• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1225-1233
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• 2009

ISPs and Enterprises are equipping their networks with sufficiently high speed facilities and provide large bandwidths members. However the high speed enterprise network does not have satisfying end-to-end network performance within the network in spite of under utilization. The root cause of this performance degradation is a micro-congestion, which is a short-live event of traffic congestion. A micro-congestion causes packet loss, delay and packet reodering, and finally results in end-to-end network performance degradation. In this paper, we propose a micro-congestion detection method and find out the characteristics of performance degradation by analyzing traffic archives which is collected from a network link when a micro-congestion occurs.

• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.68-78
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• 2010

As the IP multicast function is very slowly deployed in Internet due to its scalability problem and inter-domain interoperability problem, interest in the P2P(Peer-to-Peer) streaming technologies for the realtime multimedia multicast applications such as IPTV is highly growing. This paper proposes a P2P streaming method for the QoS-sensitive multimedia multicast applications such as highly-interactive personal IPTV and video conferences. The proposed P2P streaming method allows an application to construct a reliable streaming tree in which a proper number of backup peers are placed according to its reliability requirement. The reliable streaming tree reduces the reconnection delay, occurred in the case of a normal and/or abnormal peer leave, so as to minimize the loss of streaming data. In the proposed P2P streaming method, the join delay of a peer called startup delay is also substantially reduced because the bandwidth and end-to-end delay information of every peer kept in a distributed way allows the target peer for a joining peer to be able to be quickly determined. Moreover, the proposed method's peer admission control mechanism based on the bandwidth and end-to-end delay enables the delay-bounded streaming services to be provided for its corresponding applications.

• Journal of Institute of Control, Robotics and Systems
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• v.2 no.1
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• pp.35-44
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• 1996

A design method for a teleoperator robot system is presented in this paper. The control system consists of two phases; approach phase and contact one. The end-effector position of the estimated slave robot and the contact force between the end-effector and wall are displayed on the monitors at control site, using which the operator controls the teleoperator system. The approach phase controller is designed using Smith's principle and the contact one designed based upon the structured singular value ${\mu}$ in order to increase the robustness of the system. The uncertainatices such as communication time delay and the variations of system parameters are considered as a muliplicative pertubation. Computer simulations are conducted in order to evaluate the performance of the proposed design method. It is found that desirable control performance, especially in the contact phase, is obtained if the control mode is switched into contact phase when the estimated position of the slave robot end-effector is in front of the wall.

• Journal of Communications and Networks
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• v.16 no.6
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• pp.656-666
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• 2014

In delay tolerant networks (DTNs), delay is inevitable; thus, making better use of buffer space to maximize the packet delivery rate is more important than delay reduction. In DTNs, epidemic routing is a well-known routing protocol. However, epidemic routing is very sensitive to buffer size. Once the buffer size in nodes is insufficient, the performance of epidemic routing will be drastically reduced. In this paper, we propose a buffer scheme to optimize the performance of epidemic routing on the basis of the Lagrangian and dual problem models. By using the proposed optimal buffer scheme, the packet delivery rate in epidemic routing is considerably improved. Our simulation results show that epidemic routing with the proposed optimal buffer scheme outperforms the original epidemic routing in terms of packet delivery rate and average end-to-end delay. It is worth noting that the improved epidemic routing needs much less buffer size compared to that of the original epidemic routing for ensuring the same packet delivery rate. In particular, even though the buffer size is very small (e.g., 50), the packet delivery rate in epidemic routing with the proposed optimal buffer scheme is still 95.8%, which can satisfy general communication demand.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.5
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.

• Journal of the Korean Society for Aviation and Aeronautics
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• v.21 no.1
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• pp.15-22
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• 2013

In oceanic flight routes, HF radio and satellite data links have weather restrictions, long propagation delay and low data throughput. In this paper, we propose oceanic aeronautical communications scheme in the VHF band based on ad-hoc communication. The proposed scheme organizes autonomously a multi-hop network that is divided into multiple local network using aircraft to fly long-distance communication and supports a hybrid type of multiple access, which consists of random access and TDMA (Time Division Multiple Access) scheme. In addition, several algorithms to apply spatial reuse of transmission to multi-hop long range communication environments have been proposed. The proposed system proves performance improvement on delay time as an effective solution to communicate end-to-end on the oceanic flight routes and strengthens the reliability of oceanic aeronautical communication.

