• 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.

• ETRI Journal
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• 제29권1호
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• pp.18-26
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• 2007

This paper presents an improved redundant picture coding method that efficiently enhances the error resiliency of H.264. The proposed method applies polyphase downsampling to residual blocks obtained from inter prediction and selectively encodes the rearranged residual blocks in the redundant picture coding process. Moreover, a spatial-temporal sample construction method is developed for the redundant coded picture, which further improves the reconstructed picture quality in error prone environments. Simulations based on JM11.0 were run to verify the proposed method on different test sequences in various error prone environments with average packet loss rates of 3%, 5%, 10%, and 20%. Results of the simulations show that the presented method significantly improves the robustness of H.264 to packet loss by 1.6 dB PSNR on average over the conventional redundant picture coding method.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2008년도 정보통신설비 학술대회
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• pp.579-583
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• 2008

본 논문은 이더넷이 비동기식으로 전송됨으로 인해, 동일한 전송속도를 가진 장비라 할지라도 링크가 제공하는 명목상의 최대속도로 전송될 경우, 상호 접속한 장비간의 전송클럭 차이로 인해 프레임의 손실이 발생한다. 본 논문에서는 PHY에서 복원된 수신 클럭과 송신 클럭의 차이를 비교하고 동시에 프레임버퍼에 쌓인 큐를 참조하여 프레임 손실이 방생할 수 있는 임계치를 넘어설 경우, 전송프레임의 프리엠블 길이를 조정함으로써 이더넷 장비에서 전송클럭의 차이로 인한 프레임손실을 줄일 수 있음을 확인하였다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제11권8호
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• pp.3809-3822
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• 2017

Wireless Sensor Networks hold the limelight because of significant potential for distributed sensing of large geographical areas. The radio duty cycling mechanism that turns off the radio periodically is necessary for the energy conservation, but it deteriorates the network congestion when the traffic load is high, which increases the packet loss and the delay too. Although many papers for WSNs have tried to mitigate network congestion, none of them has mentioned the congestion problem caused by the radio duty cycling of MAC protocols. In this paper, we present a simple and efficient congestion control technique that operates on the radio duty cycling MAC protocol. It detects the congestion by checking the current queue size. If it detects the congestion, it extends the network capacity by adding supplementary wakeup times. Simulation results show that our proposed scheme highly reduces the packet loss and the delay.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2010년도 추계학술발표대회
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• pp.1482-1484
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• 2010

A new TCP protocol can succeed for large bandwidth delay product when it meets network bandwidth utilization efficiency and fair sharing. We introduce a novel congestion control algorithm which employs queueing delay information in order to calculate the amount of congestion window increment in increase phase, and reduces congestion window to optimal estimated bound as packet loss occurs. Combination of such methods guarantees that the proposal utilizes fully network bandwidth, recovers quickly from packet loss in wireless link, and preserves fairness for competing flows mixed short RTT and long RTT. Our simulations show that features of the proposed TCP meet the desired requirements.

• 대한전자공학회논문지TC
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• 제47권1호
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• pp.69-76
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• 2010

FTTH(Fiber To The Home)가입자 망에 있어 패킷단위의 정보처리를 위한 3 비트, 4 비트의 전광학 헤더인식 과정을 설명하고 실험을 성공적으로 수행하였다. 패킷의 목적지를 나타내는 3, 4 비트 헤더정보에 따른 하나의 신호가 각각 8개, 16개의 타임슬롯 상의 지정된 한 곳에 나열되었다. RN(remote node)에서의 self-routing을 위해 제안된 헤더인식 기술은 TDM 방식의 원리를 응용하여 스위치와 지연라인만을 사용함으로써 매우 간단한 회로를 구성하여 높은 신뢰도와 낮은 비용으로 가입자 망을 구축할 수 있다. 또 RN으로부터 각 가입자의 독립 전송선로를 확립함으로써 TDM-PON 방식의 최대 취약점이었던 광 신호의 감쇄와 보안성의 문제를 해결할 수 있다.

