• Journal of KIISE:Information Networking
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• v.36 no.3
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• pp.251-255
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• 2009

Network coding is a promising technology that increases the system throughput via reducing the number of transmission for a packet delivered from the source node to the destination node. Nevertheless, it suffers from the metrics of end-to-end delay. Network Coding scheme takes more processing delay which occurs as coding node encodes (XOR) a certain number of packets that relayed by the coding node, and more queuing delay which occurs as a packet waits for other packets to be encoded with. Therefore, in this paper, we analyze the dependency of the queuing delay to the arrival rate of each packet. In addition, we analyze and compare the delay in DCF without XOR and DCF with XOR under dynamic traffic.

• ETRI Journal
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• v.28 no.1
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• pp.31-44
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• 2006

In a packet switching network, congestion is unavoidable and affects the quality of real-time traffic with such problems as delay and packet loss. Packet fair queuing (PFQ) algorithms are well-known solutions for quality-of-service (QoS) guarantee by packet scheduling. Our approach is different from previous algorithms in that it uses hardware time achieved by sampling a counter triggered by a periodic clock signal. This clock signal can be provided to all the modules of a routing system to get synchronization. In this architecture, a variant of the PFQ algorithm, called digitized delay queuing (DDQ), can be distributed on many line interface modules. We derive the delay bounds in a single processor system and in a distributed architecture. The definition of traffic contribution improves the simplicity of the mathematical models. The effect of different time between modules in a distributed architecture is the key idea for understanding the delay behavior of a routing system. The number of bins required for the DDQ algorithm is also derived to make the system configuration clear. The analytical models developed in this paper form the basis of improvement and application to a combined input and output queuing (CIOQ) router architecture for a higher speed QoS network.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.774-779
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• 2012

We design and implement a RTT (Round Trip Time) based TCP (Transmission Control Protocol) for USN (Ubiquitous Sensor Network). We adopt a basic update algorithm for window size from FAST TCP that uses the queuing delay at link as the congestion measure. The designed TCP estimates the queuing delay at link from the measured RTT in the network layer, and updates the window size based on the estimated queuing delay. The designed TCP allows to utilize the full capacity of USN links and avoids the waste of the given link capacity that is common without the flow control in the transport layer. The experiment results show that the window size of the sender converges within a small range of variations without any packet loss, and verify the stability and performance of the designed TCP.

• The KIPS Transactions:PartC
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• v.15C no.6
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• pp.523-530
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• 2008

In this paper, we analyze the mean queuing delay of selective-repeat automatic repeat request (SR-ARQ) protocol with the finite retransmission persistence. The retransmission persistence means the willingness of the protocol to retransmit a lost (or corrupted) packet to ensure reliable packet delivery across a lossy link. According to the retransmission persistence, SR-ARQ protocols have a different performance in terms of both packet delay and link reliability. So far, however, there is no serious study in the effect of the retransmission persistence on the SR-ARQ performance. We present a simple M/G/1 queuing model for the SR-ARQ protocol with the finite retransmission persistence by using the ideal SR-ARQ approximation. The mean queuing delay is obtained from the queuing model and verified its accuracy through the simulation results using the OPNET simulator. Both the analytical predictions and simulation results clearly show the effect of retransmission persistence on the queuing delay of the SR-ARQ protocol in various network conditions: packet loss rate and traffic condition over a wireless link.

• Journal of Korea Multimedia Society
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• v.10 no.1
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• pp.83-93
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• 2007

WFQ (Weighted Fair Queuing) is the most popular fair queuing algorithm, but it had the inherent drawback of a poet bandwidth utilization, particularly under the traffic requiring a low rate but tight delay bound such as internet phone. It was recently identified that the poor utilization is mainly due to the non-optimized latency of a flow and then LOFQ(Latency-Optimized Fair Queuing) to overcome the drawback was introduced. In this paper, we improve the performance of LOFQ by introducing an occupied resource optimization function and reduce the implementation complexity of recursive resource transformation by revising the transformation scheme. We also prove the superiority of LOFQ over WFQ in terms of utilization. The simulation result shows that the improved LOFQ provides $20{\sim}30%$ higher utilization than that in the legacy LOFQ.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7A
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• pp.717-724
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• 2006

