• Macromolecular Research
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• v.14 no.2
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• pp.140-145
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• 2006

E-science refers to the large-scale science that will increasingly be carried out through distributed global collaborations enabled by the Internet. The Grid is a service-oriented architecture proposed to provide access to very large data collections, very large scale computing resources and remote facilities. Web services, which are server applications, enable online access to service providers. Web portal interfaces can further hide the complexity of accessing facility's services. The main use of synchrotron radiation (SR) facilities by protein crystallographers is to collect the best possible diffraction data for reasonably well defined problems. Significant effort is therefore being made throughout the world to automate SR protein crystallography facilities so scientists can achieve high throughput, even if they are not expert in all the techniques. By applying the above technologies, the e-HTPX project, a distributed computing infrastructure, was designed to help scientists remotely plan, initiate and monitor experiments for protein crystallographic structure determination. A description of both the hardware and control software is given together in this paper.

• Journal of KIISE:Information Networking
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• v.33 no.5
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• pp.380-394
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• 2006

Traditional Internet services provide the same level of qualify to all users without considering about their application's QoS (quality of service) requirements. Because the wireless and Internet technologies are evolving rapidly, the powerful mobile devices are more widely used. Therefore, the new applications such as DMB (digital multimedia broadcasting), VoD (video on demand), telemedicine, and etc. demand appropriate QoS and host mobility. To satisfy free mobility and QoS, we deploy Diffserv (differentiated services) model to the mobile IPv6 networks. We propose the operational procedures and cost analyzing models for providing an MN's mobility and QoS. Also its performance is presented by cost analyzing and the future works are described.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1257-1260
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• 2003

현재 국방전산망은 인터넷과 마찬가지로 Best Effort 서비스만을 지원하고 있으나 급속도로 발전하는 외부의 미래 네트워크 환경에 대비하여 지속적으로 진화가 추진되고 있다. 본 논문에서는 미래 국방전산망의 요구사항인 서비스의 차별화에 관점을 두고 신속하고 신뢰성 있는 중요 의사결정 자료 전송을 위해 국방전산망에 ATM기반 MPLS 기술을 적용하여 군 기능별 차등서비스 제공방안 및 서비스 우선순위에 따른 트래픽 재 할당에 대하여 제안 하고자 한다.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.145-148
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• 2003

A lot of institutes have proposed RMT (Reliable Multicast Transport) techniques over IP multicast environment which provides only best-effort services. The ECTP (Enhanced Communications Transport Protocol) is a kind of RMT technique and an international standard of ITU-T and JTC1/SC6. It is a protocol that provides reliability fer IP multicast environment. In this paper, we tested ECTP applications which are MFTP (Multicast file transport protocol) and WB (white Board) on the APAN (Asia-Pacific Advanced Network) with the Kyushu University in Japan. We measured ECTP traffic using a packet capture tool that programmed by Pcap library that is developed in Lawrence Berkeley institute. In according to the result, we think that ECTP is an efficient protocol in transporting files to many receivers simultaneously.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.30 no.1
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• pp.59-65
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• 2007

HPI (High-speed Portable Internet) system which provides high speed internet services is going to be commercialized soon. Since HPI provides simultaneously four different service types such as UGS (Unsolicited Grant Service), rtPS (real time Polling Service), nrtPS(non-real time Polling Service), and BE (Best Effort) under different QoS (Quality of Service) requirements and limited wireless channel resources, efficient packet scheduling mechanisms are necessary to increase the utilization of channels as well as to satisfy the various QoS requirements. This study regards the traffic data to be served as time series and proposes a new packet scheduling algorithm based on the nonparametric statistical test. The performance of the newly proposed algorithm is evaluated through the simulation analysis using a simulator that can evaluate the performance of packet scheduling mechanisms under various values of system parameters and measures such as packet delay time, data transmission rate, number of loss packets, and channel utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1517-1531
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• 1999

