• Journal of the Korea Computer Industry Society
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• v.3 no.9
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• pp.1251-1260
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• 2002

Real-time application programs have synchronization constraints which need to be met between media-data. Synchronization method represents feedback method including virtual client-side buffer. This buffer is used in buffer level method. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control. It is the important factor for playback rate and QoS if the buffer level is normal or not. To solve the problems, we can control the start of transmission in multimedia server by appling filtering, control and network evaluation function. Synchronization method is processing for smooth presentation without cut-off while media is playing out. When audio frame which is master media is in high threshold buffer level we decrease play out time gradually, otherwise we increase it slowly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.421-430
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• 2006

In this paper, we propose a practical implementation of network-adaptive HD(high definition) MPEG-2 video streaming with a cross-layered channel monitoring(CLM) over the IEEE 802.11a WLAN(wireless local area network). For wireless channel monitoring, AP(access point) periodically measures the MAC(medium access control) layer transmission information and sends the monitoring information to a streaming server. This makes that the streaming server reacts more quickly as well as efficiently to the fluctuated wireless channel than that of the end-to-end monitoring(E2EM) scheme for the video adaptation. The streaming sewer dynamically performs the priority-based frame dropping to adjust the video sending rate according to the measured wireless channel condition. For this purpose, our streaming system nicely provides frame-based prioritized packetization by using a real-time stream parsing module. Various evaluation results over an IEEE 802.11a WLAM testbed are provided to verify the intended QoS adaptation capability The experimental results show that the proposed system can effectively mitigate the quality degradation of video streaming caused by the fluctuations of time-varying wireless channel condition.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.23-26
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• 2003

최근 대두되고 있는 멀티미디어 응용을 지원하기 위해서는 광대역 네트워크의 링크, 실시간 전송, 보다 효과적인 멀티캐스트 프로토콜이 요구된다. 멀티캐스트 프로토콜 알고리즘에는 DVMRP, MOSPF, CBT, PIM 구조의 멀티캐스팅 프로토콜 있다. 이는 네트워크 패킷의 특성을 송수신의 경로비용으로 라우터 자체의 근거리 알고리즘이 적응되고 있다. 본 연구에서는 인터넷에서 멀티캐스트 라우팅 프로토콜인 QosMIC를 제안한다. QosMIC는 멀티캐스트 네트워크에서 신뢰성과 능률적 자원관리와 탄력적인 방법으로 Qos특징을 제공한다 QosMIC는 확인된 다중경로에서 하나의 라우터를 선택하여 경로를 선택하고, 경로에서 요구하는 Qos를 제공한다. QosMiC는 데이터를 미리 제공하는 방식에 있어 이전의 프로토콜과 다르다. 본 연구에서는 QosMIC와 기존의 멀티캐스트 프로토콜을 분석하여 각 데이터 패킷 전송을 위한 공정하고 실용적인 대역폭 사용과 대역폭과 데이터처리능력을 시뮬레이션 상에서 구현하고 성능을 평가한다.

• Journal of KIISE:Computer Systems and Theory
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• v.29 no.7
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• pp.394-400
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• 2002

Networks of Workstations(NOWs), that has features of flexibility and scalability, recently has emerged as an inexpensive alternative to massively parallel multicomputers. However it is not easier to perform deadlock-free multicast than regular topologies like mash or hypercube. Single phase differential multicast(SPDM) is a modified multicast algorithm with less burden of the root node. By applying quality of serviece(QoS), a specific node can have differentiated service and artificial change of message flow pattern is also available. As the results of performance evaluation experiments, SPDM has lower latency and lower packet concentration rate of the root node than the case of SPAM, and has ability to control network load distribution among switch nodes by controlling the assignment rate among nodes.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.5
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• pp.37-43
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• 2009

Streaming video service over wireless and mobile communication networks has received significant interests from both academia and industry recently. Specifically, mobile WiMAX (IEEE 802.16e) is capable of providing high data rate and flexible Quality of Service (QoS) mechanisms, supporting mobile streaming very attractive. However, we need to note that streaming videos can be partially deteriorated in their macroblocks and/or slices owing to errors on OFDMA subcarriers, as we consider that compressed video sequence is generally sensitive to the error-prone channel status of the wireless and mobile network. In this paper, we introduce an OFDMA subcarrier-adaptive mobile streaming server based on cross-layer design. This streaming server system is substantially efficient to reduce the deterioration of streaming video transferred on the subcarriers of low power strength without any modifications of the existing schedulers, packet ordering/reassembly, and subcarrier allocation strategies in the base station.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.490-492
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• 2003

