• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.15-19
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• 2000

This paper is about MPEG-4 error resilience tools of video streaming on packet service(ex, Internet). It is need to packetization for MPEG-4 video transport by packet unit on MPEG-4 system, this paper suggest packetization method of minimizing packet error on packet service[1]. FEC(Forward Error Correction) and retransmission is usually used for recovery of packet loss, and this paper suggest applying these method to DMIF(Delivery Multimedia Integration Framework) for minimizing packet loss[2].

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.530-541
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• 2011

In this paper, we propose an effective network-adaptive ARQ based error control scheme to provide video streaming services through IP networks where packet error usually occurs. If time delay and feedback channel are allowed, client can request server to retransmit lost packets through IP networks. However, if retransmission is unconditionally requested without considering network condition and number of simultaneous feedback messages, retransmitted packets may not arrive in a timely manner so that decoding may not occur. In the proposed ARQ, a client conditionally requests retransmission based on assumed network condition, and it further determines valid retransmission time so that effective ARQ can be applied. In order to verify the performance of the proposed adaptive ARQ based error control, NIST-Net is used to emulate packet-loss network environment. It is shown by simulations that the proposed scheme provides noticeable error resilience with significantly reduced traffics required for ARQ.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.798-806
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• 2005

In this paper, we suggest a game system that can support network modules for multi-platform based video games, and built a system that can convert from a single-user game to multi-user game. In this system, we bring in an initial delay buffering scheme on clients to handle any periods of latency occurring from the load fluctuation in a network, when a real-time game is played, and shows that stable play for a game is achieved as the result of the scheme. This paper also presents a retransmission algorithm based on the history of game commands to handle drawbacks of UDP mechanism. And, we evaluate the network delay and packet loss using the simulation tool NS2, and shows the case of 0.3 second buffer delay is the most suitable for recovery.

• Journal of Internet Computing and Services
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• v.7 no.6
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• pp.143-155
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• 2006

In this paper, we suggest a network game system that can support video game based on multi-platform for multi-user video game, Latency occurring from the load fluctuation in realtime network game is overcomed by using an initial delay buffering scheme on client, when a real-fire game is played, and shows that stable play for a game is achieved as the result of the scheme. Also, We suggest a retransmission algorithm based on the history of game commands, and this Algorithm supplement shortcomings for packet loss and pocket error on UDP communication.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.322-334
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• 2014

Multimedia content is very sensitive to packet loss and therefore multimedia streams are typically protected against packet loss, either by supporting retransmission requests or by adding redundant forward error correction (FEC) data. However, the redundant FEC information introduces significant additional bandwidth requirements, as compared to the bitrate of the original video stream. Especially on wireless and mobile networks, bandwidth availability is limited and variable. In this article, an adaptive FEC (A-FEC) system is presented whereby the redundancy rate is dynamically adjusted to the packet loss, based on feedback messages from the client. We present a statistical model of our A-FEC system and validate the proposed system under different packet loss conditions and loss probabilities. The experimental results show that 57-95%bandwidth gain can be achieved compared with a static FEC approach.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1061-1064
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• 1999

UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission In general, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol(TCP) traffic as opposed to UDP's simplicity, TCP adopts a unique flow control in this paper, I report new methods to minimize a udp packet loss considering TCP flow control on the real-time application the better performance of real time application can be obtained when they reduce a packet size and FIFO buffer scheduling method competing with TCP bandwidth for the bandwidth and buffering.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.9
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• pp.4572-4586
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• 2019

Mobile Edge Computing (MEC) and Information-Centric Networking (ICN) are essential network architectures for the future Internet. The advantages of MEC and ICN such as computation and storage capabilities at the edge of the network, in-network caching and named-data communication paradigm can greatly improve the quality of video streaming applications. However, the packet loss in wireless network environments still affects the video streaming performance and the existing loss recovery approaches in ICN does not exploit the capabilities of MEC. This paper proposes a Deep Learning based Loss Recovery Mechanism (DL-LRM) for video streaming over MEC based ICN. Different with existing approaches, the Forward Error Correction (FEC) packets are generated at the edge of the network, which dramatically reduces the workload of core network and backhaul. By monitoring network states, our proposed DL-LRM controls the FEC request rate by deep reinforcement learning algorithm. Considering the characteristics of video streaming and MEC, in this paper we develop content caching detection and fast retransmission algorithm to effectively utilize resources of MEC. Experimental results demonstrate that the DL-LRM is able to adaptively adjust and control the FEC request rate and achieve better video quality than the existing approaches.

• Journal of Broadcast Engineering
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• v.21 no.5
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• pp.662-671
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• 2016

In this paper, we propose a wireless video transmission system, providing medium and long communication distance in the 5.8GHz band to decrease the loss of video data. Beamforming with multiple antennas in a radio environment to enable more than 10km communication distance and an image transmission method that combines the retransmission to MCS(modulation and coding scheme) are applied. By applying the proposed method in the indoor environment of 80dB attenuation using the payload size of 1000 bytes, the packet loss rate of at least 0.92% is achieved through the FPING with the time-out of 10ms. The transmission distance of 21.2Km is achieved with the transmission rate of 13Mbps in the outdoor environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.5
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• pp.435-446
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• 2014

Even if it is important to transmit high rate multimedia information without any transmission errors for surveillance systems, it is difficult to achieve error-free transmission due to infra-less adhoc networks. In order to reduce the transmission errors furthermore, additional signal overheads or retransmission of signals should be required, but they may lead to transmission delay. This paper represents a study on low delay and adaptive video transmission for the unmanned surveillance systems by developing system protocols. In addition, we introduce an efficient and adaptive control algorithm using system parameters for exploiting unmanned surveillance system properly over multi-channels.