• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.155-158
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• 2003

This paper proposes a quality-of-service(QoS)-adaptive proxy-switching scheme for multimedia streaming over the Internet. characteristic of multimedia-object about that effect on weight and transport-time causes Streaming Application is restricted to high-quality-service. But using of Proxy or Cache can reduce restraint of media stream. However previous scheme propose only structure of proxy and skill of streaming-object. It couldn't propose proper scheme about limited cache and proxy error occasion. We propose adaptive-proxy-switching mechanism for solution about some problems that caused by single-proxy and limited cache in case of streaming service.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• Journal of Broadcast Engineering
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• v.10 no.4 s.29
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• pp.548-556
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• 2005

In wireless network environments, wireless channels are characterized by time-varying fading and interference conditions, which may lead to burst packet corruptions and delay variation. This can cause severe quality degradation of streaming media. To guarantee successful transmission of media over the hostile wireless networks, where channel conditions are highly fluctuating, a flexible and network-adaptive transport method is required. Thus, we propose a network-adaptive transport error control consisting of packet-level interleaved FEC and delay-constrained ARQ, which acts as an application-level transport method of streaming media to alleviate burst packet losses while adapting to the changing channel condition in wireless networks. The performances of the proposed network-adaptive transport error control, general error control schemes, and hybrid schemes are evaluated by a developed simulator at the transport-level and video quality of streaming media. Simulation results show that the proposed mechanism provides the best overall performance among compared other schemes in terms of the transport-level performance of error control and the performance of video quality for streaming media.

• Journal of Korea Multimedia Society
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• v.7 no.3
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• pp.368-376
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• 2004

Currently, the package tools and software development kits of Microsoft's Windows Media Technology provides users an environment to write, distribute and play multimedia contents. However, it is very difficult for common users to create a media streaming content using the Windows Media technology, because of its technical hardship. Therefore, we developed an easy-to-use authoring tool for multimedia streaming content which can be used in the Windows Media Technology environment. Our authoring tool is based on the web page templates by which HTML codes can be generated automatically. Using the existing streaming media authoring tool provided by the Microsoft, users have to code the temporal markers and the event script commands directly to the ASF(Advanced Systems Format) files or ASX (ASF Stream Redirector) format files. We developed an easy-to-use interface for insetting temporal markers and the event script commands to the ASF files by simple mouse clicking and dragging to minimize the effort of users for creating multimedia streaming content. In addition, we implemented an ASX file creation wizard that allows users to create ASX format file without coding ASX tags. We also implemented various web page templates where the ASF files and the ASX files can be embedded by just clicking the corresponding templates. The target web pages can be automatically generated from those templates and previewed directly on the web browser at the time of creation.

• Journal of KIISE
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• v.43 no.4
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• pp.470-479
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• 2016

As the number of various types of media data such as UCC (User Created Contents) increases, research is actively being carried out in many different fields so as to provide meaningful media services. Amidst these studies, a semantic web-based media classification approach has been proposed; however, it encounters some limitations in video classification because of its underlying ontology derived from meta-information such as video tag and title. In this paper, we define recognized objects in a video and activity that is composed of video objects in a shot, and introduce a reasoning approach based on description logic. We define sequential rules for a sequence of shots in a video and describe how to classify it. For processing the large amount of increasing media data, we utilize Spark streaming, and a distributed in-memory big data processing framework, and describe how to classify media data in parallel. To evaluate the efficiency of the proposed approach, we conducted an experiment using a large amount of media ontology extracted from Youtube videos.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.5
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• pp.275-281
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• 2012

A streaming service is considered as an important fact to listen musics at mobile device in the popular smart-phone era. An apple's HTTP Live Streaming method is used in iOS and a RTSP(Real-time Streaming Protocol) method is used in Android OS. A Wowza Media Sever will be used to support both apple and android mobile streaming. This paper describes a design and implementation of smart-phone app. to record user's songs by userself with smart-phone recording function, upload to the streaming server and to service real-time mobile streaming play for an user's selected song and provide song sharing SNS functions.

