• Transactions of the Korean Society of Mechanical Engineers B
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• v.33 no.12
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• pp.977-982
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• 2009

Direct numerical simulation (DNS) of a turbulent boundary layer with moderate Reynolds number was performed to scrutinize streamwise-coherence of hairpin packet motions. The Reynolds number based on the momentum thickness (${\theta}_{in}$) and free-stream velocity (U${\infty}$) was varied in the range $Re_{\theta}$=1410${\sim}$2540 which was higher than the previous numerical simulations in the turbulent boundary layer. In order to include the groups of hairpin packets existing in the outer layer, large computational domain was used (more than 50${\delta}_o$, where ${\theta}_o$ is the boundary layer thickness at the inlet in the streamwise domain). Characteristics of packet motions were investigated by using instantaneous flow fields, two-point correlation and conditional average flow fields in xy-plane. The present results showed that a train of hairpin packet motions was propagating coherently along the downstream and these structures induced the very large-scale motions in the turbulent boundary layer.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.12
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• pp.4856-4871
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• 2015

Backpressure based scheduling has been considered as a promising technique for improving the throughput of a wide range of communication networks. However, this scheduling technique has not been well studied for heterogeneous wireless networks. In this paper, we propose a virtual-queue based backpressure scheduling (VQB) algorithm for heterogeneous multi-hop wireless networks. The VQB algorithm introduces a simple virtual queue for each flow at a node for backpressure scheduling, whose length depends on the cache size of the node. When calculating flow weights and making scheduling decisions, the length of a virtual queue is used instead of the length of a real queue. We theoretically prove that VQB is throughput-optimal. Simulation results show that the VQB algorithm significantly outperforms a classical backpressure scheduling algorithm in heterogeneous multi-hop wireless networks in terms of the packet delivery ratio, packet delivery time, and average sum of the queue lengths of all nodes per timeslot.

• 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.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1483-1486
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• 2002

Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• Journal of Institute of Control, Robotics and Systems
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• v.9 no.10
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• pp.789-797
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• 2003

An implementation scheme and some improvements are proposed to adopt public-licensed operating system, Linux and de-facto world-wide network standard, TCP/IP into the field of behavior-based autonomous mobile robots. To demonstrate the needs of scheme and the improvement, an analysis is performed on a server/client communication problem with real time Linux previously proposed, and another analysis is also performed on interactions among TCP/IP communications and the performance of Linux system using them. Implementation of behavior-based control architecture on real time Linux is proposed firstly. Revised task-scheduling schemes are proposed that can enhance the performance of server/client communication among local tasks on a Linux platform. A new method of TCP/IP packet flow handling is proposed that prioritizes TCP/IP software interrupts with aperiodic server mechanism as well. To evaluate the implementation scheme and the proposed improvements, performance enhancements are shown through some simulations.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7B
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• pp.642-651
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• 2006

In ad-hoc network, the most of existing packet scheduling schemes provides throughput-based fairness. To provide throughput-based fairness, it basically supposes that the channel capacity is fixed. But, the supposing that the channel capacity is fixed is not appropriate because IEEE 802.11b and 802.11g which are normally used for organizing ad-hoc network can provide various data rate according to channel conditions. So, we define time-based fairness for each flow to consider multi-rate and suggest the MRADPS reaching the defined time-based fairness. Simulation result shows that MRADPS improves the total network throughput in ad-hoc network with providing time-based fairness to each flow.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.203-210
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two design schemes in terms of the reference time used in calculating the timestamp of arriving packet: Finish-time Design (FD) and Start-time Design (SD) schemes. Since the former can adjust the latency of a flow with raising the flow's reserved rate, it has been applied to a router for the guaranteed service of the IETF (Internet Engineering Task Force) IntServ model. However, the FD scheme may incur severe bandwidth loss for traffic flows requiring low-rate but strong delay bound such as internet phone. In order to verify the usefulness of the SD scheme based router for the IETF guaranteed service, this paper analyzes and compares two design schemes in terms of bandwidth and payload utilizations. It is analytically proved that the SD scheme is better bandwidth utilization than the FD one, and the simulation result shows that the SD scheme gives better payload utilization by up to 20%.

• Journal of Korea Multimedia Society
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• v.9 no.3
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• pp.323-332
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two schemes in terms of the reference time used in calculating the timestamp of arriving packet; the Finish-Time (FT) and Start-Time (ST) schemes. The FT scheme, used in most schedulers, that has the property of an inversely rate-proportional latency is suitable to support various delay bounds because it can adjust the latency of a flow with raising the flow's reserved rate. However, the scheme may incur some bandwidth loss due to excess rate reservation. Meanwhile, although the ST scheme does not suffer from the bandwidth loss, it is hard to support multiple delay bounds because of its latency property relying on the number of flows. This paper is devoted to propose a ST scheme based scheduler to effectively support multiple delay bounds and analyze its performance comparing to the FT scheme based scheduler. The comparison results show that the proposed scheduler gives better utilization by up to 50%.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.4
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• pp.552-558
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• 2002

Differentiated services (DiffServ) architecture defines a new framework for the support of quality of service (QoS) in IP-based networks. RIO has received the most attention among all the active queue management mechanisms proposed for assured service in DiffServ environment. The paper proposed the enhanced RIO which could alleviate the impact of flow's packet sire on the realized throughput. The simulation results indicate that this mechanism, when combined with TSW as traffic conditioner, provide better throughput assurance and fair distribution of excess bandwidth independent of packet size of flows in case of well-provisioned network environment.