• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.4
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• pp.13-21
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• 2022

Recently, in wireless video sensor networks(WVSN), various schemes for efficient video data transmission have been studied. In this paper, a layer based cooperative relaying(LCR) algorithm is proposed for minimizing scalable video transmission distortion from packet loss in WVSN. The proposed LCR algorithm consists of two modules. In the first step, a parameter based error propagation metric is proposed to predict the effect of each scalable layer on video quality degradation at low complexity. In the second step, a layer-based cooperative relay algorithm is proposed to minimize distortion due to packet loss using the proposed error propagation metric and channel information of the video sensor node and relay node. In the experiment, the proposed algorithm showed that the improvement of peak signal-to-noise ratio (PSNR) in various channel environments, compared to the previous algorithm(Energy based Cooperative Relaying, ECR) without considering the metric of error propagation.The proposed LCR algorithm minimizes video quality degradation from packet loss using both the channel information of relaying node and the amount of layer based error propagation in scalable video.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.1
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• pp.42-52
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• 2017

DASH (Dynamic Adaptive Streaming over HTTP) is an adaptive streaming technique that enables transmission of multimedia content when clients request the multimedia contents to server. In this system, to ensure the best quality of the content to satisfy users, it is necessary to precisely measure the residual bandwidth. However, the measured residual bandwidth by the DASH, which is not considering the transmission features of TCP, varies by the size of previous media segment, which makes it hard to ensure QoE to users. In this paper, we excluded the TCP Slow start range from measurement of residual bandwidth and suggested the new DASH bandwidth measuring method to decrease the error. Then, we realized the method in DASH system based on open source, and compared the existing measuring method. The new method showed that the accuracy of result has increased by 20%. Also, it could improve the QoE of users in terms of service quality and number of changes of segment quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.595-604
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• 2009

The HCCA reserves the channel resources based on the mean data rate in IEEE 802.11e. It may cause either the waste of channel resource or the increase of transmission delay at MAC layer if the frame size is rapidly varied when a compressed mode video codec such as MPEG video is used. To solve these problems, it is developed that the packet scheduler allocates the wireless resource adaptation by according to the packet size. However, it is difficult to perform the admission control because of the difficulty with calculating the available resources. In this paper, we propose a CAC mechanism to solve the problem that may not satisfy the QoS by increasing traffic load in case of using EDCA. Especially, the proposed CAC mechanism calculates the EB of TSs using the traffic information transmitted by the application layer and the number of average transmission according to the wireless channel environment, and then determines the admission of the TS based on the EB. According to the simulation results of the proposed CAC mechanism, it admitted the TSs under the loads which are satisfied within the delay bound. Therefore, the proposed mechanism guarantees QoS of streaming services effectively.

• Journal of Broadcast Engineering
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• v.27 no.2
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• pp.216-227
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• 2022

The MPEG-I (Immersive) group is working on a standardization project for immersive video that provides 6 degrees of freedom (6DoF). The MPEG Immersion Video (MIV) standard technology is intended to provide limited 6DoF based on depth map-based image rendering (DIBR) technique. Many efficient coding methods have been suggested for MIV, but efficient transmission strategies have received little attention in MPEG-I. This paper proposes group-based adaptive rendering method for immersive video streaming. Each group can be transmitted independently using group-based encoding, enabling adaptive transmission depending on the user's viewport. In the rendering process, the proposed method derives weights of group for view synthesis and allocate high quality bitstream according to a given viewport. The proposed method is implemented through the Test Model for Immersive Video (TMIV) test model. The proposed method demonstrates 17.0% Bjontegaard-delta rate (BD-rate) savings on the peak signalto-noise ratio (PSNR) and 14.6% on the Immersive Video PSNR(IV-PSNR) in terms of various end-to-end evaluation metrics in the experiment.

• Journal of KIISE:Software and Applications
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• v.35 no.4
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• pp.207-218
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• 2008

With the growth of the Internet, multimedia streaming becomes an important means to deliver video contents over the Internet and the amount of the streaming multimedia contents is also getting increased. However, it becomes difficult to guarantee the quality of service in real-time over the IP network environment with instantaneously varying bandwidth. In this paper, we propose an optimal adaptation framework for streaming contents over the Internet in the sense that the perceptual quality of the multi-angie content with multiple visual objects is maximized given the constraints such as available bandwidth and transcoding cost. In the multi-angle video service framework, the user can select his/her preferred alternate views among the given multiple video streams captured at different view angles for a same event. This enhanced experience often entails streaming problems in real-time over the network, such as instantaneous bandwidth changes in the Internet. In order to cope with this problem, we assume that multi-angle video contents are encoded at different bitrates and the appropriate video streams are then selected or transcoded for delivery to meet such bandwidth constraints. For the user selective consumption of the various bitstreams in the multi-angle video service, the bitstream in each angle can be encoded in various bitrate, and the user can select a sub-bitrstream in the given bitrstreams or transcode the corresponding content in order to deliver the optimally adapted video contents to the instantaneously changing network condition. Therefore, we define the transcoding cost which means the time taken for transcoding the video stream and formulate a unified optimization framework which maximizes the perceptual quality of the multiple video objects in the given constraints such as the transcoding cost and the network bandwidth. Finally, we present plenty of the experimental results to show the effectiveness of the proposed method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7B
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• pp.608-618
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• 2006

