• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.271-274
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• 2003

This paper presents a robust error concealment method for burst-packet-loss networks. The proposed error concealment algorithm can reduce the computational complexities of the existing error concealment methods. Moreover, experimental results show that the proposed method produces the better video quality than the conventional boundary matching algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.11
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• pp.1472-1480
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• 2016

When disasters such as earthquake, flood, typhoon, fire and terrorism are occurred a video management system is required which can shoot video on the disaster site, and send it to a server located in the command center that can grasp at a glance the site conditions. In this paper, to know the performance elements information needed to construct the video management system. we presented a method of configuring an ad hoc network based on Wi-Fi direct communication. And we also evaluated the performance through built of environment for the video management system. The evaluated performance test items are 'packet flow tests according to the video compression ratios and the image sizes', 'packet flow tests according to the distance and traffic tests', 'packet flow tests according to whether LOS or not', and 'performance test of scenarios'.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.549-551
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• 2004

This paper used OPNET to simulate video streaming service a test IP network and MPLS network with the traffic shaping that have with CQ_ LLQ algorithm, LSP of fixed bandwidth, policy of limitation users and measures parameters such as delay, throughput, packet loss. To verify the performance of video streaming service in IP network and MPLS network, two scenario that have same topology and traffic source. One is the simulation for best-effort service in pure IP network. The other is the simulation for QoS-enabled service in MPLS Network. Based on simulation result, the MPLS network with CQ_ LLQ algorithm and fixed LSP show advantage of the video streaming service QoS, specially delay and packet loss

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.250-253
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• 2009

This paper presents NR (No Reference) Quality assessment method for IPTV or mobile IPTV. Because No Reference quality assessment method does not access the original signal so it is suitable for the real-time streaming service. Our proposed method use decoding parameters, such as quantization parameter, motion vector, and packet loss as a major network parameter. To evaluate performance of the proposed algorithm, we carried out subjective test of video quality with the ITU-T P.910 ACR (Absolute Category Rating) method and obtained the mean opinion score (MOS) value for QVGA 180 video sequence coded by H.264/AVC encoder. Experimental results show the proposed quality metric has a high correlation (84%) to subjective quality.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• ETRI Journal
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• v.43 no.3
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• pp.447-458
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• 2021

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.

• Proceedings of the IEEK Conference
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• 2002.06e
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• pp.83-86
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• 2002

In this paper while a router is routing all packet to the next hop, it inspects whether there is congestion on this current hop router or not and if the router discovers that it has some congestion, it informs that the packet is experienced to congestion. The packet arrived to next hop including some information about the congestion is processed first and it has wider bandwidth than another packet The amount of congestion is recorded to the DS field of IP header by congestion experience level. In the next hop when the packet including the congestion information is routed, the standard packet dropping ratio of the current router is changed in proportion to congestion experience that is recorded in IP header on of that. When the packet that has experienced congestion before is arrived, the router extends the drop threshold value not to drop the packet. It mean that transferring the audio or video stream, if the packet is already experienced the congestion in another hop, the router can provide the better service quality about 15∼25% than another.

• 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 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.