• Journal of Internet Computing and Services
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• v.17 no.1
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• pp.55-64
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• 2016

Existing Android applications for streaming video in real time are dependent on the codec, which composes the encoding function, and the version of Android operating system. Also, for streaming video in real time, most applications should be connected with a separate desktop PC. To overcome these disadvantages, we propose a new application, which records and streams video in real time. Specifically, the proposed application uses the flash video file format, which is the common media file format supported by various versions of Android operating system. Through experiments, we show that it is possible for the proposed application to record the video screens more than 20 frames per second and to stream it in real time while using the existing video encoding methods.

• The KIPS Transactions:PartA
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• v.11A no.3
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• pp.207-212
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• 2004

This paper provides the higher user QoS(Quality of Service) by analyzing media service model between server and client in single VoD(Video on Demand) environment and applying it to parallel VoD environment. Media service model is divided into the Client Pull, Server Push, and IPP(Interleaving Pull & Push) model. A server sends data based on client's request in the Client Pull Model. A server one-sidedly sends data without client's request in the Server Pull model. And the WP model unites above two models. For a parallel VoD environment, We built the PMSS system which provides the parallel media streaming services that one client is simultaneously served by several servers. In the single and parallel VoD environment, We compare and analyze the performance of service models with respect to network delay and data size in buffer. In this experiment, we found that IPP service model keeps the least network delay and stable client buffer in the parallel VoD environment. This result shows that PMSS can provide the more quality of service.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.721-732
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• 2000

Multimedia communication systems require not only high-performance computer hardwares and high-speed networks, but also a buffer management mechanism to process many data efficiently. Two buffer handling methods, Push and Pull, are commonly used. In the Push method, a server controls the flow of dat to a client, while in the Pull method, a client controls the flow of data from a server. Those buffering schemes can be applied to the data transfer between the packet receiving buffer, which receives media data from a network server, and media playout devices, which play the recived media data. However, the buffer management mechanism in client-sides mainly support either one of the Push or the Pull method. Consequently, they have some limitations to support various media playout devices. Futhermore, even though some of them support both methods, it is difficult to use since they can't provide a unified structure. To resolved these problems, in this paper, we propose an efficient and flexible Push/Pull buffer management mechanism at client-side. The proposed buffer management scheme supports both Push and Pull method to provide various media playout devices and to support buffering function to absorb network jitter. The proposed scheme can support the various media playback devices using a single buffer space which in consequence, saves memory space compared to the case that a client keeps tow types of buffers. Moreover, it facilitates the single buffer as a mechanism for the absorbing network jitter effectively and efficiently. The proposed scheme has been implemented in an existing multimedia communication system, so called ISSA (Integrated Streaming Service Architecture), and it shows a good performance result compared to the conventional buffering methods in multimedia communication environments.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.10
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• pp.2430-2447
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• 2013

In recent years, HTTP adaptive streaming (HAS) has attracted considerable attention as the state-of-the-art technology for video transport. HAS dynamically adjusts the quality of video streaming according to the network bandwidth and device capability of users. Content-Centric Networking (CCN) has also emerged as a future Internet architecture, which is a novel communication paradigm that integrates content delivery as a native network primitive. These trends have led to the new research issue of harmonizing HAS with the in-network caching provided by CCN routers. Previous research has shown that the performance of HAS can be improved by using the H.264/SVC(scalable video codec) in the in-network caching environments. However, the previous study did not address the misbehavior that causes video freeze when overestimating the available network bandwidth, which is attributable to the high cache hit rate. Thus, we propose a new SVC-based adaptation algorithm that utilizes a drop timer. Our approach aims to stop the downloading of additional enhancement layers that are not cached in the local CCN routers in a timely manner, thereby preventing excessive consumption of the video buffer. We implemented our algorithm in the SVC-HAS client and deployed a testbed that could run Smooth-Streaming, which is one of the most popular HAS solutions, over CCNx, which is the reference implementation of CCN. Our experimental results showed that the proposed scheme (SLA) could avoid video freeze in an effective manner, but without reducing the high hit rate on the CCN routers or affecting the high video quality on the SVC-HAS client.

