• Journal of Communications and Networks
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• v.5 no.1
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• pp.82-89
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• 2003

In this paper, an efficient threshold-based filtering (TF) buffer management scheme is proposed. The TF is capable of minimizing the overall loss performance and improving the fairness of buffer usage in a shared buffer packet switch. The TF consists of two mechanisms. One mechanism is to classify the output ports as sctive or inactive by comparing their queue lengths with a dedicated buffer allocation factor. The other mechanism is to filter the arrival packets of inactive output ports when the total queue length exceeds a threshold value. A theoretical queuing model of TF is formulated and resolved for the overall packet loss probability. Computer simulations are used to compare the overall loss performance of TF, dynamic threshold (DT), static threshold (ST) and pushout (PO). We find that TF scheme is more robust against dynamic traffic variations than DT and ST. Also, although the over-all loss performance between TF and PO are close to each other, the implementation of TF is much simpler than the PO.

• Journal of Internet Computing and Services
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• v.3 no.6
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• pp.43-51
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• 2002

In this paper, we study the new packet dropping scheme using an active queue management algorithm. Active queue management mechanisms differ from the traditional drop tail mechanism in that in a drop tail queue packets are dropped when the buffer overflows, while in active queue management mechanisms, packets may be dropped early before congestion occurs, However, it still incurs high packet loss ratio when the buffer size is not large enough, By detecting congestion and notifying only a randomly selected fraction of connection, RED causes to the global synchronization and fairness problem. And also, it is the biggest problem that the network traffic characteristics need to be known in order to find the optimum average queue length, We propose a new efficient packet dropping method based on the active queue management for congestion control. The proposed scheme uses the per-flow rate and fair share rate estimates. To this end, we present the estimation algorithm to compute the flow arrival rate and the link fair rate, We shows the proposed method improves the network performance because the traffic generated can not cause rapid fluctuations in queue lengths which result in packet loss

• ETRI Journal
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• v.31 no.1
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• pp.86-88
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• 2009

In this letter, we consider an m-node tandem queue (queues in series) with a Poisson arrival process and either deterministic or non-overlapping service times. With the assumption that each node has a finite buffer except for the first node, we show the non-increasing convex property of stationary waiting time with respect to the finite buffer capacities. We apply it to an optimization problem which determines the smallest buffer capacities subject to probabilistic constraints on stationary waiting times.

• ETRI Journal
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• v.29 no.5
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• pp.685-687
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• 2007

In this letter, we propose a new online buffer management algorithm to simultaneously provide diverse multimedia traffic services and enhance network performance. Our online approach exhibits dynamic adaptability and responsiveness to the current traffic conditions in multimedia networks. This approach can provide high buffer utilization and thereby improve packet loss performance at the time of congestion.

• Journal of applied mathematics & informatics
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• v.7 no.3
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• pp.861-873
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• 2000

Rate-based flow control plays an important role for efficient traffic management of Available Bit Rate(ABR) services. We deal with the problem of the buffer dimension for rate-based ABR control. In this paper, we analyze the Allowed Cell Rate(ACR) of a source and the queue size in a steady state. First, we investigate the effect of the number of active sources on the behavior of the ACR and the maximum queue size. Reflecting the effect of this real scenes, we determine the optimal buffer size and buffer threshold. Furthermore, our analytic results are compared with the case when the effect of the number of active sources is disregarded.

• The Journal of the Korea Contents Association
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• v.10 no.3
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• pp.80-91
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• 2010

Due to the convergence of broadcasting and communications, IPTV services are spotlighted as the that next-generation multimedia services. IPTV services should have functionality such as unlimited channel capacity, extension of media, QoS awareness and are required increasing traffic and quality control technology to adapt the attributes of IPTV service. Consequently, flow based quality control techniques are needed. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED(Random Early Detection) and modified RED algorithms have proposed. However, these algorithms have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the Flow based AQM(Active Queue Management) algorithm for the multimedia services that request various QoS requirements. The proposed algorithm can converge the packet loss ratio to the target packet loss ratio of required QoS requirements. And we present a performance evaluation by the simulations using the ns-2.

• ETRI Journal
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• v.43 no.3
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• pp.447-458
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• 2021

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• Journal of the Korean Operations Research and Management Science Society
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• v.39 no.3
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• pp.1-5
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• 2014

We study the paper Zhernovyi and Zhernovyi [Zhernovyi, K.Y. and Y.V. Zhernovyi, "An $M^{\Theta}/G/1/m$ system with two-threshold hysteresis strategy of service intensity switching," Journal of Communications and Electronics, Vol.12, No.2(2012), pp.127-140]. In the paper, authors used the Korolyuk potential method to obtain the stationary queue length distribution. Instead, our note makes an attempt to apply the most frequently used methods : the embedded Markov chain and the supplementary variable method. We derive the queue length distribution at a customer's departure epoch and then at an arbitrary epoch.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12C
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• pp.261-267
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• 2001

This paper presents the architecture of multiple-queue manager for input-queued switch, which has arbitration latency, and the design of the chip. The proposed architecture of multiple-queue manager provides wire-speed routing with a pipelined buffer management, and the tolerance of requests and grants data transmission latency between the input queue manager and central arbiter using a new request control method, which is based on a high-speed shifter. The multiple-input-queue manager has been implemented in a field programmable gate array chip, which provides OC-48c port speed. It enhances the maximum throughput of the input queuing switch up to 98.6% with 128-cell shared input buffer in 16$\times$16 switch size.