• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2012.05a
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• pp.488-490
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• 2012

Nowadays QoS / QoE using technology as part of HTTP, dynamic adaptive streaming technology that became a big issue. In this paper, using HTTP, dynamic adaptive streaming technology is introduced. This dynamic adaptive streaming using HTTP-based system design and implementation. In this system, a number of MPEG2-TS file bitrate's conversion of image quality, a certain amount of time by the division of Segment, standing before the final note of the files that are used in transmission MPD (Media Presentation Description) File creation and streaming between servers and clients using HTTP to a dynamic and adaptive network environment consists of bitrate's streamers. In a network environment in which an unspecified number of seamless continuous playback of video streaming for a variety of sectors and will be utilized in the field of multimedia.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.12
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• pp.5669-5684
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• 2018

The new mobile edge network architecture has been required for an increasing amount of traffic, quality requirements, advanced driver assistance system for autonomous driving and new cloud computing demands on highway. This article proposes a hierarchical cloud computing architecture to enhance performance by using adaptive data load distribution for buses that play the role of edge computing server. A vehicular dynamic cloud is based on wireless architecture including Wireless Local Area Network and Long Term Evolution Advanced communication is used for data transmission between moving buses and cars. The main advantages of the proposed architecture include both a reduction of data loading for top layer cloud server and effective data distribution on traffic jam highway where moving vehicles require video on demand (VOD) services from server. Through the description of real environment based on NS-2 network simulation, we conducted experiments to validate the proposed new architecture. Moreover, we show the feasibility and effectiveness for the connected car media service on highway.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.8 no.4
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• pp.201-206
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• 2009

The third generation mobile communications system requiring the reliable multimedia data transmission has provided with the reliable voice, data and video services over the variable propagation environment. However the broadband wireless multiple access technologies cause Inter Symbol Interference (ISI) or Multiple Access Interference (MAI) to degrade the performance of W-CDMA (Code Division Multiple Access) system. Constant Modulus Algorithm (CMA) which is frequently used as the adaptive blind equalizers to remove the interfering signal has ill-convergence phenomenon without proper initialization. In this paper, new blind equalization method based on conventional CMA is proposed to improve the channel efficiency, and through computer simulation this is tested over the time varying fading environment of mobile communication system. Consequently, new blind equalization method into concatenated Kalman filter with CMA is verified better than conventional CMA through adopting minimum mean square errors and eye-pattern obtained from algorithm are compared.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.12
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• pp.1329-1336
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• 2015

Due to the advances in hardware technologies for low-power CMOS cameras, there have been various researches on wireless video sensor network(WVSN) applications including agricultural monitoring and environmental tracking. In such a system, its core technologies include video compression and wireless transmission. Since data of video sensors are bigger than those of other sensors, it is particularly necessary to estimate precisely the traffic after video encoding. In this paper, we present an estimation method for the encoded video traffic in WVSN networks. To estimate traffic characteristics accurately, the proposed method first measures complexities of frames and then applies them to the bit rate estimation adaptively. It is shown by experimental results that the proposed method improves the estimation of bit rate characteristics by more than 12% as compared to the existing method.

• Journal of Broadcast Engineering
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• v.20 no.1
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• pp.40-47
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• 2015

This paper discusses QoS enhancement in the Best-effort services of the service plan provided by MPEG Media Transport (MMT) systems for video streaming applications. Among MMT services, i.e. per-flow, per-class, and best-effort services, the server does not provide guaranteed bandwidth for the best-effort service only. Therefore, in the best-effort services, a bandwidth access priority is defined for various services, where the lowest priority is assigned to the low-level video services. To alleviate the issue of bandwidth limitation in the best-effort services, this paper investigates transmission of low-resolution video with low bitrate and up-sampling. Our experimental results prove the superiority of the proposed method in terms of delivered video quality.

• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.75-80
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• 2013

In this paper, the Ka-band maritime satellite communication systems for mobile terminals are proposed. The design includes the link budget analysis, determination of modulation and coding schemes and the overall structure of a transmitter. To avoid the harmful effects on the existing DVB-S2 services, the proposed maritime satellite system using the same spectrum with DVB-S2 at the same time employs the very wideband spreading transmission. Additionally, omni-directional low-gain antennas should be equipped in a mobile terminal to reduce the system cost. These two considerations limit the maximum transmission rate of the proposed system. Due to the limitations, the proposed system includes 36 dB or 39 dB spreading gain depending on the modulation scheme and a link-adaptive repetition method depending on the level of rain attenuation. To support short packets with minimal performance loss, the turbo code used in 3GPP instead of LDPC(low density parity check code) is adopted. By combining them, the overall structure of low-rate maritime satellite communication system is designed.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.1012-1021
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• 2005

This paper describes the design and implementation of an efficient multi-point multimedia conference system using IP grouping. Existing multi-point multimedia conference systems are difficult for multi-user to perform efficient cooperation due to bandwidth limitation for data transmission of video, audio and documentation. In the case that multi-user uses limited bandwidth, smooth cooperation does not accomplish due to transmission delay for the real-time transmission of image and speech data. A hybrid transfer method which is mixed with distributed and centralized methods is used for smooth cooperation, and the network bandwidth is reduced by forming multi-user conference systems of IP grouping in this paper. Also, adaptive image frame variations are used to solve bottleneck effect according to the number of users. An efficient multi-user conference system is designed to support audio quality.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.

• 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 KIISE:Computing Practices and Letters
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• v.6 no.5
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• pp.505-512
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• 2000

In this paper, an adaptive multimedia synchronization scheme for VOD (Video On Demand) services in internet environments is proposed. This scheme considers the characteristics of MPEG (Moving Picture Expert Group) system stream. Consequently, the intra-synchronization is handled at the pack layer, and the inter-synchronization is handled at the packet layer. The proposed scheme can cope adaptively with variation of packet loss, jitter and client's playback capacity. If there are variations of the packet loss or client's playback capacity, the server will change the transmission rate by selective picture skip. The client can then adjust and control the playback time according to the variation of the network jitter. Our experimental results show that the proposed scheme can quickly adapt to the network condition, and can guarantee a better quality of service than the other existing schemes.