• 한국인터넷방송통신학회논문지
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• 제11권6호
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• pp.81-86
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• 2011

네트워크 코딩위에서의 TCP 성능문제를 해결하기 위해 제안된 가장 잘 알려진 방식에서는 네트워크 코딩 계층이 혁신적인(innovative) 선형 결합을 수신하면 새로 디코드 된 패킷이 없다 하더라도 승인을 보낸다. 이 방식은 매우 효과적이지만 실제로 구현될 때에는 패킷 헤더 크기의 제한으로 인해 코딩 윈도우 크기를 제한해야 하므로 패킷 순서 바뀜 현상이 많이 발생할 때 성능이 저하될 수 있다. 본 연구에서는 네트워크 코딩 환경에서도 패킷 순서 바뀜 현상과 관련된 문제를 다루기 위해서는 중복승인을 사용하지 않고 타이머에 의존하는 TCP가 필요함을 주장한다. 또한 이러한 TCP를 위한 새로운 네트워크 코딩계층을 제안한다. 모의실험 결과는 두 개의 경로를 사용하는 라우팅 환경에서 패킷 순서가 바뀌어 수신되는 패턴에 따라 최대 19%까지 성능이 개선됨을 보여준다.

• Journal of Communications and Networks
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• 제4권4호
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• pp.266-281
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• 2002

The General Packet Radio Service (GPRS) is being deployed by GSM network operators world-wide, and promises to provide users with “always-on” data access at bandwidths comparable to that of conventional fixed-wire telephone modems. However, many users have found the reality to be rather different, experiencing very disappointing performance when, for example, browsing the web over GPRS. In this paper, we examine the causes, and show how unfortunate interactions between the GPRS link characteristics and TCP/IP protocols lead to poor performance. A performance characterization of the GPRS link-layer is presented, determined through extensive measurements taken over production networks. We present measurements of packet loss rates, bandwidth availability, link stability, and round-trip time. The effect these characteristics have on TCP behavior are examined, demonstrating how they can result in poor link utilization, excessive packet queueing, and slow recovery from packet losses. Further, we show that the HTTP protocol can compound these issues, leading to dire WWW performance. We go on to show how the use of a transparent proxy interposed near the wired-wireless border can be used to alleviate many of these performance issues without requiring changes to either client or server end systems.

• 정보처리학회논문지C
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• 제10C권7호
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• pp.937-942
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• 2003

본 논문은 일본 기가비트 네트워크 테스트베드인 JNG상에서 대표적인 스트림형 트래픽인 DV와 MPEG2에 대한 최종 사용자간에 실측을 통하여 초고속 네트워크 상에서 정상 부하의 변화에 따른 각 스트림 트래픽의 전송 지연과 패킷 손실에 대하여 고찰한 것이다. 이 결과에 의하면 50Mbps의 IP over ATM상에서 DV와 MPEG2 스트림을 전송할 때, 부가된 정상 부하의 패킷 크리(512바이트 이상)에 무관하게 부하율이 네트워크 대역폭의약 95％에 달하는 시점에서 패킷 손실이 다량 발생하고 전송 지연도 급증하였다. 또한 최종 사용자 시스템상의 버퍼 크기와 송수신 버퍼 수는 전송 지연과 패킷 손실에 영향을 미치지 않음을 확인하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 심포지엄 논문집 정보 및 제어부문
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• pp.215-217
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• 2005

There are two types of packet loss probabilities used in both the network layer and the physical layer within the wireless transmitter such as a queueing discard probability and transmission loss probability. We analyze these loss performances in order to guarantee Quality of Service (QoS) which is the basic of the future network. The queuing loss probability is caused by a maximum allowable delay time and the transmission loss probability is caused by a wireless channel error. These two types of packet loss probabilities are not easily analyzed due to recursive feedback which, originates as a result at a queueing delay and a number of retransmission attempts. We consider a wireless transmitter to a M/D/1 queueing model. We configurate the model to have a finite-size FIFO buffer in order to analyze the real-time traffic streams. Then we present the approaches used for evaluating the loss probabilities of this M/D/1/K queueing model. To analyze the two types of probabilities which have mutual feedbacks with each other, we drive the solutions recursively. The validity and accuracy of the analysis are confirmed by the computer simulation. From the following solutions, we suggest a minimum of 'a Maximum Allowable Delay Time' for real-time traffic in order to initially guarantee the QoS. Finally, we analyze the required service rate for each type utilizing real-time traffic and we apply our valuable analysis to a N-user's wireless network in order to get the fundamental information (types of supportable real-type traffics, types of supportable QoS, supportable maximum number of users) for network design.