheme with differentiated loss guarantees for the future Internet. The sides of a packet drop rate, an each class to differential drop probability on achieving a low delay and high traffic intensity. Improved a queue management scheme to be enhanced to offer a drop probability is desired necessarily. This paper considers multiple random early detection with differential drop probability which is a slightly modified version of the Multiple-RED(Random Early Detection) model, to get the performance of the best suited, we analyzes its main control parameters (max_th, min_th, max_p) for achieving the proportional loss differentiation (PLD) model, and gives their setting guidance from the analytic approach. we propose Dynamic-multiple queue management scheme based on differential drop probability, called Dynamic-VQSDDP(Variable Queue State Differential Drop Probability)T, is proposed to overcome M-RED's shortcoming as well as supports static max_p parameter setting values for relative and each class proportional loss differentiation. M-RED is static according to the situation of the network traffic, Network environment is very dynamic situation. Therefore max_p parameter values needs to modify too to the constantly and dynamic. The verification of the guidance is shown with figuring out loss probability using a proposed algorithm under dynamic offered load and is also selection problem of optimal values of parameters for high traffic intensity and show that Dynamic-VQSDDP has the better performance in terms of packet drop rate. We also demonstrated using an ns-2 network simulation.

• 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.