• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8A
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• pp.859-867
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• 2004

To provide high-quality media streaming service over the best-effort Internet, a streaming methodology is required to response to the dynamic fluctuation of underlying networks. In this paper, we implement the MPEG-4 streaming system with adaptive transport based on priorities of media packets. The implemented system is composed of the common MPEG-4 streaming components such as elementary stream provider, sync and DMIF layer, and adaptive transport module including data prioritization and FEC control. More specifically, the prioritized sync layer packets (based on object level) are delivered to the transport module and then are encoded by an adaptive FEC encoder to help reliable transport. The FEC combination is dynamically adjusted by the feedback information from the receiver. In addition, low priority packets are selectively dropped to meet the limitation of available bandwidth. The experimental results over the emulation-based testbed show that the Proposed system can mitigate the impact of network fluctuation and thus improve the quality of streaming.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.6
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• pp.1048-1057
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• 2015

In this paper, we propose a fast MAS reallocation method that can avoid reservation conflicts in the DRP process of WiMedia system in real time. WiMedia MAC DRP has been developed aimed QoS support for audio and video transmission that generate traffic periodically. The size variation of the video streaming data causes DRP reservation conflicts frequently, so that DRP could not support QoS of the video streaming service which is delay-sensitive. Fast MAS reallocation method could avoid reservation conflicts in real-time analyzing reservation requests information of the other device. Our proposed method is evaluated with test sequences using the software simulation and implemented in hardware/software with the integrated WiMedia MAC/PHY system.

• The KIPS Transactions:PartB
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• v.12B no.7 s.103
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• pp.783-788
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• 2005

Recently researches that apply P2P networks to VOD streaming have been actively published In the previous works on VOD streaming, they aimed at achieving two major goals, which are zero-delay transmission and minimization of service rejection ratio. This paper proposes a new method, called ForwardCast, for VOD streaming based on a P2P-based multicasting tree in order to achieve theses two goals. In this method,- basically a new client selects one of the preceding clients and starts receiving a whole video from the selected one without any delay. In some situation, two preceding clients are selected to transfer the ending part of the video and its the remaining part simultaneously. In our experiment, ForwardCast can reduce the rejection ratio compared to previous works without increasing server stress.

• Journal of the Korea Society of Computer and Information
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• v.18 no.8
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• pp.53-64
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• 2013

The existing research on the QoS of the Streaming video system(SVS) adjust the video stream being sent from the network perspective, the focus is to prevent data loss and delay. The other hand, changing the quality of the video stream will not receive the guaranteed QoS from the perspective of the users who want uniform quality. In order to solve these problems, this paper applied to ensure that the benefits in the amount of bits per frame occurs Encoder CBR streaming video is kept constant and uniform picture quality advantages of VBR streaming video VBR to CBR Encoder and CBR to VBR Decoder, Video was designed to control the playback stream for And to ensure a uniform quality of the user based on the design and implementation of uniform quality guaranteed SVS. PSNR evaluated for several characteristics of the sample video to demonstrate the superiority of the SVS ensure uniform quality, the proposed Performance evaluation of the SVS ensure uniform quality CBW the ABR from 100% to ensure uniform image quality from 85% to environmental excellence is proved.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.11b
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• pp.319-322
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• 2002

MPEG-4 파일 포맷은 사용자에게 다양한 기능을 제공하고 다수의 데이터를 유연성있게 제어하기 위해 많은 양의 메타데이터를 정의하고 있고, 이에 접근하는 서버와 로컬 시스템은 이들 메타데이터를 분석하는데 오랜 시간을 소비해야 한다. 본 논문은 이런 분석 지연을 없애기 위해 MPEG-4 파일에서 메타데이타를 실시간으로 분석하여 초기 지연 시간을 줄이는 방법을 제시하며, 그 결과를 보인다.

• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.6-16
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• 2008

An instable wireless channel condition causes more packet losses and retransmissions due to interference, fading, station mobility, and so on. Therefore video streaming service over a wireless networks is a challenging task because of the changes in the wireless channel conditions and time-constraints characteristics of the video streaming services. To provide efficient video streaming over a wireless networks, QoS-enhanced MAC protocol, IEEE 802.11e, is standardized recently. Tn this paper, we propose a new network-adaptive H.264 video streaming mechanism in the IEEE 802.11e wireless networks. To improve the quality of video streaming services, video stream has to adapt to the changes in the wireless channel conditions. The wireless channel conditions are estimated by the packet loss probability and informed to the application layer by the cross-layering. According to the wireless channel information, the video streaming application filters out the low-priority data. This adaptation mechanism efficiently uses system resources because it drops the low-priority data in advance. Therefore, our cross-layer design can provide improved video streaming services to the end-user. Through the implementation and performance evaluation, we prove that the proposed mechanism improves the QoS of the video streaming by providing the smoothed playback.

• Journal of Digital Contents Society
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• v.17 no.6
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• pp.581-592
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• 2016

WebRTC has quickly grown to be the world's advanced real-time communication in several platforms such as web and mobile. In spite of the advantage, the current technology in WebRTC does not handle a big-streaming efficiently between peers and a large amount request of users on the Signaling server. Therefore, in this paper, we put our work to handle the problem by delivering the flow of data with dynamical load balancing algorithms. We analyze the request source users and direct those streaming requests to a load balancing component. More specifically, the component determines an amount of the requested resource and available resource on the response server, then it delivers streaming data to the requesting user parallel or alternately. To show how the method works, we firstly demonstrate the load-balancing algorithm by using a network simulation tool OPNET, then, we seek to implement the method into an Ubuntu server. In addition, we compare the result of our work and the original implementation of WebRTC, it shows that the method performs efficiently and dynamically than the origin.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.501-510
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• 2006

Software streaming allows the execution of stream-enabled software on desktop or portable computing devices like PC, PDA, laptop, cellular phone, etc., even while the transmission/streaming from the server may still be in progress. In this paper, we present an efficient streaming system called Software On-Demand(SOD) streaming system to transmit stream-enabled applications in addition to automatic installation of program registry, environment variables, configuration files, and related components. In particular, we design and implement a SOD system in Linux to provide the user with the instant look-and-click software execution environment such that software download and installation are internally proceeded in a completely user-transparent way. Therefore, the SOD system relieves the user from the tricky, failure-prone installation business. In addition, the software developer now obtains a new, powerful means to advertise and propagate their software products since the user can use software packages via user-friendly UI window or web browser by look-and-click interactive operation. In the paper, we also make a couple of SOD streaming experiments using a spectrum of popular softwares. Based on the analysis of the experiment results, we also propose two performance improvement schemes.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.9
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• pp.626-636
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• 2009

In Mobile IP-enabled wireless LAN (WLAN), packet flows are corrupted due to the handoff of a mobile node (MN) at the link and network layers, which results in burst packet losses and can cause temporary buffer underflow in a streaming client at the MN. This transient behavior hurts time-sensitive streaming media applications severely. Among many suggestions to address this handoff problem, few studies are concerned with empirical issues regarding the practical validation of handoff options on the time-sensitive streaming media applications. In this paper, targeting seamless streaming over Mobile IP-enabled WLAN, we introduce a seamless media streaming framework that estimates accurate pre-buffering level to compensate the handoff latency. In addition, we propose a link-layer (L2) assisted seamless media streaming system as a preliminary version of this framework. The proposed system is designed to reduce the handoff latency and to overcome the playback disruption from an implementation viewpoint. A packet buffering and forwarding mechanism with L2 trigger is implemented to reduce the handoff latency and to eliminate burst packet losses generated during the handoff. A pre-buffering adjustment is also performed to compensate the handoff latency. The experimental results show that the proposed approach eliminates packet losses during the handoff and thus verify the feasibility of seamless media streaming over Mobile IP-enabled WLAN.