• 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

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.579-603
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• 2018

In this paper we propose a QoS-enhanced intelligent stochastic optimal fair real-time packet scheduler, QUEST, for 4G LTE traffic in routers. The objective of this research is to maximize the system QoS subject to the constraint that the processor utilization is kept nearly at 100 percent. The QUEST has following unique advantages. First, it solves the challenging problem of starvation for low priority process - buffered streaming video and TCP based; second, it solves the major bottleneck of the scheduler Earliest Deadline First's failure at heavy loads. Finally, QUEST offers the benefit of arbitrarily pre-programming the process utilization ratio.Three classes of multimedia 4G LTE QCI traffic, conversational voice, live streaming video, buffered streaming video and TCP based applications have been considered. We analyse two most important QoS metrics, packet loss rate (PLR) and mean waiting time. All claims are supported by discrete event and Monte Carlo simulations. The simulation results show that the QUEST scheduler outperforms current state-of-the-art benchmark schedulers. The proposed scheduler offers 37 percent improvement in PLR and 23 percent improvement in mean waiting time over the best competing current scheduler Accuracy-aware EDF.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.12
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• pp.54-62
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• 2007

Mobile IPv6 is the representative standard protocol proposed to offer mobility of mobile hosts, but it has drawbacks such as high handover latency and packet loss during handover. To complement these drawbacks, several protocols have been proposed and they can support faster handover than the Mobile IPv6, but there still exists inevitable packet loss during handover in those protocols. In this paper, we propose a new handover procedure of mobile host who has a dual link for minimizing packet loss rate during micro handover to improve mobility capability of Cellular Ethernet system that supports mobility at layer 2. And we derive the superiority of proposed scheme by comparing in terms of packet loss rate, handover latency and throughput of proposed scheme with those of Hierarchical Mobile IPv6 which is the representative IP-layer micro mobility protocol by simulation.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.11
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• pp.2124-2126
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• 2008

In the wireless communication systems, the demand of multimedia services is also increased. Unlike typical data packets, realtime service such as VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that be able to allocate resources to the different kinds of services such as VoIP and data packet. The proposed algorithm considers both time delay and channel condition toe determine the priority. Simulation results show that the proposed algorithm works more efficiently than the conventional algorithms.

• ETRI Journal
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• v.33 no.1
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• pp.136-139
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• 2011

A modular upgrade design for packet transport switch nodes is presented where packet loss is dramatically reduced by intra-module and inter-module buffer sharing. This modular design offers significant cost and power reduction in a high-data-rate system where buffers are highly costly and power-greedy.

• Journal of Korea Multimedia Society
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• v.9 no.2
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• pp.197-207
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• 2006

When the latency of a flow in a fair packet scheduler, which is determined by its rate, violates its required delay bound, the scheduler should reduce the latency with even raising the rate being reserved for the flow. The excessively reserved rate win enforce some outgoing link bandwidth be lost. This loss can not be, unfortunately, evaluated by the three metrics of latency, fairness and implementation complexity used in previous works. This paper is aimed to first introduce the metric of bandwidth utilization to investigate the bandwidth loss in a scheduler and then evaluate the timestamp based schedulers in terms of the bandwidth and payload utilizations. The results show that the bandwidth utilization increases with loosing the required delay bound and, in particular, schedulers with the latency property of WFQ have much better payload utilization by up to 50% than that in the SCFQ one.

• 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.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.45-50
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• 2008

Many applications of sensor network require connection to the Internet. The transmission protocol of traditional sensor network was designed within the sensor network itself. However, based on 6LoWPAN which can be accessed using IPv6, direct connection is possible between the sensor network and the TCP/IP network outside. Transmission of data in applications of sensor network falls into two main categories. One is a small packet that is periodically produced such as packet related to temperature and humidity. The other is a relatively large packet that brings about network overheads such as images. We investigated the conformance test and pros and cons of application data over the transmission protocol of Zigbee and 6LoWPAN. As a result, both Zigbee and 6LoWPAN have shown low rate of loss for periodic data and have in creased reliability of data transfer. When transmitting streaming image data, both ACK, non ACK mode of Zigbee and UDP of 6LoWPAN minimized transmission time but suffered the consequences of high packet loss. Even though TCP of 6LoWPAN required a long transmission time, we were able to confirm that no loss has occurred.