The increasing popularity of multimedia streaming services introduces new challenges in content distribution. Especially, it is important to provide the QoS guarantees as they are increasingly expected to support the multimedia applications. The service providers can improve the performance of multimedia streaming by caching the initial segment (prefix) of the popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while requesting the remainder of the stream from the server. In this paper, in order to apply the prefix caching service based on IETF's RTSP environment to the wireless networks, we propose the effective RTSP handling scheme that can adapt to the radio situation in wireless network and reduce the cutting phenomenon. Also, we propose the traffic based caching algorithm (TSLRU) to improve the performance of caching proxy. TSLRU classifies the traffic into three types, and improve the performance of caching proxy by reflecting the several elements such as traffic types, recency, frequency, object size when performing the replacement decision. In simulation, TSLRU and RTSP handling scheme performs better than the existing schemes in terms of byte hit rate, hit rate, startup latency, and throughput.

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.2
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• pp.29-38
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• 2021

The Internet including mobile networks is developing to overcoming the limitation of physical distance and providing or acquiring information from remote locations. However, the systems that use video as primary information require higher bandwidth for recognizing the situation in remote places more accurately through high-quality video as well as lower latency for faster interaction between devices and users. The emergence of the 5th generation mobile network provides features such as high bandwidth and precise location recognition that were not experienced in previous-generation technologies. In addition, the Mobile Edge Computing that minimizes network latency in the mobile network requires a change in the traditional system architecture that was composed of the existing smart device and high availability server system. However, even with 5G and MEC, since there is a limit to overcome the mobile network state fluctuations only by enhancing the network infrastructure, this study proposes a high-definition video streaming system in ultra-low latency based on the SRT protocol that provides Forward Error Correction and Fast Retransmission. The proposed system shows how to deploy software components that are developed in consideration of the nature of 5G and MEC to achieve sub-1 second latency for 4K real-time video streaming. In the last of this paper, we analyze the most significant factor in the entire video transmission process to achieve the lowest possible latency.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.16 no.3
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• pp.2141-2145
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• 2015

A new broadcasting signal transmission, which can save its channel bandwidth using compressive sensing(CS), is proposed in this paper. A new compression technique, which uses two dimensional discrete wavelet transform technique, is proposed to get high sparsity of multimedia image. A L1 minimization technique based on orthogonal matching pursuit is also introduced in order to reconstruct the compressed multimedia image. The CS enables us to save the channel bandwidth of wired and wireless broadcasting signal because various transmitted data are compressed using it. A $256{\times}256$ gray-scale image with compression rato of 20 %, which is sampled by 10 Gs/s, was transmitted to an optical receiver through 20-km optical transmission and then was reconstructed successfully using L1 minimization (bit error rate of $10^{-12}$ at the received optical power of -12.2 dB).

• Journal of Satellite, Information and Communications
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• v.1 no.1
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• pp.12-19
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• 2006

In this paper, we propose the bandwidth borrowing scheme which improves the performance of the cal admission control of the integrated GEO satellite networks. In general, target transmission rates of communications and streaming services are fixed, but data services do not have the target transmission rates. Therefore, we can control the transmission rates for data services flexibly according to the system loading situation. When the available bandwidth of the system is insufficient, the bandwidth borrowing scheme gives the bandwidth to request real time services by the transmission rates control of data services through packet scheduler. We make the queueing model for our system model and demonstrate the results through simulations. The simulation results show that there is a 8.7-35.2 dB gain at the total blocking probability according to the use of bandwidth borrowing scheme.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.401-410
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• 2007

Multimedia streams transmitted through the Internet have a strict playback delay time. Multimedia data arriving aster the playback time cannot be played in the receiver and are discarded. Thus, this paper proposed a protocol, in which the multimedia stream server determines whether data can be played in the receiver before sending the data. The proposed model has a PSCTP sub-layer on top of existing PR-SCTP and decides whether to send data messages, which have come from the multimedia applicationserver, and which PR-SCTP stream the data will be sent to. In addition, the proposed model uses the differentiated retransmission function of PR-SCTP. We evaluated the performance of SCTP, PR-SCTP and PSCTP using NS2 simulator. According to the results of the evaluation, the PSCTP protocol decreased the volume of transmission and increased the video decodable ratio compared to other protocols.