• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.1
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• pp.15-21
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• 2010

This paper proposes the multimedia fingerprinting code insertion algorithm when video content is distributed in P2P environment, and designs the collusion codebook SRP(Small RISC Processor) embedded system for the collusion attack prevention. In the implemented system, it is detecting the fingerprinting code inserted in the video content of the client user in which it requests an upload to the web server and in which if it is certified content then transmitted to the streaming server then the implemented system allowed to distribute in P2P network. On the contrary, if it detects the collusion code, than the implemented system blocks to transmit the video content to the streaming server and discontinues to distribute in P2P network. And also it traces the colluders who generate the collusion code and participates in the collusion attack. The collusion code of the averaging attack is generated with 10% of BIBD code v. Based on the generated collusion code, the codebook is designed. As a result, when the insert quantity of the fingerprinting code is 0.15% upper in bitplane 0~3 of the Y(luminance) element of I-frame at the video compression of ASF for a streaming service and MP4 for an offline offer of video content, the correlation coefficient of the inserted original code and the detected code is above 0.15. At the correlation coefficient is above 0.1 then the detection ratio of the collusion code is 38%, and is above 0.2 then the trace ratio of the colluder is 20%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.380-389
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• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.4
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• pp.121-126
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• 2009

Video poxy server which is located near clients can store the frequently requested video data in storage space in order to minimize initial latency and network traffic significantly. However, due to the limited storage space in video proxy server, an appropriate deletion algorithm is needed to remove the old video data which is not serviced for a long time. Thus, we propose an efficient video data deletion algorithm for video proxy server. The proposed deletion algorithm removes the video which has the lowest request possibility based on the user access patterns. In our algorithm, we arrange the videos which are stored in video proxy server according to the requested time sequence and then, select the video which has the oldest requested time. The selected video is partially removed in order to free up storage space in video poky server. The simulation results show that the proposed algorithm performs better than other algorithms in terms of the block hit rate and the number of block deletion.

• Journal of Broadcast Engineering
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• v.27 no.4
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• pp.548-560
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• 2022

Video based Point Cloud Compression (V-PCC) is one of the compression methods for compressing point clouds, and shows high efficiency in dynamic point cloud compression with movement due to the feature of compressing point cloud data using an existing video codec. Accordingly, V-PCC is drawing attention as a core technology for immersive content services such as AR/VR. In order to effectively service these V-PCC contents through a media streaming platform, it is necessary to encapsulate them in the existing media file format, ISO based Media File Format (ISOBMFF). However, in order to service through an adaptive streaming platform such as Dynamic Adaptive Streaming over HTTP (DASH), it is necessary to encode V-PCC contents of various qualities and store them in the server. Due to the size of the 2D media, it causes a great burden on the encoder and the server compared to the existing 2D media. As a method to solve such a problem, it may be considered to configure a streaming platform based on content obtained through V-PCC content encoding based on SHVC. Therefore, this paper encapsulates the SHVC-based V-PCC bitstream into ISOBMFF suitable for DASH service and proposes a configuration method to service it. In addition, in this paper, we propose ISOBMFF encapsulation and DASH configuration method to effectively service SHVC-based V-PCC contents, and confirm them through verification experiments.

• IEMEK Journal of Embedded Systems and Applications
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• v.10 no.4
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• pp.241-255
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• 2015

Recent network environment has been rapidly evolved to cloud computing environment based on the development of the Internet technologies. Furthermore there is an increasing demand on mobile cloud computing due to explosive growth of smart devices and wide deployment of LTE-based cellular networks. Thus mobile Desktop-as-a-Service(DaaS) could be a pervasive service for nomadic users. In addition, video streaming traffic is currently more than two-thirds of mobile traffic and ever increasing. All such trends enable that video streaming in mobile DaaS could be an important concern for mobile cloud computing. It should be noted that the performance of the Transmission Control Protocol(TCP) on cloud host servers greatly affects Quality of Service(QoS) of video streams for mobile users. With widely deployed Linux server platforms for cloud computing, in this paper, we conduct an experimental analysis of the twelve Linux TCP variants in mobile DaaS environments. The results of our experiments show that the TCP Illinois outperforms the other TCP variants in terms of wide range of packet loss rate and propagation delay over LTE-based wireless links between cloud servers and mobile devices, even though TCP CUBIC is usually used in default in the current Linux systems.

• Journal of IKEEE
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• v.13 no.4
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• pp.29-36
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• 2009

The packet loss and the disconnection during handoff are the most critical problems which degrade the video quality in wireless video streaming. To cope with these problems, we propose an efficient video streaming method in this paper, which does not only dynamically adjust the video transmission rate based on the raptor forward error correction (FEC) level, but also minimize the error propagation during handoff. Firstly, the channel bandwidth of the wireless broadband internet, called WiBro, is estimated by analyzing channel parameters including the carrier to interference and noise ratio (CINR) and the handoff. Secondly, the streaming server adjusts the next transmission rate according to the estimated channel bandwidth and the raptor FEC level to avoid packet error. Also, the encoder performs the intra refresh method that inserts an intra frame (I-frame) right after handoff to reduce the error propagation effectively. Experimental results indicate that the proposed method can improve the performance of the video streaming over WiBro network.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.

• Journal of the Korea Society of Computer and Information
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• v.15 no.3
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• pp.81-89
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• 2010

The research on peer-to-peer streaming schemes has largely focused on tree-push and mesh-pull structures. However, the tree-push structure has a defect that the tree restructuring time is long, and the mesh-pull structure has long startup delay and lag time from source servers. In this paper, we propose a new peer-to-peer live streaming scheme based on a push-mesh structure that takes advantages of tree-push and mesh-pull structure simultaneously. This structure basically provides the mesh-pull mechanism for data transmission and utilizes peers with high network upload capacity. It also supports the push mechanism along with paths from a source server, super peers, and selected general peers. By NS-2 simulation experiments, we finally show that our proposed scheme can achieve shorter startup delay than the mesh-pull structure, similar lag time to tree-push structure and best playback continuity among the three schemes.

• Journal of the Korean Institute of Intelligent Systems
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• v.19 no.1
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• pp.40-47
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• 2009

Recently, dynamic proxy caching has been proposed on the distributed environment so that media objects by user's requests can be served directly from the proxy without contacting the server. However, it makes caching challenging due to multimedia large sizes, low latency and continuous streaming demands of media objects. To solve the problems caused by streaming demands of media objects, this paper has been proposed the grouping scheme with fuzzy filtering based on partition and mapping. For partition and mapping, this paper divides media block segments into fixed partition reference block(R$_f$P) and variable partition reference block(R$_v$P). For semantic relationship, it makes fuzzy relationship to performs according to the fixed partition temporal synchronization(T$_f$) and variable partition temporal synchronization(T$_v$). Simulation results show that the proposed scheme makes streaming service efficiently with a high average request response time rate and cache hit rate and with a low delayed startup ratio compared with other schemes.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.527-530
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• 2014

DASH is a well-known streaming technology, which was proposed in 2010 by MPEG and standardized in 2011. Major multimedia contents service providers, including Apple, Microsoft, and Adobe are all using this technology to support their media streaming services. Whenever a new service is requested to the server, the DASH technology helps servicing the multimedia streaming to client by recognizing the capacity of network and by adapting the quality of the multimedia contents. In DASH, the quality of multimedia contents will be automatically lowered to meet the fluctuating network status, when undesirable breaks interrupt the network. In this paper, we classified and analysed the advantages and disadvantages of DASH researches in three aspects: bit-rate measurement method, bandwidth aggregation method; rate adaptation metrics, algorithms and logics; user's experiences and QoE.