• Proceedings of the Korean Information Science Society Conference
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• 2006.06a
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• pp.211-213
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• 2006

실시간으로 변화하는 컴퓨터 통신 환경에서 멀티미디어 응용 프로그램은 QoS를 만족하기 위해 안정적으로 튜닝 되고 재구성되는 것이 필요하다. 그러나 안정적으로 QoS를 보장하는 것은 응용 프로그램의 자원 예약이나 실시간 보장과 같은 메카니즘을 제공하지 않은 일반적인 목적의 시스템 상에서 수행될 때 많은 어려움을 가지게 된다. 특히, 예측 불가능한 개방형 환경에서 최우선 자원 할당에 의해 발생되는 자원의 유효성에 대응하기 위해 QoS 적응은 수행되어야 한다. 그러나 적응을 언제, 어떻게 조정해야 하고 폭 넓은 범위에서 응용 프로그램에 어떻게 적용시킬지를 알기 위해 일반적인 알고리즘을 제시해야할 필요가 있다. 이러한 목적을 위해, 본 논문에서는 멀티미디어 어플리케이션의 파라미터를 모델링하고, 파라미터간의 관계를 정량적으로 얻기 위해 계층적 QoS 프로빙 알고리즘을 적용한다. 이것을 기반으로 설계된 베이지안 네트워크를 이용하여 불확실한 정보를 확률값으로 처리함으로써 적응 행동을 예측하도록 한다. 마지막으로 실제 실험을 통해 제안된 미들웨어의 유용성을 확인한다.

• Journal of KIISE:Computing Practices and Letters
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• v.13 no.5
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• pp.260-270
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• 2007

Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. However, since mobile clients use wireless network and the wireless network have less stable bandwidth according to distance from AP and environments, it is hard to support stable QoS to mobile clients. In this paper, we propose a dynamic bit rate control method for transcoding in order to supprot stable QoS of streaming media service to mobile clients via wireless network. The proposed method is shown to serve seamless streaming media service with adaptive bit rate control according to state of wireless network in real time and to reduce transmission failure from experiment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.641-649
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• 2004

We propose a dynamic traffic management model which uses adaptive priority reassignment algorithm to deliver service differentiation in home networks, and implement adaptive priority reassignment algorithm using FPGA. The proposed architecture is designed to handle home network traffic without the need for signaling protocol. We categorize home network traffic into three kinds of traffic class: control data traffic class, the Internet data and non-real-time data traffic class, and multimedia data traffic class (include non-real-time and real-time multimedia data traffic). To support differential service about these kinds of traffic class, we designed and implemented a traffic management framework that dynamically change each traffic class priority depending on bandwidth utilization of each traffic class.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.674-676
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• 2000

인터넷과 같은 이질적인 네트워크환경에서 대용량의 비디오 데이터를 실시간으로 전송하기 위해서 계층적 코딩기법에 관한 연구가 진행되고 있으며, MPEG-2 비디오에서도 Scalability를 제공하고 있다. 하지만 이러한 계층적 코딩기법을 사용한다 하더라도 기본계층의 데이터량이 크기 때문에 실시간 서비스를 위해서는 상당량의 네트워크 자원을 필요로 하게 된다. 이에 네트워크 대여폭 변화에 능동적으로 적응하면서 네트워크 자원을 보다 효율적으로 사용할 수 있는 새로운 계층적 코딩기법과 전송 기법을 제안한다. 제안된 기법은 MPEG 비디오를 픽처층을 이용하여 공간적 계층적 코딩 후 다시 DCT블록에 대한 공간적 계층적 코딩을 수행하여 15개의 레이어로 분할하게 된다. 이렇게 분할된 레이어는 네트워크의 상태에 따라 선별적으로 전송되게 된다.

• Journal of Korea Multimedia Society
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• v.12 no.3
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• pp.419-426
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• 2009

The WiMedia Alliance has specified a Distributed Medium Access Control (D-MAC) protocol based on UWB for high speed wireless home networks and WPANs. The WiMedia D-MAC fundamentally removes the problems of the centralized MAC revealed at IEEE 802.15.3 MAC, by adopting a distributed architecture. However, there is no completely distributed method for allocating fair data rates to all traffic streams by considering each stream's QoS parameter. In this paper, a novel fair distributed QoS admission control method is proposed. The proposed method is a fair, adaptive QoS provisioning method, by allocating time slots to devices according to the current traffic load condition, through executing only a proposed single satisfaction of QoS (SoQ) algorithm at each device.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.398-398
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• 2004

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.05a
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• pp.384-387
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality if the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management algorithms and this new algorithm in various network settings.

• Proceedings of the KIEE Conference
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• 2004.05a
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• pp.68-70
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• 2004

In this paper we develop an adaptive interface between video compression and transission protocols to handle QoS fluctuations that are common to mobile communication systems. We consider various generic design alternatives for QoS adaptation and identify 'QoS negotiation' as the most promissing. This method gives the best possibilities to obtain system-wide efficiency. To handle the indued system complexity we apply a design philosophy (called ARC) that separates implementation dependencies by introducing QoS interfaces between system modules. In the ARC phlosophy the implementation details are hidden in the subsystems. To assure efficient adaptation, the QoS must be negotiated between modules. We select the QoS parameters that are both necessary and sufficient for efficient negotiation between the video encoder and protocol modules. We describe the relation between the QoS parameters at the interface and the internal parameters of common video coding methods and protocol elements. Furthermore, we describe a negotiation procedure that allows a system-wide optimum to emerge.

• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.55-62
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• 2014

This paper has proposed Hybrid QoS Routing Link State Update(LSU) Algorithm that has had a both advantage of LSU message control in periodic QoS routing LSU algorithm and QoS routing performance in adaptive LSU algorithm. Hybrid LSU algorithm can adapt the threshold based network traffic information and has the mechanism that calculate LSU message transmission priority using the flow of statistical request bandwidth and available bandwidth and determine the transmission of the message according to update rate per a unit of time. We have evaluated the performance of the proposed algorithm and the existing algorithms on MCI simulation network using the performance metric as the QoS routing blocking rate and the mean update rate per link, it thus appears that we have verified the performance of this algorithm that it can diminish to 10% of the LSU message count.

• Journal of KIISE:Information Networking
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• v.34 no.4
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• pp.239-245
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• 2007

New multimedia services over the cellular/WLAN overlay networks require different Quality of Service (QoS). In the cellular/WLAN overlay structure, the main issue of load balancing is to balance the available bandwidth among different networks. In this paper, we propose a new online network management scheme. Our proposed scheme is designed to improve bandwidth utilization by balancing the traffic load between the cellular network and WLANs. Simulation results indicate the superior performance of our proposed framework to strike the appropriate performance balance between contradictory QoS requirements under widely varying diverse traffic loads.