In order to maximize the throughput and provide the fairness between users in MIMO-OFDM system, FATM(fairness-aware throughput maximization) scheduling algorithm was proposed. In this paper, a QoS-aware scheduling algorithms for MINO-OFDM system are proposed, each of which is based on FATM. These scheduling algorithms aim to satisfy the different service requirements of various service classes. Three proposed QoS scheduling algorithms called SPQ (Strict Priority Queueing), DCBQ (Delay Constraint Based Queuing), HDCBQ (Hybrid Delay Constraint Based Queuing) are compared through computer simulations. It is shown that HDCBQ algorithm outperforms other algorithms in satisfying different requirements of various service classes.

• The KIPS Transactions:PartC
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• v.14C no.2
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• pp.155-162
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• 2007

WFQ (Weighted Fair Queuing) is the most popular fair queuing algorithm, but it had the inherent drawback of a poor bandwidth utilization, particularly under the traffic requiring a low rate but tight delay bound such as internet phone, It was recently identified that the poor utilization is mainly due to non optimized latency of a flow and then LOFQ(Latency-Optimized Fair Queuing) to overcome the drawback was introduced, LOFQ was also improved through introducing an occupied resource optimization function and the implementation complexity of recursive resource transformation was reduced with revising the transformation scheme. However, the performance of LOFQ has been evaluated by means of simulation, so that there are some difficulties in evaluating the performance in the terms of the accuracy and evaluation time, In this paper, we develop how to analytically compute the bandwidth utilization in LOFQ.

• ETRI Journal
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• v.25 no.6
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• pp.475-488
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• 2003

In this paper, we propose an efficient and simple fair queuing algorithm, called new starting potential fair queuing (NSPFQ), which has O(1) complexity for virtual time computation and also has good delay and fairness properties. NSPFQ introduces a simpler virtual time recalibration method as it follows a rate-proportional property. The NSPFQ algorithm recalibrates the system virtual time to the minimum virtual start time among all possible virtual start times for head-of-line packets in backlogged sessions. Through analysis and simulation, we show that the proposed algorithm has good delay and fairness properties. We also propose a hardware implementation framework for the scheduling algorithm.

• Journal of Information Processing Systems
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• v.19 no.6
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• pp.817-829
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• 2023

Variable wireless channel is a big challenge for real-time video applications, and the rate adaptation of realtime video streaming becomes a hot topic. Intra-video coding is important for high-quality video communication and industrial video applications. In this paper, we proposed a novel adaptive scheme for real-time video transmission with intra-only coding over a wireless network. The key idea of this scheme is to estimate the instantaneous remaining capacity of the network to adjust the quality of the next several video frames, which not only can keep low queuing delay and ensure video quality, but also can respond to bandwidth changes quickly. We compare our scheme with three different schemes in the video transmission system. The experimental results show that our scheme has higher bandwidth utilization and faster bandwidth change response, while maintaining low queuing delay.

• ETRI Journal
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• v.34 no.6
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• pp.807-815
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• 2012

In this paper, we propose a scheme for partially dynamic lane control for energy saving in multilane-based high-speed Ethernet. In this scheme, among the given transmission lanes, at least one lane is always operating, and the remaining lanes are dynamically activated to alleviate the network performance in terms of queuing delay and packet loss in the range of acceptance. The number of active lanes is determined by the decision algorithm based on the information regarding traffic and queue status. The reconciliation sublayer adjusts the transmission lane with the updated number of lanes received from the algorithm, which guarantees no processing delay in the media access control layer, no overhead, and minimal delay of the exchanging control frames. The proposed scheme is simulated in terms of queuing delay, packet loss rate, lane changes, and energy saving using an OPNET simulator. Our results indicate that energy savings of around 55% (or, when the offered load is less than 0.25, a significant additional savings of up to 75%) can be obtained with a queuing delay of less than 1 ms, a packet loss of less than $10^{-4}$, and a control packet exchange time of less than $0.5{\mu}s$ in random traffic.