• ETRI Journal
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• v.22 no.1
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• pp.30-37
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• 2000

Existing tree construction mechanisms are classified into source-based trees and center-based trees. The source-based trees produce a source-rooted tree with a low delay. However, for the applications with multiple senders, the management overheads for routing tables and resource reservations are too high. The center-based trees are easy to implement and manage, but a priori configuration of candidate center nodes is required, and the optimization mature such as tree cost and delay is not considered. In this paper, we propose a new multicast tree building algorithm. The proposal algorithm basically builds a non-center based shared tree. In particular, any center node is not pre-configured. In the purposed algorithm, a multicast node among current tree nodes is suitably assigned to each incoming user: Such a node is selected in a fashion that tree cost and the maximum end-to-end delay on the tree are jointly minimized. The existing and proposed algorithms are compared by experiments. In the simulation results, it is shown that the proposed algorithm approximately provides the cost saving of 30% and the delay saving of 10%, compared to the existing approaches. In conclusion, we see that the cost and delay aspects for multicast trees can be improved at the cost of additional computations.

• Annual Conference of KIPS
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• 2011.11a
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• pp.97-100
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• 2011

Underwater Wireless Acoustic Sensor Networks have become an important area of research over the recent decades. Designing an underwater network, especially a media access control (MAC) protocol, faces many challenges due to the peculiarities of underwater environment. One of the most important problems is resulted from long and variable propagation delay of the acoustic wave. In this paper, we propose a new method, namely Dynamic Timeout over Multiple Access with Collision Avoidance (DT/MACA), which is designed to handle long and high variable propagation delay in underwater acoustic sensor networks. In this proposed method, the difference timeout intervals are evaluated and applied to each network transmission. Simulation results show that our work not only improves the network throughput, but also decreases the unnecessary retransmission and end-to-end delay.

• Journal of Digital Contents Society
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• v.3 no.1
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• pp.101-111
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• 2002

Lately development of network around Internet expands range of data traffic to multimedia information, and so for the guarantee of multimedia services end-to-end QoS(Quality of Service) must service because comparing with existing Internet service can not support For satisfying those QoS requirements, network have to guarantee not only network on parameter, such as delay, jitter, throughput but also system resources like CPU utilization, memory usage. Therefore it is urgent to develop QoS based middleware to distribute multimedia data and maximize network utilization in the limited resource environment. And it must be necessary of network to provide end-to-end QoS(Quality of Service) for multimedia applications. Multimedia applications want that QoS which satisfy their own service properties be guaranteed Then, We must analyze those necessary QoS requirements md define QoS parameter which specify as two viewpoint, user's and network's perspective. Therefore network provider supplying network for usual user and university, enterprise must want to find about their own network performance and problem. It is essential for network manager to want to use a tool like this. On the basis of technique about QoS test-bed in the AIM network, We studied on the method of QoS measurement and management about end-to-end connection in the Internet. We measured network status about end-to-end connection and analyze the result of performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10B
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• pp.630-638
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• 2005

As real-time media applications based on IP multicast networks spread widely, the end-to-end QoS (quality of service) provisioning for these applications have become very important. To guarantee the end-to-end QoS of multi-party media applications, it is essential to monitor the time-varying status of both network metrics (i.e., delay, jitter and loss) and system metrics (i.e., CPU and memory utilization). In this paper, targeting the multicast-enabled AG (Access Grid) group collaboration tool based on multi-Party real-time media services, a hybrid monitoring scheme that can monitor the status of both multicast network and node system is investigated. It combines active monitoring and passive monitoring approaches to measure multicast network. The active monitoring measures network-layer metrics (i.e., network condition) with probe packets while the passive monitoring checks application-layer metrics (i.e., user traffic condition by analyzing RTCP packets). In addition, it measures node system metrics from system API. By comparing these hybrid results, we attempt to pinpoint the causes of performance degradation and explore corresponding reactions to improve the end-to-end performance. The experimental results show that the proposed hybrid monitoring can provide useful information to coordinate the performance improvement of multi-party real-time media applications.