• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2005.11a
• /
• pp.155-158
• /
• 2005

본 논문에서는 대화형 기능을 가지는 MPEG-4기반의 멀티미디어를 IP 네트워크를 통해 전송할 수 있는 대화형 멀티미디어 스트리밍 시스템을 제안한다. 제안한 대화형 멀티미디어 시스템은 MPEG-4 콘텐츠를 분석, 전송할 수 있는 스트리밍 서버와 전송되는 MPEG-4 콘텐츠를 수신, 재생할 수 있는 클라이언트, 그리고 효율적인 멀티미디어의 전송을 위해 도입한 오버레이 네트워크로 구성된다. 오버레이 네트워크는 실제 IP 네트워크 상에 콘텐츠를 중계하는 릴레이 노드와 릴레이 노드들을 관리하는 네트워크 분배 관리자로서 구성된다.

• Journal of Broadcast Engineering
• /
• v.25 no.4
• /
• pp.560-568
• /
• 2020

In recent years, as network technology develops, the usage of streaming services by users is increasing. However, the complexity of streaming services is also increasing due to various terminal environments. Even when streaming the same content, it is necessary to re-encode the content according to the type of service. In order to solve the complexity and latency of the streaming service, Moving Picture Experts Group (MPEG) has standardized the Common Media Application Format (CMAF). In addition, as content transmission using a communication network becomes possible, the Republic of Korea's Ultra High Definition (UHD) broadcasting standard has been enacted as a hybrid standard using a broadcasting network and a communication network. The hybrid service enables various services such as transmitting additional information of contents or providing user-customized contents through a communication network. The Republic of Korea's UHD transmission standard utilizes MPEG Media Transport (MMT), and Mobile MMT is an extension of MMT to provide mobile network-specific functions. This paper proposes a method of inserting CMAF contents suitable for various streaming services using signaling messages of MMT and Mobile MMT. In addition, this paper proposes a model for content insertion system in heterogeneous network environment using broadcasting and communication networks, and verifies the validity of the proposed technology by checking the result of content insertion.

• Journal of Broadcast Engineering
• /
• v.24 no.4
• /
• pp.602-612
• /
• 2019

Various streaming technologies are being studied to guarantee the QoE of viewers due to the development of realistic media. HTTP adaptive streaming is a typical example, and it is based on HTTP / 1.1 and TCP. These protocols have become one of the causes of delaying the image delay and increasing the waiting time of web pages. Therefore, in this paper, we propose a QUIC-DASH system applying the UDP-based transmission protocols QUIC and HTTP / 2 to the MPEG-DASH system after analyzing various transmission protocols and development process of HTTP. Through experiments, the QUIC-DASH system confirmed the possibility of providing optimal performance in terms of transmission speed of LTE environment than existing system. We also suggest various future studies for better performance.

• The KIPS Transactions:PartC
• /
• v.16C no.4
• /
• pp.487-494
• /
• 2009

VOD(Video On Demand) that provides streaming service according to the user's requirement in real time, consists of the video streaming server and the client system. Since it is very hard to apply the traditional server-client model that a server communicates with many clients through 1:1 connection to VOD system because it requires very high network bandwidth, many researches have been done to address this problem. Batching technique is one of VOD system based on Multicast that requires very small network bandwidth. However, the batching based VOD system has a limitation that it is very hard to provide VCR(Video Cassette Recorder) ability. In this paper, we propose a technique that reduces the required network bandwidth to provide VCR function by using the characteristic of MPEG, one of international video compression standard. In the proposed technique, a new video stream for VCR function is constructed with I pictures that is able to be decoded independently. The new video stream for VCR function is transmitted with the video stream for normal play together in Batching manner. The performance evaluation result shows that the proposed technique not only reduces the required network bandwidth and memory usage but also decreases the CPU usages.

• The Transactions of The Korean Institute of Electrical Engineers
• /
• v.56 no.5
• /
• pp.987-989
• /
• 2007

This paper describes a digital convergence platform(DCP) which is implemented based on the MPEG-21 multimedia framework. DCP is a newly proposed solution in this research for the convergence service of future home multimedia environment. DCP is a common platform designed to have the feature of reconfigurability, by means of S/W. which supports to run diverse digital multimedia services. A distributed peer to peer service and transaction model is also a new feature realized in DCP using the MPEG-21 multimedia framework. A prototype DCP is implemented to verify its functions of multimedia service and transactions. The developed DCPs are networked with IP clustering storage systems for the distributed service of multimedia. Successful streaming services of the MPEG-2/4 video and audio are verified with the implemented test-bed system of DCP.

• Journal of Internet Computing and Services
• /
• v.17 no.4
• /
• pp.19-26
• /
• 2016

In this paper, the dynamic buffering based smart OTT platform was proposed, and analyzed for adaptive bit-rate video delivery with the optimization of HLS (HTTP Live Streaming). This platform consists of the software platform between sever and client which detects the bandwidth capacity, and adjusts the quality of the streaming for multiple bit-rates resolutions. In order to apply adaptive buffering, two buffers are added to the basic HLS player, and each buffer is responsible for constantly buffering a previous and the next channels relative to the current channel. This adaptive transmitting with smart OTT platform is superior to delivering a static video file at a single buffering, because the video stream of adaptive double buffers can be switched streaming according to client's available network speed. As a result, this proposed smart OTT can be cooperated to the application of HLS server with segmented H.265 MPEG-2 TS video & m3u8 files with its information based on the optimized transmission channel state of live and VOD, and applied to PLC transmission, too.

• The KIPS Transactions:PartC
• /
• v.18C no.6
• /
• pp.433-440
• /
• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.347-350
• /
• 2000

인터넷을 이용한 디지털 오디오 방송 서비스에 대한 관심이 집중되면서 디지털 오디오 데이터를 서버로부터 사용자에게까지 실시간으로 전송하기 위한 연구가 진행되고 있다. 디지털 오디오 방송의 실시간 전송을 위하여 효율적인 오디오 압축 기술의 개발도 중요하지만 이들 오디오 압축 기술과 연계되는 오디오 스트리밍 기술이 매우 중요하다. 본 논문에서는 현재 사용되고 인터넷 오디오 방송관련 기술을 분석하고, 특히, IETF에서 논의되고 있는 MPEG-2AAC 및 MPEG-4 오디오를 인터넷을 통하여 전송하기 위한 RTP payload 포맷을 분석하고, 기술개발을 위한 고려사항을 제안한다.

• Journal of Broadcast Engineering
• /
• v.5 no.2
• /
• pp.227-238
• /
• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.

• Proceedings of the IEEK Conference
• /
• 2005.11a
• /
• pp.1193-1196
• /
• 2005

In this paper, we presents a new multimedia service transmit method using IPv6. The IPv6 provides the address system of 128 bit and the address space which is infinite it provides. But it will not become the IPv4 and interchange not to be, it uses the DSTM Transition mechanism which will reach and the IPv4 center in the packet header the service type it will be able to support the service class of multi type (TOS) it secures the weak point of data transfer delay it puts a base in the IPv6. The efficiency of this proposed technique have been proven by MPEG-4 streaming video streaming of the IPv6 namely, 6Xtream embodied a order form/live streaming server and the client which it uses to be possible in base and real-time decoding method.