• Proceedings of the Korea Society for Simulation Conference
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• 2001.10a
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• pp.440-443
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• 2001

As inter-network traffics grows rapidly, the router systems as a network component becomes to be capable of not only wire-speed packet processing but also plentiful programmability for quality services. A network processor technology is widely used to achieve such capabilities in the high-end router. Although providing two such capabilities, the network processor can't support a deep packet processing at nominal wire-speed. Considering QoS may result in performance degradation of processing packet. In order to achieve foster processing, one chipset of network processor is occasionally not enough. Using more than one urges to consider a problem that is, for instance, an out-of-order delivery of packets. This problem can be serious in some applications such as voice over IP and video services, which assume that packets arrive in order. It is required to develop an effective packet processing mechanism leer using more than one network processors in parallel in one linecard unit of the router system. Simulation analysis is also needed for verifying the mechanism. We propose the packet processing mechanism consisting of more than two NPs in parallel. In this mechanism, we use a load-balancing algorithm that distributes the packet traffic load evenly and keeps the sequence, and then verify the algorithm with simulation analysis. As a simulation tool, we use DEVSim++, which is a DEVS formalism-based hierarchical discrete-event simulation environment developed by KAIST. In this paper, we are going to show not only applicability of the DEVS formalism to hardware modeling and simulation but also predictability of performance of the load balancer when implemented with FPGA.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.865-874
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• 2002

HTTP (HyperText Transfer Protocol) is a transfer protocol used by the World Wide Web distributed hypermedia system to retrieve the objects. Because HTTP is a connection oriented protocol, it uses TCP (Transmission control Protocol) as a transport layer. But it is known that HTTP interacts with TCP badly. it is discussed about factors affecting the performance or HTTP over TCP, the transaction time obtained by the per-transaction TCP connections for HTTP access and the TCP slow-start overheads, and the transaction time for T-TCP (Transaction TCP) which is one or methods improving the performance or HTTP over TCP. Average waiting time and average turnaround time are important parameters to satisfy QoS (Quality of Service) of end users. Formulas for calculating two parameters are derived. Such formulas can be used for the environment in which each TCP or T-TCP transaction time is same or different. Some experiments and computational experiences indicate that the proposed formulas are well acted, can be applied to the environment which the extension of bandwidth is necessary, and time characteristics of T-TCP are superior to that of TCP. Also, the load distribution method of web server based on the combination of bandwidths is discussed to reduce average waiting time and average turnaround time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1B
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• pp.27-38
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• 2005

IP Multicast has not been deployed because of hardware problems. So a new scheme that is called Overlay Multicast for group communication has been emerged. It supports IP Multicast functions, which is located on application level. For developing it, we have been focused on efficient overlay tree construction among group members with low stretch and stress. However, we should consider a variety of transmission or receipt condition since a real internet environment has users with various transmission/receipt rates. Thus, we make one-time source specific tree depending on required bandwidth informationof group members when a member requests data transmission. Our mechanism provides satisfied data quality limited maximum transmission rate of the source to each group members. Furthermore, we manage a large group enough as distributing control information to cores that are designated membersfor maintaining host member information. Lastly, we prove that our tree guarantees data quality to each group members, and show low tree consruction time is required. In addition, for evaluating group scalability, we analyze control information increasing rate via group size, and validate its scalability.

• Journal of the Korea Society for Simulation
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• v.28 no.2
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• pp.119-129
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• 2019

Although line production systems with finite buffers have been studied over several decades, except for some special cases there are no explicit expressions for system performances such as waiting times(or response time) and blocking probability. Recently, a max-plus algebraic approach for buffer-sharing systems with constant processing times was introduced and it can lead to analytic expressions for (higher) moment and tail probability of stationary waiting. Theoretically this approach can be applied to general processing times, but it cannot give a proper way for computing performance measures. To this end, in this study we developed simulation models using @RISK software and the expressions derived from max-plus algebra, and computed and compared blocking probability, waiting time (or response time) with respect to two blocking policies: communication(BBS: Blocking Before Service) and production(BAS: Blocking After Service). Moreover, an optimization problem which determines the minimum shared-buffer capacity satisfying a predetermined QoS(quality of service) is also considered.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.2
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• pp.834-841
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• 2012

In order to the mobile service uses the wireless terminal including the development of the wire wireless network and Smart phone, and etc. use including mobile office, which and etc. increases. Because in order that the u-Healthcare service using this appeared before the footlights and the existing quality measurement reference considered the speed, error rate, and etc. just, guarantees the stability of the u-Healthcare, the quality control by service are necessary. In this paper, the quality measurement reference by mobile service considering the radio environment as the method for satisfying the quality guarantee of the u-Healthcare mobile service user and user needs was presented. The WiBro u-Healthcare wireless data service quality based system in the end user perspective was established through the main performance index and entrepreneur case presented in the international standardization institute including 3GPP, WiMAX forum, GSMA, and etc. through the related research and the validity of the quality index establishment was presented according to each service.

• Archives of design research
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• v.11 no.1
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• pp.291-302
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• 1998

This paper covers the development process of multimedia evaluation system, especially focused on customer satisfactory factors while customers navigating net-based Interactive multimedia system. Customers usually experience new level of interaction cased by newly developed web-based technology In ordinary multimedia system. However, if it gives customers satisfactory experience is a matter of question. To find out the relationship between customer satisfaction and interactivity factors exposed by multimedia system, a model has been developed which describes the structure of web-based multimedia system and its relation to customer satisfactory factors. Five different experiments, including 'semantic differential', 'focus group interview', and 'expert review', has been conducted and four customer satisfactory factors were identified. Those are 'customery value', 'structural perfectness', 'visual perfectness', and 'contemporaneity'. With these factors and newly delveoped evaluation system, 7 different web-site has been evaluated and analyzed at the end of this report.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11A
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• pp.1908-1924
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• 2001

Interoperability and inter-working in the various network and media environment with different technology background is very important to enlarge the opportunity of service access and to increase the competitive power of service. The ITU-T and advanced counties are planning ahead for provision of GII enabling user to access advanced global communication services supporting multimedia communication applications, embracing all modes of information. In this paper, we especially forced the heterogeneity of end user applications for multimedia networking. The heterogeneity has several technical aspects, like different medium access methods, heterogeneous coding algorithms for audio-visual data and so on. Among these elements, we have been itemized bit rate shaping algorithm on the compressed moving video. Previous manipulations of video has been done on the uncompressed signal domain. That is, compressed video should be converted to linear PCM signal. To do such a procedures, we should decode, manipulate and then encode the video to compressed signal once again. The traditional approach for processing the video signa1 has several critical weak points, requiring complexity to implement, degradation of image quality and large processing delay. The bit rate shaping algorithm proposed in this paper process the manipulation of moving video on the completely compressed domain to cope with above deficit. With this algorithms. we could realized efficient video bit rate shaping and the result of software simulation shows that this method has significant advantage than that of pixel oriented algorithms.