• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.1C
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• pp.76-88
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• 2012

Traffic classification for application services in military communication networks is one of the core requirements to provide efficient resource management for NCW operations. Though several standards and organizations provided their own traffic classification methods, the methods have been mainly focused on commercial services, but not reflected military specifics. In addition, though various military application services have been emerged, some of those services have been classified into different traffic classes due to implicit criteria for the classification among organizations. In this paper, we propose a methodology to classify traffic classes and to determine delivery requirements for military application services based on various standards by DoD as well as several commercial standards based on Y.1541. The validation of the proposed methodology was carried out by comparing the results by proposed one with the traffic classification suggestions by existing commercial standards, DoD and FCS. We expect that the proposed methodology can contribute to achieve efficient operations of limited military network resources, since the proposed method can provide systematic and consistent way to assign traffic classes for new and existing military applications services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.7B
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• pp.433-439
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• 2005

In this paper, we propose a Flow Based Waiting Time Priority (FB-WTP) scheduler that supports the proportional delay differentiated services between classes. Existing scheduling algorithms utilize the configuration of scheduler, which is operated as class unit, applied to wired network. However, FB-WTP scheduler is operated as flow unit and can take advantage of multi-flow diversity effect in time-varying channel state environment. As a result, FB-WTP improves the average queueing delay on each class as well as supporting the average queueing delay ratio between classes. It also solves the HOL packet blocking problem implicitly. Simulation results show that FB-WTP scheduler has better system queueing delay performance than Look-ahead Waiting Time Priority (LWTP) scheduler and supports the larger queueing delay ratio between classes that network operator set.

• 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.27 no.8C
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• pp.758-770
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• 2002

In this paper, we propose a novel resource allocation scheme which can maximize capacity for QoS guarantee of Assured Service in Differentiated Services. Performance of the proposed resource allocation scheme is analyzed with each buffer management scheme such as RIO and Adaptive RIO. To prevent an early random drop of the admitted In-profile packet, Adaptive RIO scheme updates parameters of RIO scheme every time interval according to the estimated numbers of maximum packet arrivals of In-profile traffic and Out-of-profile traffic during the next time interval. The numbers of maximum packet arrivals during the next time interval are estimated based on the buffer size determined by the network topology and the ratio of bandwidth allocated to each subclass. We can find from simulation results that proposed resource allocation scheme with Adaptive RIO can guarantee QoS and can maximize capacity for Assured Service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.711-723
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• 2002

A finer granular scalable (FGS) version of ISO/IEC MPEG-4 video streaming is investigated in this work with the prioritized stream delivery over loss-rate differentiated networks. Our proposed system is focused on the seamless integration of rate adaptation, prioritized packetization, and simplified differentiation for the MPEG-4 FGS video streaming. The proposed system consists of three key components: 1) rate adaptation with scalable source encoding, 2) content-aware prioritized packetization, and 3) loss-based differential forwarding. More specifically, a constant-quality rate adaptation is first achieved by optimally truncating the over-coded FGS stream based on the embedding rate-distortion (R-D) information (obtained from a piecewise linear R-D model). The rate-controlled video stream is then packetized and prioritized according to the loss impact of each packet. Prioritized packets are transmitted over the underlying network, where packets are subject to differentiated dropping and forwarding. By focusing on the end-to-end quality, we establish an effective working conditions for the proposed video streaming and the superior performance is verified by simulated MPEG-4 FGS video streaming.

• Journal of Korea Multimedia Society
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• v.7 no.4
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• pp.518-531
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• 2004

This paper discusses a testbed architecture for implementing scalable service differentiation in the Internet. The differentiated services (DiffServ) testbed architecture is based on a model in which a bandwidth broker (BB) can control network resources, and the ALTQ can reserve resources in a router to guarantee a Quality of Service (QoS) for incoming traffic to the testbed. The reservation and releasemessage for the ALTQ is contingent upon a decision message in the BE. The BB has all the information in advance, which is required for a decision message, in the form of PIB. A signaling protocol between the BB and the routers is the COPS protocol proposed at the IETF. In terms of service differentiation, a user should make an SLA in advance, and reserve required bandwidth through an RAR procedure. The SLA and RAR message between a user and the BB has implemented with the COPS extension which was used between a router and the BB. We evaluates the service differentiation for the video streaming in that the EF class traffic shows superb performance than the BE class traffic where is a network congestion. We also present the differentiated service showing a better packet receiving rate, low packet loss, and low delay for the EF class video service.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.