본 논문에서는 3G 이동 통신 시스템과 IEEE 802.11 WLAN 시스템 간의 통합을 위한 프로토콜 계층을 설계한다. 통합 알고리즘을 휘한 핵심은 서로 다른 다양한 시스템의 MAC 계층을 IP 계승을 통해 통합하는 AII IP 구조에서, 이종간 핸드오프 지원을 위한 새로운 적응 계층(Adaptation Layer)을 추가하는 것이다. 적응 계승에서 요구되는 기능들은 끊김 없는 계층적 핸드오프, 시스템의 발견과 최적의 인터페이스 선택. QoS 지원. AAA 등이다. 다양한 이슈들 중에 계층적 핸드오프를 위한 시스템 발견과 최적의 인터페이스 선택을 위한 알고리즘에 대해 살펴보도록 하겠다.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.26 no.4
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• pp.51-54
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• 2015

많은 무선 스마트 단말의 보급으로 무선 데이터 트래픽이 폭발적으로 증가해 왔고, 이로 인한 주파수 자원의 부족은 피할 수 없는 문제가 되었다. 본 과제에서는 획득한 인지정보를 기반으로 하여 다중 라디오 환경에서 주파수 스펙트럼 감지, 동적 스펙트럼 할당, 다중 라디오 선택 및 다중화, 최적 자원 할당, 적응형 전송 기술을 개발하였고, 이 기술들의 개발로 추가적인 주파수 사용 없이 주파수 사용 효율을 최대화함으로써 네트워크 성능 향상을 이끌어 내었다. 본 기술은 고유 주파수 대역 접속, 공통 주파수 대역 접속, 고유/공통 주파수 대역 동시 접속의 경우로 나누어 개발되었다. 개발된 기술은 사용자의 QoS(Quality of Service)를 만족하는 동시에 네트워크 성능 향상을 이끌어낼 수 있음을 모의실험을 통해 검증하였다. 따라서 본 연구의 결과는 4G뿐만 아니라, 5G에서 핵심적인 기술 개발의 중요한 역할을 담당할 것이며, 앞으로도 추가적인 연구가 필요하다.

• Convergence Security Journal
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• v.6 no.1
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• pp.55-63
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• 2006

In this paper, an algorithm that provisions absolute differentiation of packet delays is proposed with an objective for enhancing quality of service (QoS) in future packet networks. It features an adaptive scheme that compensates the deviation for prediction on the traffic to be arrived continuously. It predicts the traffic to be arrived at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot, and derives the deviation between the two quantity. The deviation is utilized to the delay control operation for the next time slot to offset it. As it compensates the prediction error continuously, it shows superior adaptability to the bursty traffic as well as the constant rate traffic. It is demonstrated through simulation that the algorithm meets the quantitative delay bounds and shows superiority to the traffic fluctuation in comparison with the conventional mechanism.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.2
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• pp.455-461
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• 2008

Enhanced Distributed Channel Access (EDCA) is a channel access scheme adopted by the IEEE 802.11e draft standard for QoS-enabled wireless local area networks. It classifies traffic into separate Access Categories (ACs) and achieves service differentiation by allowing each AC to have its own values of channel access parameters. This paper proposes a scheme to dynamically adapt the EDCA parameters to traffic environment so that they both maximize the throughput of non real-time traffics and meet the delay and throughput constraints of real-time traffics. For this purpose, we develop a design algorithm for efficient exploration of the EDCA parameter space. Using the algorithm, we derive a Pareto curve fur delay-throughput trade-off in each anticipated traffic environment. The Pareto database can then be used to optimally adjust the parameter according to the traffic environment and administrative policies. Simulation results show that compared with the default parameter configuration, the proposed scheme is better for delay, throughput guarantee and can improve the throughput of non real-time traffics by 12% on average.