• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12B
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• pp.735-742
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• 2007

We propose an effective packet marking algorithm for video streaming in DiffServ network. Because legacy packet markers such as srTCM(single rate three color marker) cannot distinguish the importance of packet, these markers can cause quality degradation of streaming during the network congestion period. Recently proposed TMS (Two Marker System) [4] shows effectiveness in such scenario that video streaming service is struggling with other types of service traffic. However, if many video streaming services co-exist in DiffServ network and result in competition among themselves, then both legacy packet markers and even TMS cannot prevent drastic streaming quality degradation. To cope with this, we suggest A-TCPM (Adaptive time sliding window Three Color Marker) algorithm. In this algorithm, an A-TCPM module decides the color of a racket based upon the probability which is lead by current channel status and frame importance ratio. Simulation results show that proposed A-TCPM algorithm can enhance streaming service quality especially when overbooked video streaming sessions struggle with themselves.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.280-286
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• 2004

The widespread use of the Internet and the maturing of digital video technology have led to an increase in various streaming media application. However, new classes of hosts such as mobile devices are gaining popularity, while the transmission became more heterogeneous. Due to the characteristics of mobile networks such as low speed, high error bit rate, etc., the applications over the wireless channel have different needs and limitations from desktop computers. An intermediary between two communicating endpoints to hide the heterogeneous network links is thought as one of the best approaches. In this paper, we adopted the concept of inter-packet gap and the sequence number between continuously received packets as the error discriminator, and designed an adaptive packet sizing mechanism to improve the network efficiency under varying channel conditions. Based on the proposed mechanism, the packetization scheme with error protection is proposed to scalable encoded video delivery. Finally, simulation results reveal that our proposed mechanism can react to the varying BER conditions with better network efficiency and gain the obvious improvement to video quality for stored MPEG-4 FGS video delivery.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.614-622
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• 2006

In this paper, we propose an efficient packetization method to reduce the packetization overhead. For the purpose, we firstly verify the relationship between packet length and packet loss rate. The empirical results show that as the packet length is larger than the path MTU, the packet loss rate is drastically increased, producing poor visual quality at the receiver side. However, as the length of the packet is reduced, we should transmit more packets per frame and the packetization overhead will be increased. This increase in the packetization overhead reduces the number of bits allocated to the video data, resulting in the low visual quality. Therefore, each packet should be packetized to have the packet length close to the path MTU. In this paper, we show that the this process of the packetization with the constraint on the packet length is very similar to the dynamic storage allocation in the operating system. We had thoroughly surveyed the dynamic storage allocation methods used in the recent operating systems and propose to use the allocation methods for the video packetization. We empirically show that the proposed method can reduce the packetization overhead upto 28.3%, compared with the conventional sequential packetization method which have been widely used in Internet video transmission.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.44 no.1
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• pp.94-101
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• 2007

Streaming video is expected to become a key service in the developing heterogeneous wireless network. However, sufficient quality of service is not offered to video applications because of bursty packet losses. An effective solution for packet loss in wireless network is to perform a proper concealment at the receiver. However, most concealment methods can not conceal effectively the consecutively damaged macro blocks, since the neighboring blocks are lost. In the previous work, bidirectional motion vector tracking (BMVT) method has been proposed which uses the moving trajectory feature of the damaged macro blocks. In this paper, a channel-adaptive redundancy coding method for the better BMVT error concealment is presented. The proposed method provides enhanced video quality at the cost of a little bit overhead in the wireless error-prone network.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.579-603
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• 2018

In this paper we propose a QoS-enhanced intelligent stochastic optimal fair real-time packet scheduler, QUEST, for 4G LTE traffic in routers. The objective of this research is to maximize the system QoS subject to the constraint that the processor utilization is kept nearly at 100 percent. The QUEST has following unique advantages. First, it solves the challenging problem of starvation for low priority process - buffered streaming video and TCP based; second, it solves the major bottleneck of the scheduler Earliest Deadline First's failure at heavy loads. Finally, QUEST offers the benefit of arbitrarily pre-programming the process utilization ratio.Three classes of multimedia 4G LTE QCI traffic, conversational voice, live streaming video, buffered streaming video and TCP based applications have been considered. We analyse two most important QoS metrics, packet loss rate (PLR) and mean waiting time. All claims are supported by discrete event and Monte Carlo simulations. The simulation results show that the QUEST scheduler outperforms current state-of-the-art benchmark schedulers. The proposed scheduler offers 37 percent improvement in PLR and 23 percent improvement in mean waiting time over the best competing current scheduler Accuracy-aware EDF.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.11C
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• pp.1068-1075
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• 2005

In this paper we propose a video coding method and associated packetization and decoding methods for error resilient transmission over the Internet. The proposed method re-organizes the input image into several mutually similar subimages. For this case, if the one of the subimage is lost in the network, the lost one is recovered by the proposed error concealment method which uses the correctly received other subimages. The performance of the proposed method is confirmed by the empirical results. The proposed method is not limited to the Internet communications but is applicable to the other packet-based networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.11C
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• pp.655-661
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• 2011

The transmission issues for the scalable video coding extension of H.264/AVC (H.264 SVC) video has been widely studied. In this paper, we propose an objective video-quality assessment metric based on no-reference for H.264 SVC using scalability information. The proposed metric estimate the perceptual video-quality reflecting error conditions with the consideration of the motion vectors, error propagation patterns with the hierarchical prediction structure, quantization parameters, and number of frame which damaged by packet loss. The proposed metric reflects the human perceptual quality of video and we evaluate the performance of proposed metric by using correlation relationship between differential mean opinion score (DMOS) as a subjective quality and proposed one.