• Journal of Korea Multimedia Society
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• v.13 no.3
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• pp.349-358
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• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.4
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• pp.1768-1789
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• 2016

In contemporary video streaming systems based on HTTP protocol, video players at the client side are responsible for adjusting video quality to network conditions and user expectations. However, when multiple video clips are streamed simultaneously, an intricate application logic implemented in the video players overlays the TCP mechanism which is responsible for a balanced access to a shared network link. As a result, some video players may not obtain a fair share of network throughput and may be vulnerable to an unstable video bit-rate. Therefore, we propose to simplify the algorithms implemented in the video players, which are responsible for the adjustment of video quality and constrain their functionality only to sending feedback to a server about a state of the player buffer. The main logic of the system is shifted to the server, which is now responsible for bit-rate selection and prioritisation of the video streams transmitted to multiple clients. To verify our proposition, we performed several experiments in a laboratory environment which show that when the server cooperates with the clients, the video players experience fewer quality switches and the system achieves better fairness when allocating network throughput among the video players. However, this comes at the cost of worse utilisation of network bandwidth.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.135-142
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• 2016

When video service is performed in simulation using NS-2(Network Simulation-2), the video replay is performed as the received frame order. In the existing video replay method based on the received frame order, as the frame orders of receiver and transmitter are different, the receiver buffer does not have the effect that the packets between the frames of transmitter buffer holds a regular size and packet dense and sparsity phenomenon in the receiver buffer is made by the irregular packet size due to the unpredictable reversed order of received partial frames. The above dense and sparsity phenomenon increases the probability of buffer overflow and underflow generation. To prevent these problems, the proposed frame receive order relocation method adds an extra replay buffer which rearranges the order of receive frame as the order of transmit frame, so it has the effect that the packets between the transmit frames keeps a regular size. Through the simulation using NS-2 and JSVM(Joint Scalable Video Model), the generation number of buffer overflow and underflow, and PSNR(Required Peak Signal to Noise Ratio) performance between the existing method and proposed method were compared. As a result, it was found that the proposed method would have better performance than the existing method.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.1
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• pp.30-40
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• 2010

The hard disk is one of the most frequently used storage in IPTV sep-top box. It has large storage capacity and provides fast I/O speed compared to its price whereas it causes high power consumption due to mechanical characteristics of spindle motor. In order to play streaming data in the set-top box, spindle motor of hard disk keeps active mode and it causes high power consumption. In this paper, We propose an offset-buffering and multi-mode spin-down method to reduce power consumption for streaming data playback. The offset-buffering inspects the user's viewing pattern and performs buffering based on the analysis of viewing pattern. So, it can maintain the status of spindle motor as idle mode for long time. Besides, it can reduce power consumption by spinning down according to offset-buffer size. The experimental result shows that proposed offset-buffering and multi mode spin-down method is about 28.3% and 12.5% lower than the full-Buffering method in terms of the power consumption and spin-down frequency, respectively.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• Journal of KIISE:Information Networking
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• v.35 no.3
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• pp.173-182
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• 2008

Promoting quality of streaming service in wireless networks has attracted intensive research over the years. Instable wireless channel condition causes high transmission delay and packet loss, due to fading and interference. Therefore, they lead to degrade quality of video streaming service. The IEEE 802.11 Working Group is currently working on a new standard called IEEE 802.11e to support quality of service in WLANs. And several schemes were proposed in order to guarantee QoS. However, they are not adaptable to network condition. Accordingly, they suffered video quality degradation, due to buffer overflow or packet loss. In this paper, to promote quality of video streaming service in WLANs, we propose a cross-layer architecture based on IEEE 802.11e EDCA model. Our cross-layer architecture provides differentiated transmission mechanism of IEEE 802.11e EDCA based on priority of MPEG-4 video frames and adaptively controls the transmission rate by dropping video frames through the efficient bandwidth estimation based on distinction of each AC. Through the simulation, proposed scheme is shown to be able to improve end-to-end qualify for video streaming service in WLANs.

• Journal of Broadcast Engineering
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• v.22 no.5
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• pp.618-631
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• 2017

In this paper, we propose a modification of fuzzy logic based DASH adaptation algorithm(FDASH) for seamless media service in time-varying network conditions. The proposed algorithm selects more appropriate bit-rate for the next segment by the modification of the Fuzzy Logic Controller(FLC) and reduces the number of video bit-rate changes by applying Segment Bit-rate Filtering Module(SBFM). Also, we apply the Start Mechanism for clients not to watch the low quality videos in the very beginning stage of streaming service and add the Sleeping Mechanism to avoid any buffer overflow expected. Ultimately, we verified by using NS-3 Network Simulator that the proposed method shows better performance compared to FDASH. According to the experimental results, there is no buffer underflow/overflow within the limited buffer size, which is not guaranteed in FDASH on the other hand. Also, we confirmed that mFDASH has almost the same level of average video quality against FDASH and reduces about 50% of number of video bit-rate changes compared to FDASH in Point-to-Point network and Wi-Fi network.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.6
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• pp.1869-1880
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• 2014

Wireless communication is one of most active areas in modern communication researches, QoS (Quality of Service) assurance is very important for wireless communication systems design, especially for applications such as streaming video etc., which requires strict QoS assurance. The modern wireless networks multi-layer protocol stack structure results in QoS metrics layered and acting in cascade and QoS metrics vertical mapping between protocol layers. Based on virtual buffer between protocol layers and queuing technology, a unified layered QoS mapping framework is proposed in this paper, in which we first propose virtual queue concept, give a novelty united neighboring protocol layers QoS metric mapping framework, and analysis method based on dicerete-time Markov chain, and numerical results show that our proposed framework represents a significant improvement over previous model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2B
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• pp.137-145
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• 2006

In this paper, we propose a multicast P2Patching technique in the application layer. The P2Patching technique serves VOD stream effectively with Patching in P2P environment. The P2Patching provides multicast tree construction technique that reduces the server load and minimizes the start delay with extended multicast technique. And we provide a fast recovery technique by tree failure and dynamic buffering scheme that guarantees the continuous streaming by frequent tree disconnections. Comparing the method with that of $P2Cast^{[12]}$, we obtained the better performance by our scheme in terms of average join count, service rejection probability, tree recovery failure and buffer starvation. The average about 16$\%$ of the improvement is shown by the simulation. Most of all, the performance of buffer starvation and average join count shows a significant improvement than that of P2Cast.