• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Journal of Internet Computing and Services
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• v.14 no.5
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• pp.39-48
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• 2013

A cloud system has to deal with highly variable workloads resulting from dynamic usage patterns in order to keep the QoS within the predefined SLA. Aside from the aspects regarding services, another emerging concern is to keep the energy consumption at a minimum. This requires the cloud providers to consider energy and performance trade-off when allocating virtualized resources in cloud data centers. In this paper, we propose a resource provisioning approach based on dynamic thresholds to detect the workload level of the host machines. The VM selection policy uses utilization data to choose a VM for migration, while the VM allocation policy designates VMs to a host based on its service reputation. We evaluated our work through simulations and results show that our work outperforms non-power aware methods that don't support migration as well as those based on static thresholds and random selection policy.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1251-1264
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• 2002

Currently, audio streaming services such as on-demand service and live service support only a limited number of clients. They also suffer from a lack of stability and degradation of service quality due to their inefficient use of network resources. Futhermore, since the streaming services usually do not consider dynamic services, they are very inconvenience to use. In order to resolve these drawbacks, we propose a novel audio streaming system based on MPEG-2 AAC file format which are facilitated with the network bandwidths efficiently. The proposed system supports QoS for audio streaming as well as guarantees a stability while streaming service is undergoing. Moreover, the system provides a dynamic interface which enables us to use the streaming service more easily and to manage streaming servers with convenient manner. On the contrary, most of the current available static interface streaming services are mainly depending only on a single fixed web page between client and server, which in consequence lead us to use unflexible static service environment. Our implementation results show the proposed system improves the performance compared to those of the currently existing systems that use MP3 file format. It also provides some benefits such as a stability of service and a easy to management of streaming servers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.295-307
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• 2002

The IEEE 1394 protocol is a de facto standard in multimedia digital home network. It supports several advanced features such as hot plugging, dynamic network reconfiguration, isochronous transmission and so on. Since the IEEE 1394 was adapted in the field of multimedia service with QoS guarantee, back bone network protocol with reel-time digital instrumentation and control sub networks, and physical layer protocol for real-time middleware such as real-time CORBA, the additional real-time features has been required in device driver implementation because of the necessity of the predictability enhancement. To guarantee the real-time features, the device driver of the IEEE 1394 should support the priority based packet processing and also need a isochronous buffer management mechanism to deal with the periodic isochronous communication. In this paper, we proposed a new software architecture of the IEEE 1394 device driver for supporting the real-time characteristics such as priority based packet processing, priority based scheduling and so on.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 2003.05a
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• pp.8-13
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• 2003

Historically, IP-based internets have been able to provide a simple best-effort delivery service to all applications they carry. Best effort treats all packets equally, with no service level, packet loss, and delay. But the needs of users have changed. The want to use the new real-time, multimedia, and multicasting applications. Thus, there is a strong need to be able to support a variety of traffic with a variety of quality-of-service requirements. The DiffServ architecture, proposed by the Internet Engineering Task Force(IETF), has become the most viable solution for provising QoS over IP networks. The DiffServ architecture does not specify any handling method between AF out-profile packets and BE packets. This paper propose a mechanism for supporting inter class fairness in the DiffServ architecture. Ⅰ proposed a modified Weighted Round Robin method to protect the BE traffic from AF out-profile packets in the core routers. The proposed technique is evaluated through simulation. Simulation results indicate that the proposed method provides better protection not only for BE packets from AF out-profile packets, but also for the AF in-profile packets in congested networks.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.111-120
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• 2001

VC merging involves distinguishing cells from an identical merged VC label. Various approaches have been proposed to help this identification process. However, most of them incur additional buffering, protocol overhead and/or variable delay. They make the provision of QoS difficult to achieve. So it was proposed a merge capable scheduler to support VC-merging (VCMS). However, in situations where all VCs are to be merged or the incoming traffic load is very low, it could happen that there are not enough non-merging cells to snoop. In this situation the scheduler uses special control cells to fill the empty time slots out. Too many control cells can cause high cell loss ratio and an additional packet transfer delay. To overcome the drawbacks, we propose a Sequence Skipping(SS) method where the sequencers skip the empty queues and insert SS cells. We show SS method is suitable for VC-merging and can reduce the cell loss ratio and the mean packet transfer delay through simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.9
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• pp.2422-2432
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• 1996

In this thesis, we analyze the Broadcast and Select Network(BSN) architecture coming up to the issues among those in the single-hip WDM networks and the degradation of performance in consequence of the traffic imbalance which well be able to be produced in these network. We propose WDM networks architecture which can support concurrency services by using Array Wavelength Tunable Filter(AWTF) with the parrallel hardware structure as a solution. For the sake of multichannel communications, we make use of the Single Control channel, Multiple Data channel(SCMD) scheme that is one of the channel architecture. Also, we compare and analize the throughput of Uniform & Nonuniform traffic conditions, and obtain the optimized number of AWTF being able to guarantee the Quality of Service(QoS) required by the enlargement of WDM netowrks and traffic imbalance. Therefore, we can make it possible that the design of the WDM networks with BSN architecture ensures the optimized performance.

• Journal of the Korea Society of Computer and Information
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• v.17 no.6
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• pp.57-66
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• 2012

LTE-Advanced network will form the high-speed IP backbone in collaboration with heterogeneous radio access networks for dynamic optimized resource utilization. In order to implement more innovative and attractive services such as U-Cloud streaming, LBS and mobile smart TV, a mobile terminal needs to support multiple sessions simultaneously. Efficient resource allocation schemes are necessary to maintain QoS of multiple sessions because service continuity may be defected by delay and information loss during handover. This paper proposes a resource allocation scheme to accommodate multiple sessions in a mobile terminal on handover period based on session priority mechanism. Simulation is focused on the forced termination probability of handover sessions. Simulation results show that our proposed method provides a better performance than the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4B
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• pp.153-161
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• 2005

In a multi-channel network based of wavelength division multiplexing (WDM) technology, an efficient protocol is needed in order for the transmitter and receiver to be tuned to the same wavelength during message transmission time. This paper proposes a priority-based reservation protocol that can efficiently support variable-length messages that require different QoS requirements. In this proposed protocol, high-priority nodes can reserve the data channel before low-priority nodes. However, once a node reserves a data channel successfully, it can persistently use the reserved channel till message transmission is finished regardless of the priority of the node. Moreover, the protocol can operate independently of the number of nodes, and any new node can join the network anytime without requiring network re-initialization. The protocol is analyzed with a finite population model and the throughput-delay characteristics are investigated as performance measures.

• Journal of Broadcast Engineering
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• v.13 no.2
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• pp.261-273
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• 2008

Multimedia service whose use is rapidly increasing supports effective services to convert and transmit multimedia data based on network speed, noise circumstance, terminal computation, and type of contents for satisfying QoS. For supporting information protection of multimedia service, it offers middle level of singular security service or security mechanism which is based on policy of service provider, depending on present terminal computation and type of contents. It can support security mechanism for more effective multimedia service, if we study security of application layer and network layer for supporting multimedia service. In this paper, we propose Multimedia security framework reflected on quantitative analysis of the WLAN(Wireless Local Area Network) mesh network security using the utility function in the level of the sorority, violation and addictive compensation model.