• Journal of Korea Multimedia Society
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• v.20 no.2
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• pp.254-260
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• 2017

Data packets from sensor nodes scattered over measuring fields are generally forwarding to the sink node, which may be connected to the wired networks, in a wireless sensor network. So many data packets are gathered near the sink node, resulting in significant data packet collisions and severe transmission latency. In an event detection application such as object tracking and military, bursty data is generated when an event occurs. So many data packet should be transmitted in a limited time to the sink node. In this paper, we present a delay efficient and bursty traffic friendly MAC protocol called DEBF-MAC protocol for wireless sensor networks. The DEBF-MAC uses a slot-reserved mechanism and sleep period control method to send multiple data packets efficiently in an operational cycle time. Our simulation results show that DEBF-MAC outperforms DW-MAC and SR-MAC in terms of energy consumption and transmission delay.

• ETRI Journal
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• v.28 no.5
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• pp.593-606
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• 2006

Thanks to the great possibilities of providing different types of telecommunication traffic to a large geographical area, satellite networks are expected to be an essential component of the next-generation internet. As a result, issues concerning the designing and testing of efficient connection-admission-control (CAC) strategies in order to increase the quality of service (QoS) for multimedia traffic sources, are attractive and at the cutting edge of research. This paper investigates the potential strengths of a generic digital-video-broadcasting return-channel-via-satellite (DVB-RCS) system architecture, proposing a new CAC algorithm with the aim of efficiently managing real-time multimedia video sources, both with constant and high variable data rate transmission; moreover, the proposed admission strategy is compared with a well-known iterative CAC mainly designed for the managing of real-time bursty traffic sources in order to demonstrate that the new algorithm is also well suited for those traffic sources. Performance analysis shows that, both algorithms guarantee the agreed QoS to real-time bursty connections that are more sensitive to delay jitter; however, our proposed algorithm can also manage interactive real-time multimedia traffic sources in high load and mixed traffic conditions.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.159-162
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• 2005

Random Early Detection (RED), one of the most well-known Active Queue Management (AQM), has been designed to substitute Tail Drop and is nowadays widely implemented in commercially available routers. RED algorithm provides high throughput and low delay as well as a solution of global synchronization. However RED is sensitive to parameters setting, so the performance of RED, significantly depends on the fixed parameters. To solve this problem, the Adaptive RED (ARED) algorithm is suggested by S. Floyd. But, ARED also uses fixed parameters like target-queue length; it is hard to respond to bursty traffic actively. In this paper, we proposed AQM algorithm based on the variation of current queue length in order to improve adaptability about burst traffic. We measured performance of proposed algorithm through a throughput, marking-drop rate and bias phenomenon. In experimentation, we raised a packet throughput as reduced packet drop rate, and we confirmed to reduce a bias phenomenon about bursty traffic.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.1
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• pp.36-42
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• 1997

This paper proposes a high-speed contention resolution scheme featuring high flexibility to the bursty traffic for an input buffering ATM switching architecture and its hardware strategy. The scheme is based on the threshold on the occupancy of the input buffer. As the proposed scheme utilizes the threshold, it has high flexibility to the fluctuations of the input traffic. The hardware strategy for the proposed policy is provided with the aim of the simple structure that achieves the reduction of the signal path and the power consumption. The performance on the average buffer size of the proposed policy is performed and compared with the conventional schame under the bursty traffic through both the analysis based on the markov hain and the simulation. The relations among the parameters on the proposed policy is analyzed to improve the performance.

• Journal of information and communication convergence engineering
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• v.6 no.1
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• pp.19-23
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• 2008

The AQM (Active Queue Management) starts dropping packets earlier to notify traffic sources about the incipient stage of congestion. The AQM improves fairness between response flow (like TCP) and non-response flow (like UDP), and it can provide high throughput and link efficiency. In this paper, we suggest the QVARED (Queue Variation Adaptive RED) algorithm to respond to bursty traffic more actively. It is possible to provide more smoothness of average queue length and the maximum packet drop probability compared to RED and ARED (Adaptive RED). Therefore, it is highly adaptable to new congestion condition. Our simulation results show that the drop rate of QVARED is decreased by 80% and 40% compare to those of RED and ARED, respectively. This results in shorter end-to-end delay by decreasing the number of retransmitted packets. Also, the QVARED reduces a bias effect over 18% than that of drop-tail method; therefore packets are transmitted stably in the bursty traffic condition.

• Journal of the Korea Computer Industry Society
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• v.3 no.1
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• pp.19-26
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• 2002

This paper propose the queue structure of cell buffering in the output buffer of ATM switch for the traffic with the different QoS. The proposed queue structure can minimize the cell loss ratio of bursty traffic, maximize the queue utilization through the sharing of real-time queue and non-real-time queue. To evaluate the proposed queue structure, we compare the CLP and cell average delay of the proposed queue and fixed queue using the bursty traffic patterns.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.11
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• pp.2836-2849
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• 1996

Over ATM network, variable bit rate(or VBR) traffic is allowed. Control of VBR traffic is allowed. Control of VBR traffic becomes difficult if it is bursty. VBR video traffic becomes so much bursty during intra frame period that much cell loss would occur when satistical multiplexed in ATM swich. To aviod cellloss, extra communication resources should be allocated, which reduces the capability of an ATM channel. In this paper, we propose two methods which enable a channel limited in resources to serve more VBR MPEG video bit streams. Firstly, we could redue the bitrate fluction of a statiscally multiplexed bundle of VBR video bit streams by reducing the number of intra frames overlapped at the same frame period. This method can be used in ATM switch which controls multiple video sources. Secondary, in two layer enoding, statistical multiplexing gains can be icreased by letting peak bit rate durations of both layers not be overlapped. This results in more smooth traffic. The performance of proposed methods are demonstrated by a proposed calculation method of statistical multiplexing gains(or SMGs.) The proposed SMG is based on both delay and cell loss QoS requirements at the same time.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.7-13
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• 1997

This paper addresses traffic prediction issues on MPEG. A new adaptive traffic prediction scheme is proposed using MPEG picture characteristic that picture traffic depends on the coding mode of that picture, that is, I, P, and B mode. Our prediction scheme, which is based n picture decomposition (PD) and the cross-correlation of the different types of pictures, has better performance in predicting bursty MPEG traffic than that of the first-order autoregressive (AR) prediction scheme. Our simulation results show that the performance is further improved about 15% by utilizing the cross-correlations between pictures.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.477-480
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• 2003

Packet losses in the Internet can dramatically degrade quality of multimedia streams. Forward Error Correction (FEC) is one of the best methods that can protect data from packet erasures by means of sending additional redundant information. Proposed control algorithm provides the possibility of receiving real-time multimedia streams of given quality wifth minimal traffic overhead. The traffic optimization is reached by adjusting packet size as well as block code parameters. Calculations and simulation results show that for non-bursty network conditions traffic optimization can lead to more than 50% bandwidth reduction.