• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12B
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• pp.838-846
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• 2005

In this paper, we consider the following two issues for implementing the synchronous Ethernet systems. First, a synchronous Ethernet system employs a fixed size superframe which is divided into a synchronous period and an asynchronous one. We note that the starting point of a superframe is not deterministic when an ordinary data frame's transmission is overlapped the superframe boundary. This overlap may be a fatal drawback for strict jitter bounded applications. Circumventing the problem, we propose a frame fragmentation scheme to provide a zero jitter, and compare its delay performance with the hold scheme which also provides the zero jitter. We next concern that IEEE 802.3x pause frames cannot be promptly transmitted in a synchronous period, and thus asynchronous traffics may be dramatically get dropped at the input buffer of a switch. To handle the problem, we propose an efficient flow control by allowing the transmission of the pause frame in a synchronous period, and investigate the blocking probability of the asynchronous traffics by the simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.29-40
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• 2004

In this paper, we propose a new early congestion detection and notification technique called QR-AQM. Unlike RED and it's variation, QR-AQM measures the total traffic rate from TCP sessions, predicts future network congestion, and determine the packet marking probability based on the measured traffic rate. By incorporating the traffic rate in the decision process of the packet marking probability, QR-AQM is capable of foreseeing future network congestion as well as terminating congestion resolution procedure in much more timely fashion than RED. As a result, simulation results show that QR-AQM maintains the buffer level within a fairly narrow range around a target buffer level that may be selected arbitrarily as a control parameter. Consequently, compared to RED and its variations, QR-AQM is expected to significantly reduce the jitter and delay variance of packets traveling through the buffer while achieving nearly identical link utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1976-1985
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• 2015

The amount of packets in palyout buffer of video streaming receiver can be changed by network condition, and saturated and exhausted by the delay and jitter. Especially, if the amount of incoming video traffic exceeds the maximum allowed playout buffer, buffer overflow problem can be generated. It makes the deterioration of video image and the discontinuity of playout by skip phenomenon. Also, if the incoming packets are delayed by network confusion, the stop phenomenon of video image is made by buffering due to buffer underflow problem. To solve these problems, this paper proposes the video streaming receiver with token bucket scheme which automatically establishes the important parameters like token generation rate r and bucket maximum capacity c adapting to the pattern of video packets. The simulation results using network simulator-2 (NS-2) and joint scalable video model (JSVM) show that the proposed token bucket scheme with automatic establishment parameter provides better performance than the existing token bucket scheme with manual establishment parameter in terms of the generation number of overflow and underflow, packer loss rate, and peak signal to noise ratio (PSNR) in three test video sequences.

• Journal of the Korea Society of Computer and Information
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• v.24 no.11
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• pp.11-19
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• 2019

Recently, hybrid CDN/P2P video streaming architecture is specially designed and deployed to achieve the scalability of P2P networks and the desired low delay and high throughput of CDNs. In this paper, we propose a cooperative video streaming and active node buffer management technique in hybrid CDN/P2P architecture. The key idea of this streaming strategy is to minimize network latency such as jitter and packet loss and to maximize the QoS(quality of service) by effectively and efficiently utilizing the information sharing of file location in CDN's proxy server which is an end node located close to a user and P2P network. Through simulation, we show that the proposed cooperative video streaming and active node buffer management technique based on CDN and P2P network improves the performance of realtime video streaming compared to previous methods.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1998.06a
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• pp.246-251
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• 1998

The basic requirement of a distributed multimedia system are intramedia synchronization which asks the strict delay and jitter for the check period of media buffer and the scaling duration with periodic continuous media such as audio and video media, and intermedia synchronization that needs the constraint for relative time relations among them when several media are presented in parallel. In this paper, a distributed multimedia synchronization algorithm based on the fuzzy logic is presented and the performance is evaluated through simulation. Intramedia synchronisation algorithm uses the media scaling techniques and intermedia synchronization algorithm uses variable service rates on the basis of fuzzy logic to solve the multimedia synchronization problem.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.3
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• pp.271-281
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• 1999

본 논문에서는 멀티미디어 파일 시스템을 위한 새로운 버퍼관리기법을 제안한고 이를 실제 구현을 통하여 성능을 분석하였다. 제안한 관리 기법에서는 다양한 멀티미디어 파일의 참조 특성을 반영하여 참조되는 파일에 일정량의 버퍼를 할당해주고, 새로운 데이터에대한 대치는 그 파일에게 할당된 버퍼에서만 수행되도록 하였다. 또한 이런 버퍼 할당을 데이터 소비율에 맞게 동적으로 조정할 수 있는 방법과 새로운 파일 참조를 제어하는 채택 제어 방법도 설계/구현하였다. FreeBSD를 수정하여 구현한 시스템에서의 실험에 의하면 제안한 버퍼 관리 기법이 FrddBSD의 경우보다 다양한 데이터 소비율을 필요로 하는 참조인 경우에 적은 수의 버퍼를 이용하여 더 높은 적중률(hit ratio)을 제공할 수 있음을 알 수 있었다. 이런 향상된 적중률은 멀티미디어 상영시 니터(jitter)를 줄이는데 기여할 수 있다.

• Journal of Korea Multimedia Society
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• v.1 no.1
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• pp.18-25
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• 1998

The basic requirements of distributed multimedia systems are intramedia synchronization which asks the strict delay and jitter for the check period of media buffer and the scaling duration with periodic continuous media such as audio and video media, and intermedia synchronization that needs the constraint for relative time relations among them when several media are presented in parallel. In this paper, a distributed intermedia synchronization algorithm that solves the intermedia synchronization problem by using variable service rates based on the fuzzy logic is designed and then the performance is evaluated through simulation.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.164-167
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• 2003

In this paper, we aim to communicate wirelessly as appling the Bluetooth technology to the VoIP system, and we select the embedded system which can be guaranteed performance and economical efficiency for implementation that system. So we implemented embedded Bluetooth AP and embedded VoIP system based on Bluetooth. For voice quality improvement in the implemented system, the Bluetooth ACL link and the appropriate Bluetooth packet was selected. Also, it was designed about the handling method of voice packet by using variable jitter buffer and then tested on embedded VoIP system based on Bluetooth.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.48-55
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• 2010

It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.