• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• Journal of the Korea Society of Computer and Information
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• v.26 no.3
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• pp.1-7
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• 2021

In this paper, optimized media data retrieval and transmission based on the Hybrid CDN/P2P architecture and selective storage through user's prediction of requestability enable seamless data transfer to users and reduction of unnecessary traffic. We also propose a new media management method to minimize the possibility of transmission delay and packet loss so that media can be utilized in real time. To this end, we construct each media into logical segments, continuously compute weights for each segment, and determine whether to store segment data based on the calculated weights. We also designate scattered computing nodes on the network as local groups by distance and ensure that storage space is efficiently shared and utilized within those groups. Experiments conducted to verify the efficiency of the proposed technique have shown that the proposed method yields a relatively good performance evaluation compared to the existing methods, which can enable both initial latency reduction and seamless transmission.

• Journal of The Korean Association of Information Education
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• v.3 no.1
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• pp.65-74
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• 1999

People in the world can send and get useful information from anywhere via internet. Using the internet for educational purposes has been studied for several years. In this paper, we have been designed and implemented video database systems for English education. In the system we have studied ways to build and retrieve useful information from video database systems. By using our system we can easily find required video segment and can play it in real-time way by using streaming techniques.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.181-190
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• 2002

This paper proposes a motion-video multimedia system needed for computer applications like remote lecturing, distance learning, product demonstrations, and so on. The applications need a multimedia system which can author and play a motion-Video that is composed with computer screen images and audio continuously varing as time flows, not with real motion videos. Since the computer screen images are not like the real world video images in several rejects, MPEG is not competent as a compression algorithm for computer screen images raring continuously In this paper a new compression algorithm has been proposed, and a multimedia system that authors and plays a motion-video file which contains computer screen images and audio has been implemented. Also this paper shows the result of performance evaluation of both the compression algorithm and the multimedia system implemented in the paper.

• Journal of Multimedia Information System
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• v.4 no.4
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• pp.179-188
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• 2017

The Internet Video Coding (IVC) standard is due to be published by Moving Picture Experts Group (MPEG) for various Internet applications such as internet broadcast streaming. IVC aims at three things fundamentally: 1) forming IVC patents under a free of charge license, 2) reaching comparable compression performance to AVC/H.264 constrained Baseline Profile (cBP), and 3) maintaining computational complexity for feasible implementation of real-time encoding and decoding. MPEG experts have worked diligently on the intellectual property rights issues for IVC, and they reported that IVC already achieved the second goal (compression performance) and even showed comparable performance to even AVC/H.264 High Profile (HP). For the complexity issue, however, there has not been thorough analysis on IVC decoder. In this paper, we analyze the IVC decoder in view of the time complexity by evaluating running time. Through the experimental results, IVC is 3.6 times and 3.1 times more complex than AVC/H.264 cBP under constrained set (CS) 1 and CS2, respectively. Compared to AVC/H.264 HP, IVC is 2.8 times and 2.9 times slower in decoding time under CS1 and CS2, respectively. The most critical tool to be improved for lightweight IVC decoder is motion compensation process containing a resolution-adaptive interpolation filtering process.

• Journal of Digital Contents Society
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• v.3 no.1
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• pp.47-57
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• 2002

Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.75-88
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• 2009

This paper presents a video quality assessment method based on quality degradation factors of real-time multimedia streaming services. The video quality degradation is caused by video source compression and network states. In this paper, we propose a blocky metric on an image domain to measure quality degradation by video compression. In this paper, the proposed boundary strength index for the blocky metric is defined by ratio of the variation of two pixel values adjacent to $8{\times}8$ block boundary and the average variation at several pixels adjacent to the two boundary pixels. On the other hand, unnatural image movement caused by network performance deterioration such as jitter and delay factors can be observed. In this paper, a temporal-Jerkiness measurement method is proposed by computing statistics of luminance differences between consecutive frames and play-time intervals between frames. The proposed final Perceptual Video Quality Metric (PVQM) is proposed by consolidating both blocking strength and temporal-jerkiness. To evaluate performance of the proposed algorithm, the accuracy of the proposed algorithm is compared with Difference of Mean Opinion Score (DMOS) based on human visual system.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.738-749
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• 2011

In this paper, we study QoS(Quality of Service) control protocols and the effect using MAC/PHY parameters of client device in mobile network. We proposes the way of controling the bit-rate by estimating the channel condition of the client with measured MAC/PHY parameters which is sent from the client. With the proposed method, more accurate available bit-rate can be estimated compared to conventional protocol, RTCP(Real-time Transport Control Protocol). The accurate bit-rate estimation can decrease wasted bit-rate and transport delay. In the result of the advantages, the transported video quality can be enhanced. In this paper, we show the effects of enhancement using client's the field data measured in WiMAX.

• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.99-101
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• 2011

본 논문에서는 WiMAX 의 신호세기(CINR: Carrier to Interference + Noise Ratio)의 변화와 예측 방법에 대해 연구한다. WiMAX 와 같은 무선 네트워크에서는 신호세기가 불규칙하게 변화하기 때문에 비디오 스트리밍시 에러나 지연이 발생할 수 있다. CINR 은 클라이언트 측의 가용 대역폭을 결정하는 중요한 요소이다. 하지만 기존의 RTCP(Real-time Transport Control Protocol)를 이용한 비트율 제어는 RTT(Round Trip time), PLR(Packet Loss Rate) 등의 정보를 사용하기 때문에 부정확하거나 지연이 발생될 수 있다. 이를 보완하기 위한 CINR 을 직접 비트율 제어에 사용하는 방법에 대해 연구하기에 앞서, 본 논문에서는 CINR 을 분석하고 미래의 값을 예측하는 방법에 대해 연구한다. 본 논문의 분석을 통해 CINR 의 예측을 비교적 정확하게 수행할 수 있다면 앞으로의 가용 대역폭을 비교적 정확하게 예측할 수 있고 효율적인 비디오 스트리밍 시스템을 제안할 수 있다.