• 한국멀티미디어학회논문지
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• 제21권8호
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• pp.942-951
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• 2018

In these days, 360 degree videos, which is provided via VR, have high resolution and quality of 8K. These kinds of videos inevitably require streaming technology which guarantees QoS. Therefore, we suggest MPEG DASH HTTP protocol which stably provides streaming services of high quality videos in 8K, which are 360 degree tile-encoded, on low-spec devices such as OTT and IPTV Settop and Tile Segment Cache management structure for servers operating streaming services.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제11권1호
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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.

• ETRI Journal
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• 제26권3호
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• pp.203-217
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• 2004

This paper presents a method and architecture to analyze streaming media and multimedia conferencing traffic. Our method is based on detecting the transport protocol and port numbers that are dynamically assigned during the setup between communicating parties. We then apply such information to analyze traffic generated by the most popular streaming media and multimedia conferencing applications, namely, Windows Media, Real Networks, QuickTime, SIP and H.323. We also describe a prototype implementation of a traffic monitoring and analysis system that uses our method and architecture.

• 한국정보과학회:학술대회논문집
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• 한국정보과학회 2012년도 한국컴퓨터종합학술대회논문집 Vol.39 No.1(D)
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• pp.401-403
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• 2012

In this paper, we introduce a multipath congestion control algorithm for audio/video streaming, called MPSQRT. MPSQRT is derived from single-path Square-Root TCP (SQRT TCP) designed for multimedia streaming, where a lower variation of sending rate is important. Based on the fluid model of SQRT for single-path, we extend it towards spreading concurrently packets across multiple paths ensuring load-balancing and fairness to SQRT at shared bottleneck. Through simulations, we evaluate the proposed protocol under various network conditions.

• 인터넷정보학회논문지
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• 제21권4호
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• pp.51-57
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• 2020

Currently, High-efficient video coding (HEVC) has become the most promising video coding technology. However, the implementation of HEVC in video streaming systems is restricted by factors such as cost, design complexity, and compatibility with existing systems. While HEVC is considering deploying to various systems with different reached methods, H264/AVC can be one of the best choices for current video streaming systems. This paper presents an adaptive method for manipulating video streams using video coding on an integrated circuit (IC) designed with a private network processor. The proposed system allows to transfer multimedia data from cameras or other video sources to client. For this work, a series of video or audio packages from the video source are forwarded to the designed IC via HDMI cable, called Tx transmitter. The Tx processes input data into a real-time stream using its own protocol according to the Real-Time Transmission Protocol for both video and audio, then Tx transmits output packages to the video client though internet. The client includes hardware or software video/audio decoders to decode the received packages. Tx uses H264/AVC or HEVC video coding to encode video data, and its audio coding is PCM format. By handling the message exchanges between Tx and the client, the transmitted session can be set up quickly. Output results show that transmission's throughput can be achieved about 50 Mbps with approximately 80 msec latency.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 학술대회지
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• pp.600-607
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• 2004

The paper describes the implementation of the video conferencing system using public key infrastructure which is used for user authentication and media stream encryption. Using public key infrastructure, we are able to reinforce the authentication for conference participant and block several malicious hacking while protecting conference control information. The paper shows the implementation of the transportation layer secure protocol in conformity with Korea public key authentication algorithm standard and symmetric key encryption algorithm (RC2, SEED, DES and 3DES) for media stream encryption. The feature of the paper is transportation layer secure protocol that is implemented for protection of information on a user authentication and video conference and the media streaming encryption algorithm also can be envisioned with another block encryption algorithm. The key for media streaming encryption may be safely distributed by the transportation layer secure protocol.

• 한국정보과학회논문지:컴퓨팅의 실제 및 레터
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• 제8권6호
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• pp.619-629
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• 2002

본 논문은 RTSP에 기반한 MPEG-4 스트리밍 시스템의 구현에 대해 기술한다. MPEG-4 스트리밍 시스템은 컨텐츠 저작 도구, 서버, 클라이언트로 구성된다. 기존에 제공되는 멀티플렉싱 방법에 의한 컨텐츠 스트리밍시 발생하는 지연 시간 및 서버 부하 등의 문제점을 해결하기 위한 새로운 멀티플렉싱 방법을 제안한다. 구현된 시스템은 전송 제어 프로토콜로 RTSP를 사용하여 DMIF를 구현하기 위한 새로운 전송 계층의 구현에 따른 부담을 줄였으며, 로컬 재생뿐만 아니라 스트리밍 재생시에도 사용자 상호 작용 기능을 지원한다. 멀티플렉싱 기능의 지원 여부에 따른 효율성을 서버의 성능 분석을 통해 검증하고 제안된 MPEG-4 스트리밍 시스템이 올바로 작동함을 보인다.

• 한국정보과학회논문지:컴퓨팅의 실제 및 레터
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• 제5권5호
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• pp.619-627
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• 1999

서로 다른 네트워크 환경을 갖는 도메인 사이에서 멀티미디어 서비스를 제공하기 위해서는 여러 가지 측면에 대한 기능 대응이 이루어져야 한다. 본 논문에서는 ATM 기반의 DAVIC(Digital Audio Visual Council) 서비스와 인터넷 AV 서비스를 연동하기 위하여, 프로토콜 변환 및 QoS(Quality of Service) 적응을 지원하는 연동 게이트웨이를 구현하였다. 구현된 연동 게이트웨이는 ATM 망에서 DSM-CC (Digital Storage Media Command & Control)를 기반으로 서비스되는 스트림을 RTSP(Real-Time STreaming Protocol)로 제어할 수 있으며, RTP(Real-Time Transport Protocol)를 이용하여 스트림을 전달한다. 또한 네트워크 대역폭 변화를 고려한 스트림 전달을 위해 QoS 모니터링과 MPEG 필터링을 이용한 QoS 적응 기능을 제공한다.

• Journal of Communications and Networks
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• 제15권6호
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• pp.635-644
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• 2013

Adaptive hypertext transfer protocol (HTTP) streaming has become a new trend to support adaptivity in video delivery. An HTTP streaming client needs to estimate exactly resource availability and resource demand. In this paper, we focus on the most important resource which is bandwidth. A new and general formulation for throughput estimation is presented taking into account previous values of instant throughput and round trip time. Besides, we introduce for the first time the use of bitrate estimation in HTTP streaming. The experiments show that our approach can effectively cope with drastic changes in connection throughput and video bitrate.

• 정보보호학회논문지
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• 제19권2호
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• pp.155-163
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• 2009

멀티미디어 기기와 인터넷의 발달로 디지털 콘텐츠의 제작과 유통이 일반화되면서 스트리밍 방식의 DRM 시스템이 중요한 위치를 차지하게 되었다. 기존의 스트리밍 방식의 DRM 시스템은 콘텐츠의 모든 데이터를 암호화함으로써 시스템에 많은 부하를 가져왔다. 본 논문에서는 데이터의 일부만을 암호화하여 시스템 성능을 극대화하고, 암호화된 콘텐츠를 네트워크 프로토콜에 독립적으로 전송할 수 있는 스트리밍 방식의 DRM 시스템을 제안하고, 제안하는 시스템을 구현하여 성능 분석한 결과를 제시한다.