• Journal of Korea Multimedia Society
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• v.5 no.6
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• pp.697-703
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• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.5
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• pp.937-941
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• 2016

This paper proposed an efficient multiple TCP mechanism using Android smartphones for remote control video Wi-Fi stream transmission via network communications in real time. The wireless video stream transmission mechanism can be applied in various area such as real time server stream transmissions, movable drones, disaster robotics and real time security monitoring systems. Moreover, we urgently need to transmit data in timely fashion such as medical emergency, security surveillance and disaster prevention. Our parallel TCP transmission system can play an important role in several area such as real time server stream transmissions, movable drones, disaster robotics and real time security monitoring systems as mentioned in the previous sentence. Therefore, we designed and implemented a parallel TCP transmission (parallel stream) for an efficient real time video streaming services. In conclusion, we evaluated proposed mechanism using parallel TCP transmission under various environments with performance analysis.

• The Transactions of the Korea Information Processing Society
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• v.7 no.7
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• pp.2050-2059
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• 2000

In a video-on-Demand(VOD) service, a server has to return to he normal playback quickly at a certain new frame position after interactive operations such as jump or last playback. In this paper, we propose an efficient scheme to write a transmission schedule for a playback restart of a video stream at a new frame position after interactive operations. The proposed scheme is based on convergence characteristics, that is transmission schedules with different playback startup frame position in a video stream meet each other at some frame position. The scheme applies a bandwidth smoothing from a new frame position to a convergence position without considering all remaining frames of a video stream. And then the scheme transmits video dta according to the new schedule from the new frame position to the convergence position, and then transmits the remaining video data according to the reference schedule from the convergence position, and then transmits the remaining video data according to the reference schedule from the convergence position to the last frame position. In this paper, we showed that there existed the convergence position corresponding to nay frame position in a video stream through many experiments based on MPEG-1 bit trace data. With the convergence we reduced the computational overhead of a bandwidth smoothing, which was applied to find a new transmission schedule after interactive operations. Also, storage overhead is greatly reduced by storing pre-calculated schedule information up to the convergence position for each I frame position of a video stream with video data off-line. By saving information on a transmission schedule off-line along with the video data and searching the schedule corresponding to the specified restarting frame position, we expect the possibility of normal playback of a video stream with small tolerable playback startup delay.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1355-1359
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• 2004

The two way subscriber transmission systems have tendency to spread its carrier frequency bandwidth or information bit rate and average bit error rate according to popularization of high speed information through the digital communication system, transmission medium and the Internet. This fact is an important incentive to realize new systems. These two way subscriber transmission systems usually use same cable or same carrier frequency bandwidth for up stream channel and down stream channel. In the systems, the disturbances of noise, crosstalk or fading affect the characteristics. Specifically, these disturbances cause the decrease of information bit rate and degradation of transmission quality. This paper proposes the improved method of their degradations using the particular feature of two way subscriber transmission systems and it makes clear proposed method is effective by theoretically and some numerical examples.

• The KIPS Transactions:PartA
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• v.9A no.2
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• pp.207-216
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• 2002

This paper present a design and implementation of a distributed audio, Video stream service framework based on CORBA for efficient processing and control of audio/video stream. We design software components which support processing, control and transmission of audio/video streams as distributed objects. For optimization of stream transmission performance, we separate the transmission path of control data and media data. Distributed objects are defined by IDL and implemented using JAVA. And device dependent facilities like media capturing, playing and communication channels are implemented using JMF (Java Media Framework) components. We show a connection establishment and control procedure of streams communication. And for evaluation, we implement a test system and experiment a system performance. Our experiments show that test system has somewhat longer connection latency time compared to TCP connection establishment, but has optimized media transmission time compared to CORBA IIOP. Also test system show acceptable service quality of media transmission.