• Journal of Internet Computing and Services
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• v.17 no.4
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• pp.19-26
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• 2016

In this paper, the dynamic buffering based smart OTT platform was proposed, and analyzed for adaptive bit-rate video delivery with the optimization of HLS (HTTP Live Streaming). This platform consists of the software platform between sever and client which detects the bandwidth capacity, and adjusts the quality of the streaming for multiple bit-rates resolutions. In order to apply adaptive buffering, two buffers are added to the basic HLS player, and each buffer is responsible for constantly buffering a previous and the next channels relative to the current channel. This adaptive transmitting with smart OTT platform is superior to delivering a static video file at a single buffering, because the video stream of adaptive double buffers can be switched streaming according to client's available network speed. As a result, this proposed smart OTT can be cooperated to the application of HLS server with segmented H.265 MPEG-2 TS video & m3u8 files with its information based on the optimized transmission channel state of live and VOD, and applied to PLC transmission, too.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2B
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• pp.334-344
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• 2010

Recently, due to the prevalence of various mobile devices and broadband wireless networks, a significant interests and demands for multimedia streaming services over the Internet have been increasing. However, it is difficult to transmit continuous multimedia stream when mobile terminals are moving. Therefore, in order to deploy mobile IPTV service in the broadband wireless network, efficient wireless resource utilization and seamless QoE (Quality of Experience) offers to the users are an important issue. In this paper, we propose a network based adaptive streaming scheme, called MARC (Mobile Adaptive Rate Control), which controls the quality of the video and rate of the video based on the status of the wireless channel. The proposed scheme uses awareness information of the wireless channel status and controls transmitting streaming video which is suitable for the wireless channel status and mobile station location, in order to provide a seamless video playback for mobile environment in addition to improving the quality of a streaming service. The proposed MARC scheme alleviates the discontinuity of video playback and allocates suitable client buffer in broadband wireless network. Simulation results demonstrate the effectiveness of our proposed scheme.

• Journal of IKEEE
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• v.24 no.1
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• pp.46-51
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• 2020

With Wi-Fi and Bluetooth connectivity becoming more common in today's handheld mobile devices, single-chip multi-radio combo-modules, which integrate two or more heterogeneous wireless radios on a single chip, are becoming more and more popular. The key requirement for combo solutions is that the quality of the user experience (QoE) must not be compromised by degrading connectivity performance. Therefore, characterizing and understanding the behaviour of combo-module is of vital importance to ensure this requirement in various environments. In this paper, we investigate the impact of the use of combo-modules on the performance of mobile video streaming over a Wi-Fi network. Our study reveals that the use of combo-modules incurs considerable side effects on QoE for mobile video streaming applications when the Wi-Fi and Bluetooth operate at the same time in the 2.4GHz ISM band. We reveal that rate-based adaptive algorithms, including the most popular adaptive bitrate streaming MPEG-DASH, is more severely affected by this phenomenon than buffer-based adaptive algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.517-525
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• 2009

We propose a quality guaranteed scalable video streaming service over the Internet using a new rate adaptation algorithm. Because video data requires much more bandwidth rather than other types of service, therefore, quality of video streaming service should be guaranteed while providing friendliness with other service flows over the Internet. To successfully provide this, we propose a framework for providing quality-guaranteed streaming service using two-channel transport layer and rate adaptation of scalable video stream. In this framework, baseline layer for scalable video is transmitted using TCP transport for minimum qualify service. Enhancement layers are delivered using TFRC transport with layer adaptation algorithm. The proposed framework jointly uses the status of playout buffer in the client and the encoding rate of layers in media contents. Therefore, the proposed algorithm can remarkably guarantee minimum quality of streaming service rather than conventional approaches regardless of network congestion and the encoding rate variation of media content.

• Journal of KIISE
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• v.43 no.7
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• pp.827-833
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• 2016

HTTP Adaptive Streaming (HAS) is a technique that adapts its video quality to network conditions for providing Quality of Experience. In the HAS approach, a video content is encoded at multiple bitrates and the encoded video content is divided into several video segments. A HAS player estimates the network bandwidth and adjusts the video bitrate based on estimated bandwidth. However, the segment scheduler in the conventional HAS player requests video segments periodically without considering TCP. If the waiting duration for the next segment request is quite long, the TCP connection can be initialized and it restarts slow-start. Slow-start causes the reduction in TCP throughput and consequentially leads to low-quality video streaming. In this study, we propose a TCP-aware segment scheduling scheme to improve performance of HAS service. The proposed scheme adjusts request time for the next video request to prevent initialization of TCP connection and also considers the point of scheduling time. The simulation proves that our scheme improves the Quality of Service of the HAS service without buffer underflow issue.