• 한국통신학회논문지
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• 제19권1호
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• pp.35-46
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• 1994

본 논문에서는 주파수 비선택적 라이시안 페이딩 채널에서 DS/SS 초기 동기 시스템의 성능을 분석 하였다. 채널 파라메터로는 페이딩된 신호 성분대 원하는 신호 성분의 전력비 . 연속적으로 균일하게 페이딩된 칩수 k, 연속된 k 칩간의 상관관계를 이용하였다. 이에 따른 검출기에서의 오보확률과 검출확률을 유도하였고 이를 초기 동기 시스템의 상태 천이도에 적용하여 시스템의 평균 패킷 손실 확률을 구하였다. 분석 결과로부터 페이딩 체널에서 직렬 탐색 알고리즘을 사용한 DS/SS 초기 동기 시스템의 성능의 사기상관 사이드로브에 의하여 패킷 손실이 증가 하는 대역이 있었으며, 연속적으로 균일한 페이딩을 갖는 칩수 k가 증가 합에 따라 패킷 손실 확률이 감소 하였다.

• 정보보호학회논문지
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• 제23권5호
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• pp.957-962
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• 2013

본 논문은 송수신 데이터의 보안을 위한 향상된 2차원 FEC기법의 성능에 관하여 제안한다. 이를 위해 기존 2차원 FEC 기법의 전체 패킷 손실률의 확률을 분석하고 2차원 FEC 기법의 문제점을 분석해서 길버트 모델링을 이용하여 개선된 2차원 FEC기법을 수학적으로 모델링하고, 전체 패킷 손실을 줄일 수 있는 향상된 2차원 FEC기법을 연구하였다.

• 한국통신학회논문지
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• 제30권4A호
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• pp.259-270
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• 2005

본 논문에서는 비대칭적이고 불균질적인 트래픽이 혼재하는 멀티미디어 서비스 환경에서 시분할 듀플렉싱을 사용하는 시분할-코드분할 다중 접속(TD-CDMA) 시스템의 직교 코드와 시간의 2차원적인 자원을 효율적으로 운용하는 방법을 수학적 모델링을 통해서 알아본다. 호-계층에서는 상/하향 트래픽 부하를 2차원 벡터로 나타내어 대기 이론을 기반으로 하여 호손율을 구하며, 최소의 호손율을 보이는 최적의 스윗칭-포인트를 찾는다. 패킷-계층에서는 서킷호와 패킷호로 구분하여 대기 중인 패킷과 서비스 중인 서킷호를 2차원의 상태로 나타내어 패킷 손실율을 구한다. 또한 일정 수준 이상의 서비스 품질을 위해 요구되는 버퍼 크기를 알아본다.

• 대한산업공학회지
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• 제28권2호
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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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• 제6권3호
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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.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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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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• 제2권1호
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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.

• 한국통신학회논문지
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• 제28권4B호
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• pp.288-297
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• 2003

TCP Reno 는 하나의 윈도우 내에서 다수 개의 패킷 손실이 발생하는 경우 손실된 패킷들을 효율적으로 복구하는 것이 불가능한 문제점을 가지과 있다. 이 문제점을 개선하기 위해서 설계된 TCP New-Reno 는 부분 스인 패킷(partial acknowledgement)를 통해 fast recovery를 연장함으로써 다수 개의 패킷 손실이 발생하더라도 이들을 재전송에 의해 복구하는 것이 가능하다. 그러나 TCP New-Reno 역시 재전송 패킷들이 다시 손질되는 경우 불가피한 RTO(Retransmission Timeout)가 발생한다는 문제점을 가지고 있다. 이런 문제점을 개선하기 위해서 중복 승인 패킷 수를 근거로 재전송 패킷 손실을 감지할 수 있는 DAC(Duplicate Acknowledgement Counting) 알고리듬을 제안한다. TCP Reno. TCP New-Reno 그리고 DAC를 사용하는 경우에 대해 손실 복구 과정을 정확하게 모델링하고 손실된 패킷이 복구되기 위한 조건들을 유도한다. 임의 패킷 손실 확률에 대한 손실 복구 확률을 수학적으로 계산하고 이를 통해 DAC가 TCP New-Reno 의 손실 복구 기능을 향상시킬 수 있다는 것을 보인다.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2008년도 추계학술발표대회
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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.

• 대한수학회논문집
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• 제14권2호
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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.