• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• Journal of Korea Multimedia Society
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• v.2 no.2
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• pp.200-216
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• 1999

In this paper, we propose a BTPN (Beta-distributed Timed Petri Net) model which is not only an effective multimedia synchronization and authoring specification model, but also a direct control model for playout of a title. Methods of specification of relationships among all media objects in a title by using the BTPN structure and language, as well as methods of analysis of the BTPN by means of a Remaining Timed Reachability Graph and Timing diagram, are proposed. A concept of critical object path, coming from PERT/CPM, is useful for modeling the uncertainty of playout of a multimedia title and editing of title.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.3
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• pp.48-54
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• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• KIPS Transactions on Computer and Communication Systems
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• v.11 no.10
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• pp.373-380
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• 2022

This paper presents a deep learning based group synchronization that supports networked immersive interactions between remote users. The goal of group synchronization is to enable all participants to synchronously interact with others for increasing user presence Most previous methods focus on NTP-based clock synchronization to enhance time accuracy. Moving average filters are used to control media playout time on the synchronization server. As an example, the exponentially weighted moving average(EWMA) would be able to track and estimate accurate playout time if the changes in input data are not significant. However it needs more time to be stable for any given change over time due to codec and system loads or fluctuations in network status. To tackle this problem, this work proposes the Deep Group Synchronization(DeepGroupSync), a group synchronization based on deep learning that models important features from the data. This model consists of two Gated Recurrent Unit(GRU) layers and one fully-connected layer, which predicts an optimal playout time by utilizing the sequential playout delays. The experiments are conducted with an existing method that uses the EWMA and the proposed method that uses the DeepGroupSync. The results show that the proposed method are more robust against unpredictable or rapid network condition changes than the existing method.

• 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.

• 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.

• 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.

• Journal of Communications and Networks
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• v.13 no.3
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• pp.277-285
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• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10B
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• pp.899-915
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• 2003

Delivering rich and continuous media contents robustly over a wide range of network conditions of the wired/wireless Internet is a highly challenging task. To address this challenges, the continuous media applications at the edge of network has become more and more adaptive while the best-effort Internet is slowly progressing towards improved networking services. That is, the role of network adaptive media delivery, which dynamically links the quality demand of application contents to the underlying networking services, has become more crucial. In this paper, we will first review the required network adaptation functionalities seen from the application side: congestion control / rate control, error control, and synchronization / adaptive playout. Then, we start the coverage of networking support issues that helps the realization of network adaptive media streaming - from network support and protocol support toward consolidated support via middleware. Finally, we propose a dynamic network adaptation framework that efficiently leverages its awareness of both media application (including contents) and underlying networking support.