• Journal of KIISE
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• v.44 no.12
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• pp.1296-1306
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• 2017

Recently, as the number of Internet of Things users has increased, IETF (Internet Engineering Task Force) has released the CoAP (Constrained Application Protocol). So Internet of Things have been researched actively. However, existing studies are difficult to adapt to streaming service due to low transmission rate that result from buffer underflow. In other words, one block is transmitted one block to client's one request according to the internet environment of limited resources. The proposed scheme adaptively adjusts the rate of CON(Confirmable) message among all messages for predicting the exact network condition. Based on this, the number of blocks is determined by using buffer occupancy rate and content download rate. Therefore it improves the quality of user experience by mitigating playback interruption. Experimental results show that the proposed scheme solves the buffer underflow problem in Internet of Things streaming environment by controlling transmission rate according to the network condition.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.187-190
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• 1997

Test Model15, which is used widely for the MPEG-2 bit rate control, has several problems such as non-unform picture quality, scene change and buffer underflow. Therefore, various algorithms have been developed to solve these problems. In this paper, we study various algorithms for the MPEG-2 bit rate control and compare their performances using software simulations. We also propose a new bit rate control strategy based on coded types of macroblocks within a picture.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.591-596
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• 1999

In this paper, we present a joint quality control system to be able to accurately control the relative picture quality among the video programs in terms of PSNR. The joint quality control system allows variable bit rate (VBR) for each video program to maintain the pre-determined relative picture quality among the aggregated video programs while keeping a constant sum of the bit rates for all programs to be transmitted over a single constant bit rate (CBR) channel. This is achieved by simultaneous controlling the video encoders to generate VBR video streams at the central controller. Furthermore we also suggest buffer regulation method based on the analysis of the constraints imposed by sender/receiver buffer sizes and total transmission rate. Through various simulation results, it is found that our quality control systems guarantee that the video buffers do not overflow and underflow and the quality control errors do not exceed 0.1 ㏈.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.10a
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• pp.1593-1596
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• 2015

실시간 영상회의 시스템은 네트워크 및 버퍼링의 지연으로 사용자 정보의 전달이 시스템 간에 효율적으로 이루어지지 않고 있어 실시간성이 완벽하게 보장되지 않고 있는 것이 현실이다. 하지만, 네트워크 인프라의 보강과 지터 지연에 대한 연구는 활발하지만, 버퍼링에 대한 연구는 미흡한 상태이다. 본 논문에서는 버퍼링 지연에 따른 문제 해결을 위한 FRCB(Frame-Rate Control Buffer) 관리 기법을 제안하고자 한다. FRCB는 버퍼의 Overflow와 Underflow를 방지하기 위한 FTH(Fist-play THreshold)와 STH(Slow-play THreshold)로 구성되며, CPU 부하가 높은 상황에서도 좋은 성능을 보여 고품질의 실시간 영상회의에 적합함을 보였다.

• Journal of KIISE:Information Networking
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• v.34 no.4
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• pp.276-286
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• 2007

Existing streaming mechanisms have no consideration for the characteristics of streaming applications because they only consider network stability. In this paper, in order to overcome limitations of the previous work on video streaming, we propose a new video streaming mechanism called "HAViS(Hybrid Approach for Video Streaming)". The proposed mechanism includes more sophisticated features that consider both network and user requirements. Therefore, the HAViS mechanism improves the network stability by adjusting the sending rate of video stream based on the network state and it also provides the smooth playback by preventing the buffer underflow or overflow. Moreover, it is designed to take into consideration the streaming video content. Through the simulation, we prove that the HAViS mechanism efficiently uses the buffer resources and provides improved network stability and smooth playback.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.

• Journal of KIISE
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• v.45 no.2
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• pp.175-186
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• 2018

Recently, HAS (HTTP Adaptive Streaming) has been the subject of much attention to improve the QoE (Quality of Experience). In a high latency network, HAS degrades the QoE due to the lost RTT cycle since it replies with a response of one segment to the request of one segment. The server-push based HAS schemes of downloading multiple segments in one request cause QoE degradation due to the buffer underflow. In this paper, we propose a VSSDS (Video Streaming Scheme based on Dynamic Server-push) scheme to improve the QoE in a high latency network. The proposed scheme adjust video quality by estimating available bandwidth and determine the number of segments to be downloaded for each segment request cycle. Through the simulation, the proposed scheme not only improves the average video bitrate but also alleviates the buffer underflow.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.181-188
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• 2016

Video streaming traffics which are arrived in receiver have irregular traffic patterns by many problems over the network path. Particularly, if these received traffics enter into replay buffer without any operation, the overflow and underflow effects are made according to the buffer status. There was an existing scheme which automatically set up token bucket parameters using the video information file under the assumption of the lossless packet on network. The existing scheme has a problem which can set up the wrong token bucket parameters by the lost packets on the practical networks with video packet loss. Therefore, this paper proposes a new scheme which reset up video file information to correct the wrong token bucket parameters in case of packet loss in practical networks with packet loss. Through the simulation, it was found that the proposed scheme would have better performance than the existing scheme in terms of overflow generation and packet loss.

• KIPS Transactions on Computer and Communication Systems
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• v.6 no.6
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• pp.255-260
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• 2017

The limitation of real-time video conferencing system is that the delay of network and buffering and the transmission of user information are not efficiently performed between systems, so real - time performance is not guaranteed completely. In order to overcome this problem, the study on the extension of the network infrastructure and the jitter delay is actively carried out, but the study on the buffering delay is insufficient. In this paper, we propose a frame-rate control buffer management (FRCB) scheme to solve the problem caused by buffering delay. The FRCB is used to prevent overflow and underflow of the buffer by adopting the two-stage buffer threshold of Fast-play THreshold (FTH) and Slow-play THreshold (STH). Therefore, it showed better performance than jitter buffer even under high CPU load, and showed that it is suitable for high quality real time video conferencing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.667-667
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• 2004

We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the QOS for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad performance on satellite communication network that is many of connecting user at same time. This paper propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.