• 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.

• Journal of Digital Convergence
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• v.19 no.4
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• pp.251-257
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• 2021

With the development of new media, great changes are taking place in the way people get information. The change is the use of video content that can deliver content in a more three-dimensional way than words or photos. After 2016, the number of live video streaming content providers and users has increased. In this paper the write takes the 1 personal live video streaming content as the research object. And the write takes live video streaming content on YouTube live or Douyu TV as a research example. In this paper, the writer analyzes the digital information content in the live video streaming case. And the writer expounds the necessity of these visual information and the characteristics of real-time live video streaming content. Especially since 2020, because of the influence of the COVID-19, the live video streaming industry has begun to combine with the traditional industry. It is expected that the integration of digital cutting-edge technology and live video streaming will not only provide diversity in the content, but also create more social value for the video content consumption culture. Therefore, The writer thinks it is necessary to conduct in-depth research on the social responsibility of real-time live content in the future.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.380-389
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• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.3
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• pp.603-612
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• 2010

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous VoD system researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose greedy precedent frame transmission technique, GPFT, for guarantee of constant QoS. In GPFT, Streaming VoD server prefetches precedent frames and transmits the frame greedily by increasing the frame transmission rate while the available network bandwidth is high. The GPFT uses the prefetched precedent frames to guarantee the QoS while the available network bandwidth is low. The experiment result shows that the proposed GPFT could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• Journal of information and communication convergence engineering
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• v.14 no.1
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• pp.21-25
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• 2016

N-screen is a promising technology to improve support for multimedia multicasting, content sharing, content mobility, media scalability, and seamless mobility. In this paper, the WiMedia distributed-MAC (D-MAC) protocol is adopted for development of a seamless N-screen wireless service. Furthermore, to provide a multi-hop, one source multi-use N-screen service through point to point streaming in a seamless D-MAC protocol, a dynamic multi-frame transmission technology is proposed. In this technology, a dynamic time slot allocation scheme and a multi-hop resource reservation scheme are combined. In the proposed dynamic time slot allocation scheme, two thresholds, a hard threshold and a soft threshold, are included to satisfy the power consumption and delay requirements. A multi-frame DRP reservation scheme is proposed to minimize end-to-end delay during the multi-hop transmissions between N-screen devices. The proposed dynamic multi-frame transmission scheme enhances N-screen performance in terms of the multi-hop link establishment success rate and link establishment time compared to the conventional WiMedia D-MAC system.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2005.11a
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• pp.155-158
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• 2005

본 논문에서는 대화형 기능을 가지는 MPEG-4기반의 멀티미디어를 IP 네트워크를 통해 전송할 수 있는 대화형 멀티미디어 스트리밍 시스템을 제안한다. 제안한 대화형 멀티미디어 시스템은 MPEG-4 콘텐츠를 분석, 전송할 수 있는 스트리밍 서버와 전송되는 MPEG-4 콘텐츠를 수신, 재생할 수 있는 클라이언트, 그리고 효율적인 멀티미디어의 전송을 위해 도입한 오버레이 네트워크로 구성된다. 오버레이 네트워크는 실제 IP 네트워크 상에 콘텐츠를 중계하는 릴레이 노드와 릴레이 노드들을 관리하는 네트워크 분배 관리자로서 구성된다.

• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.121-123
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• 1998

본 논문에서는 자바를 사용한 고화질 MPEG-1 데이터를 전송하는 멀티캐스트 스트리밍 시스템의 구현을 기술한다. 서버는 다 채널의 MPEG-1 스트림 전송을 지원하며 각 채널의 QoS를 보장하기 위한 스케쥴링 정책을 사용한다. 서버 구현에는 실시간을 지원하는 Jave Peal Time 패키지를 사용하였고, 클라이언트는 Sun사의 Java Media Framework 패키지를 사용하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06a
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• pp.209-210
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• 2008

• Proceedings of the Korean Information Science Society Conference
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• 2010.06a
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• pp.198-199
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• 2010

• Proceedings of the Korean Information Science Society Conference
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• 2007.06a
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• pp.135-136
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• 2007