• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.953-960
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• 2002

Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3B
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• pp.144-151
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• 2008

In this paper, we propose a video streaming server that guarantees a certain level of quality when a server should serve video streaming service to multiple heterogenous clients simultaneously with TCP transport. If each heterogeneous client requests video streaming service in according to its own requirement such as bitrate of content and these requests are accepted by a server, then TCP flows for each video streaming session fairly share limited uplink bandwidth of the server. At this time, because TCP's bandwidth fair-share characteristics can result in bandwidth shrinkage of higher bitrate video streaming session, the client of higher bitrate video may suffer sluggish playback which is related with streaming QoS degradation. To tackle this problem, our proposed server system uses multiple TCP connections adaptively for each video streaming session depending on the anticipated status of the client playout buffer. Simulation results show that our proposed algorithm can successfully reduce the occurrence of playout buffer underrun and enhance streaming quality for whole video clients.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.11
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• pp.5512-5517
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• 2012

In this paper we suggest multimedia synchronization model that is based on the Petri-net and services desirable quality of service requirement. Proposed model applies variable buffer which can be allowed, and then it presents high quality of service and real time characteristics. This paper decreases the data loss resulted from variation of delay time and from loss time of media-data by means of applying delay jitter in order to deal with synchronization interval adjustment. Plus, the mechanism adaptively manages the waiting time of smoothing buffer, which leads to minimize the gap from the variation of delay time. The proposed paper is suitable to the system which requires the guarantee of high quality of service and mechanism improves quality of services such as decrease of loss rate, increase of playout rate.

• The KIPS Transactions:PartA
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• v.16A no.5
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• pp.327-338
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• 2009

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose a network adaptive prefetching technique, PAP, for guarantee of constant QoS. The PAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The PAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low and increases the playout interval to prevent buffer underflow. The experiment result shows that the proposed PAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• IE interfaces
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• v.22 no.1
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• pp.95-103
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• 2009

During the past few years, design of experiments (DOE) has been gaining acceptance in the telecommunications research community as a mean for designing and analyzing experiments economically and efficiently. In addition, the need for introducing a systematic robust design methodology (i.e., one of the most popular DOE methodologies) to network simulations has been increasing. In this paper, we present an architecture of voice over IP (VoIP) application and the E-Model for calculating the perceived quality of service (QoS). Then, we apply the Taguchi robust design methodology to optimize the perceived QoS of VoIP application, and describe the detailed step-by-step procedures. We have used ns-2 simulator to collect experimental data in which the SN ratio, a robustness measure, is analyzed to determine an optimal design condition. The analysis shows that "initial delay time in playout buffer" is a major control factor for ensuring robust behaviors of the perceived QoS of VoIP. Finally, we verify the proposed optimal design condition using a confirmation experiment.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.649-655
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• 2000

Real-time application programs have constraints which need to be met between media-data. These constraints represents the delay time ad quality of service between media-data to be presented. In order to efficiently describe the delay time and quality of service, a new synchronization mechanism is needed. Proposed paper is a dynamic synchronization that minimized the effects of adaptive transmission delay time. That is, the method meets the requirements of synchronization between media-dat by handling dynamically the adaptive waiting time resulted from variations of delay time. In addition, the mechanism has interval adjustment using maximum delay jitter time. This paper decreases the data loss resulted from variation of delay time and from loss time of media-data by means of applying delay jitter in order to deal with synchronization interval adjustment. Plus, the mechanism adaptively manages the waiting time of smoothing buffer, which leads to minimize the gap from the variation of delay time. The proposed paper is suitable to the system which requires the guarantee of high quality of service and mechanism improves quality of services such as decrease of loss rate, increase of playout rate.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.145-158
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• 1997

Method for synchronizing multimedia data is needed to support continuous transmission of multimedia data through a network in a bounded time and it also required for supporting continuous presentation of multimedia data with the required norminal playout rate in distributed network environments. This paper describes a new synchronization method for supporting delay-sensitive multimedia Presentation without degration of Quality of services of multimedia application. It mainly aims to support both intermedia and intermedia synchronization by absorbing network variations which may cause skew or jitter. In order to remove asynchonization problems, we make use of logical time system, dynamic buffer control method, and adjusting synchronization intervals based on the quality of services of a multimedia. It might be more suitable for working on distribute[1 multimedia systems where the network delay variation is changed from time to time and no global clock is supported. And it also can effectively reduce the amount of buffer requirements needed for transfering multimedia data between source and destination system by adjusting synchronization intervals with acceptable packet delay limits and packet loss rates.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.