• The KIPS Transactions:PartA
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• v.12A no.3 s.93
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• pp.197-204
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• 2005

In this paper, we consider the transmission or variable bit rate (VBR) stored video for the distributed video streaming service over Internet. In streaming service, users often suffer from the discontinuity in playback due to the decrease in bandwidth during transmission according to bandwidth renegotiation protocol. We propose a novel transmission technique to overcome this problem for stored variable bit rate video. This scheme uses a priori information of stored VBR video to continue streaming without playback discontinuity. In addition, an approximation scheme for the buffer-bandwidth relation is proposed in order to facilitate the admission control under the proposed scheme.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3804-3819
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• 2018

The increasing popularity of HTTP adaptive video streaming services has dramatically increased bandwidth requirements on operator networks, which attempt to shape their traffic through Deep Packet inspection (DPI). However, Google and certain content providers have started to encrypt their video services. As a result, operators often encounter difficulties in shaping their encrypted video traffic via DPI. This highlights the need for new traffic classification methods for encrypted HTTP adaptive video streaming to enable smart traffic shaping. These new methods will have to effectively estimate the quality representation layer and playout buffer. We present a new machine learning method and show for the first time that video quality representation classification for (YouTube) encrypted HTTP adaptive streaming is possible. The crawler codes and the datasets are provided in [43,44,51]. An extensive empirical evaluation shows that our method is able to independently classify every video segment into one of the quality representation layers with 97% accuracy if the browser is Safari with a Flash Player and 77% accuracy if the browser is Chrome, Explorer, Firefox or Safari with an HTML5 player.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.1335-1340
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• 2006

본 논문에서는 사용자의 상호작용에 대해서 끊김없이 재생하는 멀티미디어 스트리밍 기법을 제안한다. 제안하는 멀티미디어 스트리밍 기법은 기존 디코딩 버퍼와 스트림 버퍼 외에 스트림 백업 버퍼를 추가로 사용한다. 추가된 스트림 백업 버퍼는 사용자의 상호작용에 효율적으로 지원할 수 있도록 현재 재생 위치를 중심으로 기존 스트림 버퍼의 2배 크기만큼 미리 버퍼링하여 효율적인 탐색과 빠른 재생을 지원한다. 사용자가 요구하는 대부분의 탐색작용은 현재 재생 위치 근처에서 일어날 확률이 높으므로, 제안한 멀티미디어 스트리밍 기법을 이용하여 사용자 상호작용에서 거의 지연 없이 효율적으로 재생한다. 본 논문에서는 구현 환경으로 멀티미디어 스트리밍을 지원하기 위한 MS사의 MMS서버를 이용하여 멀티미디어 스트리밍을 송수신하고 기존 재생기들과 제시한 멀티미디어 스트리밍 기법을 적용한 재생기와의 성능을 비교 평가하였다. 제안한 스트리밍 기법의 적용 예로서는 원격 강의 및 네트워크 게임 등이 있다.

• Journal of KIISE
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• v.43 no.3
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• pp.327-335
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• 2016

In this paper, we propose a buffer management scheme suitable for interactive multimedia services. We consider a typical delay optimization environment so that receiver buffer lengths vary according to the round trip time estimation. In this environment, we propose an optimization technique for minimizing the loss of information that may occur when a reduced buffer length forces I/P/B frames in the buffer to drop. We modeled our problem as a Knapsack Problem for which we used dynamic programing in order to find an approximate solution. The proposed technique is compared with the existing buffer management techniques. Through simulation studies, we found that our approach could increase PSNR, which is important to video quality.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.569-572
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• 2005

This paper proposed the design and implementation of a DSP platform for the various multimedia streaming. The DSP platform synchronizes with host PC to configure DSP and to transmit multimedia streaming through PCI. The suggested DSP platform decodes high-capacity video/audio data using the suggested high-speed FIFO, CPLD and memory interface. The buffer control techniques is proposed in other to avoid the under/over-run of the audio/video data during the audio/video decoding. For the DSP platform test, host PC transmits program stream(PS) that consists of the MPEG-2 video MP@ML and 5.1ch AC3 audio data (Coyote.mov file, half hour running time) to DSP platform. The DSP platform plays continuously back the high sound-quality audio and high-definition video at once.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.3
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• pp.1416-1428
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• 2016

In this paper, we address the channel zapping time problem of video streaming services based on HTTP Live Streaming (HLS) on smart Over-The-Top Set-Top Boxes (OTT STBs). Experimental analysis of the channel zapping time, show that smart OTT STBs inevitably suffer from the accumulated zapping time through channel change request, Internet Group Management Protocol (IGMP) leave/join, synchronization delay, video buffer delay, and STB processing delay when providing HLS services. As a practical solution for the zapping time reduction, an adaptive buffering method is proposed. The proposed method exploits two adaptive buffers added to the basic HLS player. These two adaptive buffers are responsible for constantly buffering previous and next channels relative to the current channel. Implementation and test results show that a stable zapping time less than one second can be achieved even under diverse video bitrate changes and varying network conditions by the proposed adaptive buffering method.

• Journal of KIISE
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• v.44 no.4
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• pp.423-432
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• 2017

Recently, the video quality control schemes for the improvement of the QoE (Quality of Experience) of video streaming services that are based on DASH (Dynamic Adaptive Streaming over HTTP), which is a standard of HTTP adaptive streaming (HAS) services, have been studied. However, the problem of the existing schemes is the degradation that is due to unnecessary quality changes because the VBR (Variable Bitrate) characteristics of the video are not considered. In this paper, we propose a SC-DASH (Segment Characteristics-based DASH) which controls the video quality based on the segment characteristics. The SC-DASH can prevent the occurrence of the unnecessary quality changes by controlling the video quality based on the size of the next segment, the segment throughput, and the buffer occupancy. The experiment results showed that the SC-DASH improves the QoE by reducing the unnecessary quality changes compared with the existing quality control schemes.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.