• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1104-1109
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• 2001

In this paper, we describe a method for real-time transmitting video streams based on RTP. In order to guarantee synchronous video streams, we propose a method to grasp network situations by analyzing end-to-end network traffic. In addition, we present an algorithm for satisfying QoS requirements of real-time multimedia transmission and maintaining continuity of transmission. It describes a buffering method for overcoming bandwidth limitations and an analyzing method based on RTCP for grasping network traffic situation to resolve the problem of real-time transmission of video streams.

• 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.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.872-875
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• 2002

A digital watermarking technique applying psychoacoustic model for audio signal is proposed in this paper. In the watermarking scheme, the pseudo-random bit stream used as a watermark signal is embedded into the audio signal in both speech and music. The strength of the embedded signal is subject to the human auditory system in such a way that the disturbances on host audio signal are beyond the sensing of human ears. The experimental results show that the quality of the watermarked audio signal, in term of signal to noise ratio, can be improved up to 3.2 dB.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.4
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• pp.1-10
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• 1994

In the real-time delivery of multimedia data streams over networks, the interruption of continuity in a single media stream and the mismatching of the data within the same time interval in multimedia data streams transfered in paralled on different channels are considered as the most serious synchronization problems. There are several mechanisms proposed to handle these problems. In this paper, these mechanisms are analyzed and compared in various point of view by the computer simulation. According to the simulation results, it has been shown that the method which uses the segmentation and the method which uses the seperate synchronization channel are superior to the method which uses the synchronization marks in view of the real-time transmission and quality of sevice. On the other hand, it can be seen that the method which uses the segmentation is superior to the method which uses the seperate synchronization channel from a channel utilization's point of view.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.5
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• pp.324-326
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• 2005

An adaptive Chunk algorithm is newly devised and a collaborative streaming is designed for high quality multimedia streaming service under time varying traffic conditions. An LMS based prediction filter is used to compensate the effect of time varying background traffic of the WAN. The underflow is generated for the $20\~28\%$ of the data stored in the central server by applying the FARIMA(Fractional Autoregressive Integrated Moving Average) traffic modeling method. The proposed algorithm is tested with the MPEG-2 video files and compensates $71\~85\%$ of central stream underflow.

• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.17-24
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• 2008

At the view of Application layer, there are many researches to achieve the cross-layer optimization with Physical layer. Scalable video coding is good example. Although it is necessary to consider the session layer which lies halfway between two layers, the research about that is insufficient. We present a feasible solution of dynamic session control for scalable video coding over IMS.

• Journal of Korea Multimedia Society
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• v.11 no.6
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• pp.807-815
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• 2008

A simple efficient method of spotting and recognizing hand gestures in video is presented using a network of hidden Markov models and dynamic programming search algorithm. The description starts from designing a set of isolated trajectory models which are stochastic and robust enough to characterize highly variable patterns like human motion, handwriting, and speech. Those models are interconnected to form a single big network termed a spotting network or a spotter that models a continuous stream of gestures and non-gestures as well. The inference over the model is based on dynamic programming. The proposed model is highly efficient and can readily be extended to a variety of recurrent pattern recognition tasks. The test result without any engineering has shown the potential for practical application. At the end of the paper we add some related experimental result that has been obtained using a different model - dynamic Bayesian network - which is also a type of stochastic model.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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• pp.1279-1280
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• 2010

There are some cases when SOAP message, where WS-Security and WS-Policy elements are included, may consist of a sensitive and important data. In these cases, the message is highly recommended to be secured. The question exists of how to quickly identify that SOAP message satisfies security requirement and security level of a SOAP message. In this paper, we propose a technique called Bit-Stream which depends on the importance of SOAP elements to automatically identify the vulnerabilities and risks while offering advice for higher security.

• 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 the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.