• 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.

• ETRI Journal
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• v.29 no.1
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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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• 2008.08a
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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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• v.11 no.8
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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.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.1
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• pp.69-76
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• 2010

We describe an all-optical circuit which recognizes the header information of packet-by-packet in the access networks based on FTTH. The circuit's operation is confirmed by an experiment in the recognition of 3 and 4 header bits. The output from the header recognition circuit appears in a signal assigned in the time axis according to the header information. The recognition circuit of header for self-routing has a very simple structure using only delay lines and switches. The circuit is expected that it can be constructed of the high reliability and the low cost. Also, the circuit can solve the problems of the power loss and private security which is the weak point of the TDM-PON method by being established a unique transmission line to each subscriber.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.81-86
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• 2011

In one of the most impacting schemes proposed to address the TCP throughput problem over network coding, the network coding layer sends an acknowledgement if an innovative linear combination is received, even when a new packet is not decoded. Although this scheme is very effective, its implementation requires a limit on the coding window size. This limitation causes low TCP throughput in the presence of packet reordering. We argue that a TCP variant detecting a packet loss relying only on timers is effective in dealing with the packet reordering problem in network coding environments as well. Also we propose a new network coding layer to support such a TCP variant. Simulation results for a 2-path environment show that our proposed scheme improves TCP throughput by 19%.

• Journal of Communications and Networks
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• v.4 no.4
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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.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• Proceedings of the KIEE Conference
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• 2005.05a
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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.