• The Transactions of the Korea Information Processing Society
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• v.7 no.6
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• pp.1766-1777
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• 2000

In multimedia new service on the internet, there are problems such as server overload, network congestion and initial latency. To overcome these problems, we propose a proxy push-caching scheme that stores a portion of continuous media stream or entire stream, and a transmission scheme of NOD continuous media, RTP-RR and RTP-nR to exploit push-caching scheme. With the proposed push-caching scheme, NOD server pushes fixed portion of stream to a proxy when new data is generated, and the cached size of each stream changes dynamically according to the caching utility value of each stream. As a result, the initial latency of client side could be reduced and the amount of data transmitted fro ma proxy server to client could be increased. Moreover, we estimate a caching utility value of each stream using correlation between disk space occupied by the stream and the amount of data stream requested by client. And we applied the caching utility value ot replacement policies. The performance of the proxy push-caching and continuous media transmission schemes proposed were compared with other schemes using simulations. In the simulation, these schemes show better results than other schemes in terms of BHR (Byte Hit Rate), initial latency, the number of replacement and packet loss rate.

• Journal of Internet Computing and Services
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• v.20 no.2
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• pp.21-27
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• 2019

The terrestrial ATSC 3.0 broadcasting system, which is capable of converging broadcast and communication services, uses IP based technology for data transmission between broadcasting equipment. In addition, data transmission between broadcasting equipment uses IP-based technology like existing wired communication network, which has advantageous in terms of equipment construction and maintenance In case IP based data transmission technology is used, however, it may inevitably cause an error that a packet is lost during transmission depending on the network environments. In order to cope with a broadcasting accident caused by such a transmission error or a malfunction of a broadcasting apparatus, a broadcasting system is generally configured as a duplication, which can transmit a normal packet when various types of error may occur. By this reason, correction method of error packets and intelligent switching technology are essential. Therefore, in this paper, we propose a design and implementation of intelligent IP switch for Ensuring Reliability of ATSC 3.0 Broadcast Streams. The proposed intelligent IP consists of IP Stream Analysis Module, ALP Stream Analysis Module, STL Stream Analysis Module and SMPTE 2022-1 based FEC Encoding/Decoding Module.

• Journal of Communications and Networks
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• v.5 no.3
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• pp.258-265
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• 2003

Transmission of continuous media streams has been a challenging problem of multimedia service. Lots of works have been done trying to figure out the best solution for this problem, and some works presented the optimal solution for transmitting the stored video using smoothing schemes applied to each individual stream. But those smoothing schemes considered only one stream, not the whole streams being serviced, to apply themselves, which could only achieve local optimum not the global optimum. Most of all, they did not exploit statistical multiplexing gain that can be obtained before smoothing. In this paper, we propose a new smoothing scheme that deals with not an individual stream but the whole streams being serviced simultaneously to achieve the optimal network bandwidth utilization and maximize the number of streams that can be serviced simultaneously. We formally proved that the proposed scheme not only provides deterministic QoS for each client but also maximizes number of clients that can be serviced simultaneously and hence achieves maximum utilization of transmission bandwidth.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.600-602
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• 1998

RFC 959 정의한 FTP 의 세가진 전송모드, Stream, Block, Compressed전송 모드 중 현재 주로 사용되고 있는 전송 모드는 Stream 전송모드인데. 이 Stream전송모드에서 서버 또는 네트웍에 치명？인 오류 발생하였을 경우 이를 체크 할 수 없고 서버에 재접속하였을 겨우 처음부터 재전송 하영햐 하는 문제점이 발생한다. 이를 개선한 Block, Compressed전송 모드의 경우, 데이터 스트림에 정보 헤더를 추가하여야 함으로써 오버 헤드를 발생한다. 본 논문에서는 클라이언트 FTP상에서 전송 상태를 체크하게 함으로서 전송 중단시 중단된 시점부터 나머지 데이터를 보냄으로써 FTP의 스트림 전송 모드의 문제점을 개선하는 방법을 제시한다.