The Internet is recently evolving into the directions which is capable of incorporating stream-type services such as voice and video services. However, the existing Internet can only support a best-effort delivery model and thus it is difficult to satisfy these requirements as well as to provide the QoS beyond a certain degree. Accordingly, it is imperative to develop QoS based routing algorithm in order to allow the routing by the user's QoS demand and to be applied into wide area networks. In this thesis, we presented new routing algorithms and made performance evaluations. The proposed algorithms allow the routing by the user’s QoS demand level in order to provide the diverse Internet application services based on ATM network environment that is expected to play a role as an infrastructure of high-speed communication network. They also have a feature of shortening the waiting time for address analysis and have a quality-proven characteristics with enhanced network scalability.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.123-134
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• 2002

The requirement for QoS (Quality of Service) has become an important Issue as real-time or high bandwidth services are increasing, such as Internet Telephony, Internet broadcasting, and multimedia service etc. In order to guarantee the QoS of Internet application services, several approaches are being sought including IntServ (Integrated Service) DiffServ(Differentiated Srvices), and MPLS(Multi-Protocol Label Switching). In this paper, we describe the performance analysis of QoS guarantee mechanism using the DiffServ. To analyze how the DiffServ performance was affected by diverse input traffic models and the weight value in WFQ(Weighted Fair Queueing), we simulated and performed performance evaluation under a random, bursty, and self-similar input traffic models and for diverse input parameters. leased on the results of performance analysis, it was confirmed that significant difference exist in packet delay and loss depending on the input traffic models used. However, it was revealed that QoS guarantee is possible to the EF (expedited Forwarding) class and the service separation between RF and BE (Best Effort) classes may also be achieved. Next, we discussed the performance synthesis problem. (i. e. derived the conservation laws for a DiffServ networks, and analysed the performance variation and dynamic behavior based on the resource allocation (i.e., weight value) in WFQ.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.605-614
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• 2009

Existing Internet services were almost supported by the best effort service such as the data transmission service and were allowed the transmission delay and packet loss. However, recent Internet multimedia services such as Internet phone, Internet broadcasting are required a real-time processing and high bandwidth. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED (Random Early Detection) algorithm has proposed. However, RED algorithm has a limitation of usage of an average buffer length and unfairness. So, many algorithms proposed as the modified algorithm of RED. But these modified algorithms also have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the QoS adaptive AQM (Active Queue Management) algorithm for the multimedia services that request various QoS requirements and present a performance evaluation by the simulations using the ns-2.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.928-938
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• 2008

Though services requiring Quality-of-Service (QoS) guarantees such as Voice over Internet Protocol (VoIP) have been widely deployed on the internet, most of internet backbone networks, unfortunately, do not distinguish them from the best-effort services. Thus estimating the effective capacity meaning the traffic volume that the backbone networks maximally accommodate with keeping QoS guarantees for the services is very important for Internet Service Providers. This paper proposes a test-bed based on ns-2 to evaluate the effective capacity of backbone networks and then estimates the effective capacity of an experimental backbone network using the test-bed in terms of the service standard of the VoIP service. The result showed that the effective capacity of the network is estimated as between 12% and 55% of its physical capacity, which is depending on the maximum delay guarantee probability, and strongly affected by not only the type of offered workload but also the quality standard. Especially, it demonstrated that in order to improve the effective capacity the maximum end-to-end delay requirement of the VoIP service needs to be loosened in terms of the probability to guarantee.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.372-375
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• 2008

IEEE 802.11e standardized the EDCA mechanism to support the priority based QoS. And the virtual collision handler schedules the transmission time of each MAC frame using the internal back-off window according to the access category(AC). This can provides the differentiated QoS to real-time services at the medium traffic load condition. However, the transmission delay of MAC frame for real-time services may be increased as the traffic load of best effort service increases. It becomes more critical when the real-time service uses a compressed mode video codec such as moving picture experts group(MPEG) 4 codec. That is because each frame has the different importance. That is, the I-frame has more information as compared with the P- and the B-frame. In this paper, we proposed a buffer management algorithm based on the frame importance and the delay bound. The proposed algorithm is consisted of the traffic regulator based on the dual token bucket algorithm and the active queue management algorithm. The traffic regulator reduces the transmission rate of lower AC until that the virtual collision handler can transmit an I-frame. And the active queue management discards frame based on the importance of each frame and the delay bound of head of line(HoL) frame when the channel resource is insufficient.