• 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.

• Proceedings of the Korea Database Society Conference
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• 1997.10a
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• pp.319-334
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• 1997

불과 2년 전만 하더라도 인터넷에서 오디오 또는 비디오 데이터를 감상하기까지 일련의 과정은 그 데이터의 물리적인 양과 전달방식에 있어서 이용자들에게 많은 인내력을 요구해 왔다. 이에 대한 해결책으로 관련업계에서는 real-time streaming 기술을 도입하여 각종 비디오와 오디오 데이터 전송에 관련기술을 적용시킴으로서 실시간 비디오/오디오 서비스 이용을 용이하게 하려는 움직임이 활발히 진행되어 왔었다.(중략)

• International Journal of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.26-34
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• 2018

In this paper, we propose a scheduling scheme that can efficiently serve I/O requests having deadlines in flash storage. The I/O requests with deadlines, namely, real-time requests, are assumed to be issued for streaming services of continuous media. Since a Web-based streaming server commonly supports downloads of HTMLs or images, we also aim to quickly process non-real-time I/O requests, together with real-time ones. For this purpose, we adopt the well-known rate-reservation EDF (RR-EDF) algorithm for determining scheduling priorities among mixed I/O requests. In fact, for the use of an EDF-style algorithm, overhead of task's switching should be low and predictable, as with its application of CPU scheduling. In other words, the EDF algorithm is inherently unsuitable for scheduling I/O requests in HDD storage because of highly varying latency times of HDD. Unlike HDD, time for reading a block in flash storage is almost uniform with respect to its physical location. This is because flash storage has no mechanical component, differently from HDD. By capitalizing on this uniform block read time, we compute bandwidth utilization rates of real-time requests from streams. Then, the RR-EDF algorithm is applied for determining how much storage bandwidth can be assigned to non-real-time requests, while meeting deadlines of real-time requests. From this, we can improve the service times of non-real-time requests, which are issued for downloads of static files. Because the proposed scheme can expand flexibly the scheduling periods of streams, it can provide a full usage of slack times, thereby improving the overall throughput of flash storage significantly.

• Journal of the Korean Society of Surveying, Geodesy, Photogrammetry and Cartography
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• v.34 no.6
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• pp.569-578
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• 2016

With the recent advancement of surveying and technology, the spatial data acquisition rates and precision have been improved continually. As the updates of spatial data are rapid, and the size of data increases in line with the advancing technology, the LOD (Level of Detail) algorithm has been adopted to process data expressions in real time in a streaming format with spatial data divided precisely into separate steps. The existing GRID analysis utilizes the single DEM, as it is, in examining and analyzing all data outside the analysis area as well, which results in extending the analysis time in proportion to the quantity of data. Hence, this study suggests a method to reduce analysis time and data throughput by acquiring and analyzing DEM data necessary for GRID analysis in real time based on the area of analysis and the level of precision, specifically for streaming DEM data, which is utilized mostly for 3D geographic information service.

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.327-330
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• 2007

This article discusses a capacity planning method in QoS-guaranteed IP networks such as BcN (Broadband convergence Network). Since IP based networks have been developed to transport best-effort data traffic, the introduction of multi-service component in BcN requires fundamental modifications in capacity planning and network dimensioning. In this article, we present the key issues of the capacity planning in multi-service IP networks. To provide a foundation for network dimensioning procedure, we describe a systematic approach for classification and modeling of BcN traffic based on the QoS requirements of BcN services. We propose a capacity planning framework considering data traffic and real-time streaming traffic separately. The multi-service Erlang model, an extension of the conventional Erlang B loss model, is introduced to determine required link capacity for the call based real-time streaming traffic. The application of multi-service Erlang model can provide significant improvement in network planning due to sharing of network bandwidth among the different services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9B
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• pp.838-847
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• 2006

In order to support multiple multimedia streaming services in home networks, many critical issues must be considered. In addition, handling the shortage of network bandwidth is one of the most significant and complicated issues. In this paper, real-time bandwidth management service is suggested as a solution to the problem regarding the IEEE1394-based home network. In order to handle the shortage of network bandwidth and to enhance the bus utilization rate, the proposed service combines two methods. First, the bus bandwidth management function determines the state of the network bandwidth and restores the residual bandwidth, which is excessively occupied by a streaming service, to the available free bandwidth. Second, the Isochronous Streaming (IS) Scheduler manages all streaming services according to priority. In order to test the proposed service, we implemented a prototype steaming management middleware and evaluated it by using the IEEE1394 network test-bed.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2022.05a
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• pp.524-526
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• 2022

Considering QoS only considering real-time multimedia service, it is possible to adjust the number of terminals and ensure them appropriately, but this study considers complex services considering real-time multimedia service and general data service. Since the amount of physical network resources is limited, the guarantee of the desired QoS can not be achieved unless the appropriate CAC is done. However, given the traffic profile and QoS spec of the entire network resource and the current service being provided, and the traffic profile and QoS spec of the newly requested service, it is quite difficult to determine exactly whether the new service request is acceptable from this. To do this, it is necessary to study in various directions from mathematical analysis to various simulations and statistical research based on data obtained from actual network operation.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.118-123
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• 2009

In this paper, we propose packet selection and significance based interval allocation algorithm for real-time streaming service. In real-time streaming of inter-frame (and layer) coded video, minimizing packet loss does not imply maximizing QoS. It is true that packet loss adversely affects the QoS but one single packet can have more impact than several other packets. We exploit the fact that the significance of each packet loss is different from the frame type it belongs to and its position within GoP. Using packet dependency and PSNR degradation value imposed on the video from the corresponding packet loss, we find each packet's significance value. Based on the packet significance, the proposed algorithm determines which packets to send and when to send them. The proposed algorithm is tested using publicly available MPEG-4 video traces. Our scheduling algorithm brings significant improvement on user perceivable QoS. We foresee that the proposed algorithm manifests itself in last mile connection of the network where intervals between successive packets from the source and to the destination are well preserved.

• Journal of KIISE:Software and Applications
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• v.32 no.11
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• pp.1029-1035
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• 2005

Web streaming environment need a suitable video playback service technique, a streaming technique for real-time delivery and an efficient control technique due to its limited environment. In this paper, we propose an effective control technique for real-time media stream playback as a connection technique between the high-quality media player and web browser based on Linux environment. The proposed stream control technique presents control data through 050 on media playback screen of web browser, and process the user's input data. It acts GUI-independently, and is designed considering addition and extension for control data.

• Journal of Digital Contents Society
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• v.15 no.3
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• pp.357-363
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• 2014

In this paper, we propose a new audio forensic marking system for protecting the copyright of the Internet-based music streaming services. In the proposed method, in order to guarantee the QoS of the streaming service, high speed, and generates a forensic mark inserted MP3 file. We make pre-marking process and generate a new forensic marked MP3 file, a combination of the pre-marked MP3 frame, the inserted user information. Experimental results show that the proposed method satisfactory results robustness and imperceptibility, and real-time properties. In addition, we were confirmed that the real-time embedding and detection from the streaming-based audio forensic marking system that has been implemented on the server/client is possible.