• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.1
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• pp.36-40
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• 2002

Routing algorithm uses metric to choose the route of Least cost to destination network, the best suited routing investigates all routes to the shortest destination among networks and is decided on the route given the minimum metric. This paper analyzed packet flow for setting up the best fitted path on the same network using RIP and EIGRP as the distance vector algorithm and measured the Link-efficiency.

• Journal of Korea Multimedia Society
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• v.18 no.2
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• pp.189-198
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• 2015

Despite the fact that traffic engineering techniques have been comprehensively utilized in the past to enhance the performance of communication networks, the distinctive characteristics of Software Defined Networking (SDN) demand new traffic engineering techniques for better traffic control and management. Considering the behavior of traffic, large flows normally carry out transfers of large blocks of data and are naturally packet latency insensitive. However, small flows are often latency-sensitive. Without intelligent traffic engineering, these small flows may be blocked in the same queue behind megabytes of file transfer traffic. So it is very important to identify large flows for different applications. In the scope of this paper, we present an approach to detect large flows in real-time without even a short delay. After the detection of large flows, the next problem is how to control these large flows effectively and prevent network jam. In order to address this issue, we propose an approach in which when the controller is enabled, the large flow is mitigated the moment it hits the predefined threshold value in the control application. This real-time detection, marking, and controlling of large flows will assure an optimize usage of an overall network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1683-1690
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• 2005

This study examines dynamic flow control method in UDP, analyzes packet loss which is frequently used element in measuring existing dynamic flow control and characteristics of round trip time, and proposes a new method of measurement, RSST. The algorithm that uses the proposed RSST enables accurate measurement of network status by considering both the currently measured network status and the past history of network status in controlling the transmission rate. For comparison study, a network status measurement software program that displays detailed information about volume of transmission generation of network status, and flow pattern of network was developed. The performance test shows that the proposed algorithm can better adjust to network condition in terms of low pack loss rate over existing algorithms.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.8
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• pp.3900-3916
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• 2019

The modern mobile devices are typically equipped with multiple network interfaces, e.g., 4G LTE, Wi-Fi, Bluetooth, but the current implementation of TCP can support only a single path at the same time. The Multipath TCP (MPTCP) leverages the multipath feature and provides (i) robust connection by utilizing another interface if the current connection is lost and (ii) higher throughput than single path TCP by simultaneously leveraging multiple network paths. However, if the performance between the multiple paths are significantly diverse, the receiver may have to wait for packets from the slower path, causing reordering and buffering problems. To solve this problem, previous MPTCP schedulers mainly focused on predicting the latency of the path beforehand. Recent studies, however, have shown that the path latency varies by a large margin over time, thus the MPTCP scheduler may wrongly predict the path latency, causing performance degradation. In this paper, we propose a new MPTCP scheduler called, choose fastest subflow (CFS) scheduler to solve this problem. Rather than predicting the path latency, CFS utilizes the characteristics of these paths to reduce the overall flow completion time by redundantly sending the last part of the flow to both paths. We compare the performance through real testbed experiments that implements CFS. The experimental results on both synthetic packet generation and actual Web page requests, show that CFS consistently outperforms the previous proposals in all cases.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

• ETRI Journal
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• v.24 no.1
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• pp.43-55
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• 2002

This paper proposes a flow-based admission control algorithm through an Asynchronous Transfer Mode (ATM) based Multi-Protocol Label Switching (MPLS) network for multiple service class environments of Integrated Service (IntServ) and Differentiated Service (DiffServ). We propose the Integrated Packet Scheduler to accommodate IntServ and Best Effort traffic through the DiffServ-aware MPLS core network. The numerical results of the proposed algorithm achieve reliable delay-bounded Quality of Service (QoS) performance and reduce the blocking probability of high priority service in the DiffServ model. We show the performance behaviors of IntServ traffic negotiated by end users when their packets are delivered through the DiffServ-aware MPLS core network. We also show that ATM shortcut connections are well tuned with guaranteed QoS service. We validate the proposed method by numerical analysis of its performance in such areas as throughput, end-to-end delay and path utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.762-771
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• 2005

In this paper, an efficient Adaptive quality-of-service (QoS) traffic control scheme with priority scheduling is proposed for the multimedia traffic transmission over wireless access networks. The objective of the proposed adaptive QoS control (AQC) scheme is to realize end-to-end QoS, to be scalable without the excess signaling process, and to adapt dynamically to the network traffic state according to traffic flow characteristics. Here, the reservation scheme can be used over the wireless access network in order to get the per-flow guarantees necessary for implementation of some kinds of multimedia applications. The AQC model is based on both differentiated service model with different lier hop behaviors and priority scheduling one. It consists of several various routers, access points, and bandwidth broker and adopts the IEEE 802.1 le wireless radio technique for wireless access interface. The AQC scheme includes queue management and packet scheduler to transmit class-based packets with different per hop behaviors (PHBs). Simulation results demonstrate effectiveness of the proposed AQC scheme.

• 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 Ship and Ocean Technology
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• v.8 no.4
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• pp.10-16
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• 2004

A response of a swept wing boundary layer to a single free-stream stationary axial vortex of a limited spanwise extent is considered as an example of typical problems that one can find in laminar-turbulent transition research and control. The response is dominated by streamwise velocity perturbations that grow quasi-exponentially downstream. It is shown that the formation of the boundary layer disturbance occurs for the most part close to the leading edge. The disturbance represents itself a wave packet consisted of the waves with characteristics specific for cross-flow instability. However, an admixture of growing disturbances whose origin can be attributed to transient effects and to a distributed receptivity mechanism is also identified.

• 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.