• International Journal of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.166-172
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• 2020

Video streaming application such as YouTube is one of the most popular mobile applications. To adjust the quality of video for available network bandwidth, a streaming server provides multiple representations of video of which bit rate has different bandwidth requirements. A streaming client utilizes an adaptive bit rate scheme to select a proper video representation that the network can support. The download behavior of video streaming client player is governed by several parameters such as maximum buffer size. Especially, the size of the maximum playback buffer in the client player can greatly affect the user experience. To tackle this problem, in this paper, we propose the maximum buffer size optimization according to available network bandwidth and buffer status. Our simulation study shows that our proposed buffer size optimization scheme successfully mitigates playback stalls while preserving the similar quality of streaming video compared to existing ABR schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.6
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.11
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• pp.1066-1070
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• 2010

In wireless network, energy efficiency is an important design consideration for continuous multimedia streaming service. This paper proposes a new streaming system, called BACASS (Buffer-Aware and Content-Aware Streaming System) that working on the 802.11e U-APSD (Unscheduled Automatic Power Save). The BACASS leads the DP (Doze Period) of U-APSD for improving energy efficiency by utilizing the PSNR based on content-aware and client buffer occupancy that is hinged on a network-aware streaming system using SVC. The simulation results demonstrate the effectiveness of the proposed streaming system.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.82.1-82
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• 2001

It is known that existence of irregular transmission time delay is a major bottleneck for application of advanced robot control schemes to internet telerobotic systems. In the internet teleoperation system, the irregular transmission time delay causes a critical problem, which is unstable and inaccurate. This paper suggests a practical internet teleoperation system with streaming buffer system, which consists of a buffer, a buffer manager, and a control timer. The proposed system converts the irregular transmission time delay to constant. So, the system effectively transmits the control input to a remote site to operate a robot stably and accurately. This feature enables short control input interval. That means the entire system has ...

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.365-371
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• 2015

In wireless network, the amount of video streaming packet information in receiver replay buffer can be varied according tothe wireless network condition. By the effect, unforeseeable delay and jitter are generated and then busty video traffics can be made. If the amount of buffer information coming in receiver replay buffer is larger than the amount of a specific buffer information, buffer overflow is generated. Such a problem makes the image skip effect and packet loss, and then causes the quality degradation and replay discontinuity of the video streaming service in destination receiver. To solve the buffer overflow problem, this paper applies the token bucket for the busty traffic to the receiver terminal and analyzes the effect of the token bucket. The simulation result using NS-2 and JSVM shows that the proposed scheme with the token bucket has significantly better performance than the conventional scheme without the token bucket in terms of overflow generation number, packet loss rate and PSNR.

• Journal of the Korea Society of Computer and Information
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• v.15 no.11
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• pp.135-142
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• 2010

QoS in the proxy system are under heavy influence from interferences such as congestion, latency, and retransmission. Also, multi-level streaming services affects from temporal synchronization, which lead to degrade the service quality. This paper proposes a new segment-based buffer management mechanism which reduces performance degradation of streaming services and enhances throughput of streaming due to drawbacks of the proxy system. The proposed paper optimizes streaming services by: 1) Use of segment-based buffer management mechanism, 2) Minimization of overhead due to congestion and interference, and 3) Minimization of retransmission due to disconnection and delay. This paper utilizes fuzzy value $\mu$ and cost weight $\omega$ to process the result. The simulation result shows that the proposed mechanism has better performance in buffer cache control rate, average packet loss rate, and delay saving rate with stream relevance metric than the other existing methods of fixed segmentation method, pyramid segmentation method, and skyscraper segmentation method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.751-759
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• 2009

Due to the advance of WLAN technology, the use of the multimedia service such as the video streaming service has been increased in the home network. However, we need to study the method which decreases the transmission delay and the frame loss rate to provide QoS of the video streaming service. Therefore, this paper proposes an active buffer management mechanism to guarantee QoS of the streaming service in IEEE 802.11e EDCA. The proposed protocol discards the frame in the HoL of the buffer based on the importance of each frame and the virtual transmission delay of frame newly arriving at the buffer. In the simulation results, the proposed algorithm not only decreases the frame loss probability of important I and P frames but also stabilizes the transmission delay. It may increase the QoS of video streaming services.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3856-3872
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• 2018

HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.