• 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.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.3
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• pp.127-135
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• 2015

Context-aware mobile cloud computing is a new promising paradigm that allows to improve user experience by analyzing contextual information such as user location, time of the day, neighboring devices and current activity. In this paper we provide performance study of context-aware mobile cloud computing system with Volare middleware. Volare monitors the resources and context of the device. and dynamically adapts cloud service requests accordingly, at discovery time or at runtime. This approach allows for more resource-efficient and reliable cloud service discovery, as well as significant cost savings at runtime. We also have studied the performance of context-aware mobile cloud computing for different quality of service (QoS) adaptation policies. Our simulations results show that when battery level is low and CPU usage is high and user cannot maintain the initial QoS, service cost is decreased according to current adaptation policy. In conclusion, the current adaptation policy suggested in this paper may improve user experience by providing a dynamically adapted service cost according to a situation.

• ETRI Journal
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• v.38 no.3
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• pp.444-454
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• 2016

Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12B
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• pp.1017-1027
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• 2006

Recently, Internet service providers are offering triple play service which combines voice and video services with the existing high speed Internet service under the common network infrastructure of IP protocol, so that they can create more benefit from operating a single network architecture. In line with this movement in the convergence of network and services, in this work we propose a bandwidth dimensioning method for the subscriber network of the IP network at flow level. To that purpose, let us propose a series of bandwidth dimensioning methods: bandwidth for best effort service only, bandwidth dimensioning for premium services, and bandwidth dimensioning schemes for the premium Internet services as well as the best effort service which comprise the TPS. Our link dimensioning method is based on the flow level that incorporates the flow blocking probability as a measure of grade of services(GoS), and investigates the characteristics of the proposed methods via extensive numerical experiments. After that, let us carry out a simulation experiment concerning the delay and loss performance of the packet scheduling for the premium services (QoS) using the bandwidth designed by our proposed method, via which the packet level quality of service (QoS) for the proposed link dimensioning method can be observed.

• Journal of Information Processing Systems
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• v.5 no.1
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• pp.25-32
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• 2009

It is absolutely essential to implement advanced IP network technologies such as QoS, Multicast, High Availability, and Security in order to provide real-time services like IPTV via IP backbone network. In reality, the existing commercial networks of internet service providers are subject to certain technical difficulties and limitations in embodying those technologies. On-going research efforts involve the experimental engineering works and implementation experience to trigger IPTV service on the premium-level IP backbone which has recently been developed. This paper introduces the core network technologies that will enable the deployment of a high-quality IPTV service, and then proposes a suitable methodology for application and deployment policies on each technology to lead the establishment and globalization of the IPTV service.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.1
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• pp.191-197
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• 2012

In this paper we propose the FQSM(Fairness and QoS Supporting MAC) protocol that supports fairness and Quality of Service(QoS). The received or measured data traffics will be assigned a priority level according to its transmission urgency in the FQSM. And the load prediction algorithm is used to support the fairness between different priority traffics. For this, the buffer length values of the nodes are continuously monitored for a some period. Based on the buffer length variations for this period, the order of transmission is determined. FQSM also adapts cross-layer concept to rearrange the data transmission order in each sensor node's buffer, saves energy consumption by allowing few nodes in data transmission, and prolongs the network lifetime.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.5B
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• pp.295-303
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• 2005

Recently, according to the rapid increase of Web users, various kinds of Web applications have been being developed. Hence, Web QoS(Quality of Service) becomes a critical issue in the Web services, such as e-commerce, Web hosting, etc. Nevertheless, most Web servers currently process various requests from Web users on a FIFO basis, which can not provide differentiated QoS. This paper presents a load balancing method to provide differentiated Web QoS in clustering web server. The first is the kernel-level approach, which is adding a real-time scheduling process to the operating system kernel to maintain the priority of user requests determined by the scheduling process of Web server. The second is the load-balancing approach, which uses IP-level masquerading and tunneling technology to improve reliability and response speed upon user requests. The third is the dynamic load-balancing approach, which uses the parameters related to the MIB-II of SNMP and the parameters related to load of the system such as memory and CPU.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.281-284
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• 2004

This paper presents the design and implementation of a QoS based enhanced collaboration system in MDO. This is an efficient distributed communication tool between designers. It supports text communication, audio/video communication, file transfer and XML data sending/receiving. Specially, this system supports a dynamic QoS self-adaptation by using the improved direct adjustment algorithm (DAA+). The original direct adjustment algorithm adjusts the transmission rate according to the congestion level of the network, based on the end to end real time transport protocol (RTP), and controls the transmission rate by using the information of loss ratio in real time transport control protocol (RTCP). But the direct adjustment algorithm does not consider when the RTCP packets are lost. We suggest an improved direct adjustment algorithm to solve this problem. We apply our improved direct adjustment algorithm to our of QoS (Quality of Service) [1] based collaboration system and show the improved performance of transmission rate and loss ratio.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.27-30
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• 2000

In this paper, we propose a new dynamic bandwidth allocation and call admission control method for the VBR video sources with QoS constraints to provide user's quality of service requirements and at the same time to achieve an efficient resource management in networks. The proposed mechanism dynamically adjusts the necessary bandwidth by the networks based on the provided quality of service satisfaction degree of each connection in respect to the user's requirements in terms of loss ratio and average delay Simulation results show that our proposed dynamic method is able to provide the desired level of quality of service and high utilization.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.303-308
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• 2016

Recently, as the rapid development of network technology and the increase of services required high bandwidth such as multimedia service, the network traffic dramatically increases. This massive increase of network traffic causes some problems such as the degradation of QoS and the lack of network resources and, to solve these problems, various research to guarantee QoS have been performing. Currently, The most representative method to guarantee the QoS is the DiffServ(: Differentiated Service). The DiffServ defines the AF(: Assured Forwarding) PHB(: Per Hop Behavior) and statistically ensures the throughput over the certain level of data rate. However, the TCP congestion control method that make up the majority of the Internet traffic is not fundamentally suitable to the DiffServ that guarantees the throughput without managing the individual flow. Therefore, in this paper, we present this mismatch through the simulation as an example and propose the solution by controlling the TCP of the terminal in the network. The proposed scheme utilizes the information of the reception window size included in the ACK frame and does not require any modification of the TCP algorithms currently in use.