• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.6
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2075-2078
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• 2002

Internet became the most popular network in spite of its weakness in realtime multimedia service. Many experts believe that the Internet has the potential to become the main multimedia distribution network of the near future. Currently, it does not provide any (BoS guarantees and, even when it does, guaranteed quality delivery of video may turn out to be too expensive. Unavoidable packet losses and delay jitter caused by congestion in a best effort delivery environment require use of intelligent transport techniques for effective video delivery. According to market needs of better quality of service (QoS) fur realtime multimedia services over Internet, they have been standardizing RSVP, IntServ, and DiffServ This paper combines the benefits of QoS mechanisms such as RSVP/IntServ with scalable video encoding. We propose that more important bit stream is given more priority such that limited network resources are guaranteed far the stream. Various prioritizing approaches are proposed and compared to normal approach by using Network Simulator. The calculated QoS parameters such as packet loss rate are used to calculate degree of degradation in video quality. In this Paper, proposed methods can be implemented adaptively to Von protocol, such as H.323, SIP.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.97-100
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• 1995

In transmission of image sequences over ATM networks, the packet loss problem and channel sharing efficiency are important. As a possible solution two-layer video coding methods have been proposed. These methods transmit video information over the network with different levels of protection with respect to packets loss. In this paper, a two-layer coding method using pyramid structure is proposed and several realizations of two-layer video coding methods are presented and their performances are compared.

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.136-141
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• 2007

The frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) for applications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the telecom network. As telecom networks evolve from circuit to packet switching, proper synchronization algorithm should be meditated for IP networks to achieve performance quality comparable to that of legacy circuit-switched networks. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. But, The ToP can be affected by impairments of a network such as packet delay variation. This paper proposes the efficient method to minimize the expectable delay variation when ToP synchronizes the distributed clocks. The simulation results are presented to demonstrate the improved performance case when the efficient ToP transmit algorithm is applied.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5C
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• pp.490-496
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• 2006

H.264 is the most recent video coding standard, containing improved error resilience tools than previous video compression schemes. This paper shows an analysis of the dependency of error concealment (EC) performance on the expected number of correctly received neighboring macroblock(MB)s for a lost MB, applying error concealment schemes to the raster scan mode that is used in the previous video coding standard and the flexible macroblock ordering (FMO) which is one of error-resilience tools in H.264. We also present simulation results and performance evaluation with various packet loss rates. Simulation results show that the FMO mode provides better EC performances of $1{\sim}9dB$ PSNR improvements compared to the raster scan mode because of larger expected number of correctly received neighboring MBs. The PSNR improvement by FMO mode becomes higher as the intra-frame period is larger and the packet loss rate is higher.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.192-194
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• 2005

Acceptance of the international standards for video compression, such as H.261, MPEG-1 and MPEG-2, along with the developments in video codec hardware, has created an explosion of application. Among these, the long time quest for long-distance digital video transmission causes an increasing interest in transporting compressed video over networks which are nontraditional for this purpose, including asynchronous transfer mode networks, the Internet, and cellular and wireless channels. Transmission of compression video over packet network is improved for error resilience. And layered video coding techniques improves error resilience. We present a efficient method of scalable video coding for low bandwidth.

• Journal of Communications and Networks
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• v.7 no.2
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• pp.144-156
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• 2005

Video streaming over wireless networks is challenging due to node mobility and high channel error rate. In this paper, we propose a multi-source video streaming (MUVIS) system to support high quality video streaming service over IEEE 802.1l-based wireless networks. We begin by collocating a streaming proxy with the wireless access point to help leverage both the media server and peers in the WLAN. By tracking the peer mobility patterns and performing content discovery among peers, we construct a multi-source sender group and stream video using a rate-distortion optimized scheme. We formulate such a multi-source streaming scenario as a combinatorial packet scheduling problem and introduce the concept of asynchronous clocks to decouple the problem into three steps. First, we decide the membership of the multisource sender group based on the mobility pattern tracking, available video content in each peer and the bandwidth each peer allocates to the multi-source streaming service. Then, we select one sender from the sender group in each optimization instance using asynchronous clocks. Finally, we apply the point-to-point rate-distortion optimization framework between the selected sender-receiver pair. In addition, we implement two different caching strategies, simple caching simple fetching (SCSF) and distortion minimized smart caching (DMSC), in the proxy to investigate the effect of caching on the streaming performance. To design more realistic simulation models, we use the empirical results from corporate wireless networks to generate node mobility. Simulation results show that our proposed multi-source streaming scheme has better performance than the traditional server-only streaming scheme and that proxy-based caching can potentially improve video streaming performance.

• ETRI Journal
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• v.32 no.2
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• pp.273-280
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• 2010

The overall channel capacity of a multihop wireless path drops progressively over each hop due to the cascading effect of noise and interference. Hence, without optimal rate adaptation, the video quality is expected to degrade significantly at any client located at a far-edge of an ad-hoc network. To overcome this limitation, decoding and forwarding (DF), which fully decodes codewords at each intermediate node, can be employed to provide the best video quality. However, complexity and memory usage for DF are significantly high. Consequently, we propose syndrome-based partial decoding (SPD). In the SPD framework an intermediate node partially decodes a codeword and relays the packet along with its syndromes if the packet is corrupted. We demonstrate the efficacy of the proposed scheme by simulations using actual 802.11b wireless traces. The trace-driven simulations show that the proposed SPD framework, which reduces the overall processing requirements of intermediate nodes, provides reasonably high goodput when compared to simple forwarding and less complexity and memory requirements when compared to DF.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.385-389
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• 2009

Multi-hop wireless mesh networks (WMNs) suffer from significant packet losses due to insufficient available bandwidth and high channel error probability. To conquer packet losses, end-to-end (E2E) error control schemes have been proposed. However, in WMNs, E2E error control schemes are not effective in adapting to the time-varying network condition due to large delay. Thus, in this paper, we propose a network-adaptive error control for video streaming over WMNs that flexibly operates E2E and hop-by-hop (HbH) error control according to network condition. Moreover, to provide lightweight support at intermediate nodes for HbH error control, we use path-partition-based adaptation. To verify the proposed scheme, we implement it and evaluate its transport performance through MPEG-2 video streaming over a real IEEE 802.11a-based WMN testbed.

• Smart Media Journal
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• v.6 no.2
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• pp.9-14
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• 2017

A packet loss in wireless environments causes a severe degradation of video quality in video communications. In this paper, a novel video error concealment algorithm is presented by combining boundary errors and a block reliability measure. The block reliability measure decides the reliability of a block by checking residual errors of a block. In the proposed approach, a motion vector of a missing unreliable block in an inter coded frame is obtained initially based on the motion vector of the same block in the reference frame. Furthermore, if the block in the reference frame is unreliable according to the reliability measure, a new motion vector is decided based on block boundary errors around the initial motion vector. According to our simulations, the proposed approach shows promising results for error concealment in error-prone wireless environments.