• Journal of KIISE:Computing Practices and Letters
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• v.7 no.1
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• pp.24-37
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• 2001

This paper deals with our experience for the design and implementation of Transport and Session managers in the Integrated Streaming Service Architecture (IS SA) that was developed by the authors. The ISSA is a streaming framework that allows to develop integrated multimedia streaming applications such as VOD system in unicast/multicast and real-time broadcastOive-cast). It also facilities standard real-time transport protocols such as RTp(Real-Time Transport Protocol)/RTCP(Real-time Control Protocol) and RTSP(Real-Time Streaming ProtocoD that allows to user openness. The ISSA supports diverse media fonnats and is independent from underlying networks and operating systems, and compatible with the global real-time multimedia database system (BeeHive) so that streaming media are efficiently retrieved, stored, and serviced. The role of the Transport Manager is to do packetization and depacketization for the different types of multimedia data and delivers the packetized media data in real-time. The Transport Manager uses RTP protocol. The role of the Session Manager is to establish and control the media channel by using RTSP protocol and to deliver the database transactions for the multimedia database by using RTTP(Real-Time Transaction Protocol) protocol. Both the Transport and Session Manager are doing their functions through the network interface in the ISSA that allows developers to various network programming interfaces and provides flexibility to the system.

• Journal of Broadcast Engineering
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• v.27 no.6
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• pp.914-922
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• 2022

In this paper, we construct a commercial 4K UHD(Ultra High Definition) streaming device that utilizes a 5G mobile communication network as a transport channel and conduct a streaming performance test. It uses RTP(Realtime Transport Protocol) which has transmission quality monitoring capability as a transmission protocol to apply adaptive streaming. In addition, it provides the function to adjust the encoding rate of the video signal so that encoding can be optimized for the change in the bandwidth of the transmission channel. Through the performance test, it is confirmed that the H.265 encoding rate for 4K UHD signal is 48.69Mbps, the average glass-to-glass delay time is 293.60ms, and the average time difference between video and audio for lip sync is 120ms. With the result of performance test, it is shown that the streaming device is applied to 4K UHD Streaming device over 5G mobile communication network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.517-525
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• 2009

We propose a quality guaranteed scalable video streaming service over the Internet using a new rate adaptation algorithm. Because video data requires much more bandwidth rather than other types of service, therefore, quality of video streaming service should be guaranteed while providing friendliness with other service flows over the Internet. To successfully provide this, we propose a framework for providing quality-guaranteed streaming service using two-channel transport layer and rate adaptation of scalable video stream. In this framework, baseline layer for scalable video is transmitted using TCP transport for minimum qualify service. Enhancement layers are delivered using TFRC transport with layer adaptation algorithm. The proposed framework jointly uses the status of playout buffer in the client and the encoding rate of layers in media contents. Therefore, the proposed algorithm can remarkably guarantee minimum quality of streaming service rather than conventional approaches regardless of network congestion and the encoding rate variation of media content.

• The KIPS Transactions:PartD
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• v.14D no.5
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• pp.555-564
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• 2007

Internet streaming systems consist of a media server, a streaming sewer, and terminals. The media server delivers multimedia contents such as video and/or audio to the streaming server, which distributes the contents to terminals as well. Existing Internet streaming systems have a bottleneck problem in the streaming server because of the limit of the processing capacity of the streaming server and therefore a streaming server can not accomodate more terminals than the limit. As a solution to this problem, P2P streaming systems have been lately proposed and investigated, using P2P distributed architectures. Actually, however, there exist many difficulties in the design and implementation of P2P streaming systems, because it needs many real computers and various network constructions. In this paper, we have proposed and defined a P2P streaming system model such as the architectural model, the timing model, the behavior model, and the performance metrics. And also we have implemented an NS2 based P2P streaming system simulator called P2PStreamSim. Finally, we have verified it through test simulations and analyzed the results.