• 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].

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.950-958
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• 2007

This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8C
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• pp.810-823
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• 2005

The transmission of duplicate packets provides a great loss-resilience without undue time-delay in the video transmission over packet loss networks. But this method generally deteriorates the problem of traffic congestion because of the increased bit-rate required for duplicate transmission. In this paper, we propose an efficient packetization and duplicate transmission of video packets. The proposed method transmits only the video signal with high priority for each video macroblock that is quite small in volume but very important for the reconstruction of the video. The proposed method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. The duplicated high priority data of the Previous video slice is transmitted as a Piggyback to the data Packet of the current video slice. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

• ETRI Journal
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• v.30 no.3
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• pp.441-450
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• 2008

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real-time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter-layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter-media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter-media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.181-188
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• 2016

Video streaming traffics which are arrived in receiver have irregular traffic patterns by many problems over the network path. Particularly, if these received traffics enter into replay buffer without any operation, the overflow and underflow effects are made according to the buffer status. There was an existing scheme which automatically set up token bucket parameters using the video information file under the assumption of the lossless packet on network. The existing scheme has a problem which can set up the wrong token bucket parameters by the lost packets on the practical networks with video packet loss. Therefore, this paper proposes a new scheme which reset up video file information to correct the wrong token bucket parameters in case of packet loss in practical networks with packet loss. Through the simulation, it was found that the proposed scheme would have better performance than the existing scheme in terms of overflow generation and packet loss.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.118-123
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• 2009

In this paper, we propose packet selection and significance based interval allocation algorithm for real-time streaming service. In real-time streaming of inter-frame (and layer) coded video, minimizing packet loss does not imply maximizing QoS. It is true that packet loss adversely affects the QoS but one single packet can have more impact than several other packets. We exploit the fact that the significance of each packet loss is different from the frame type it belongs to and its position within GoP. Using packet dependency and PSNR degradation value imposed on the video from the corresponding packet loss, we find each packet's significance value. Based on the packet significance, the proposed algorithm determines which packets to send and when to send them. The proposed algorithm is tested using publicly available MPEG-4 video traces. Our scheduling algorithm brings significant improvement on user perceivable QoS. We foresee that the proposed algorithm manifests itself in last mile connection of the network where intervals between successive packets from the source and to the destination are well preserved.

• Journal of information and communication convergence engineering
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• v.14 no.4
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• pp.240-245
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• 2016

Scalable video streaming over wireless networks has many challenges. The most significant challenge is related to packet loss. To overcome this problem, in this paper, we propose an unequal loss protection (ULP) method using a new forward error correction (FEC) mechanism for robust scalable video streaming over wireless networks. For an efficient FEC assignment considering video quality, we first introduce a simple and efficient performance metric, the layer-based recovery rate (LRR), for quantifying the unequal error propagation effects of the temporal and quality layers on the basis of packet losses. LRR is based on the unequal importance in both the temporal and the quality layers of a hierarchical scalable video coding structure. Then, the proposed ULP-LRR method assigns an appropriate number of FEC packets on the basis of the LRR to protect the video layers against packet lossy network environments. Compared with conventional ULP algorithms, the proposed ULP-LRR algorithm demonstrates a higher performance for various error-prone wireless channel statuses.

• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.4
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• pp.13-21
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• 2022

Recently, in wireless video sensor networks(WVSN), various schemes for efficient video data transmission have been studied. In this paper, a layer based cooperative relaying(LCR) algorithm is proposed for minimizing scalable video transmission distortion from packet loss in WVSN. The proposed LCR algorithm consists of two modules. In the first step, a parameter based error propagation metric is proposed to predict the effect of each scalable layer on video quality degradation at low complexity. In the second step, a layer-based cooperative relay algorithm is proposed to minimize distortion due to packet loss using the proposed error propagation metric and channel information of the video sensor node and relay node. In the experiment, the proposed algorithm showed that the improvement of peak signal-to-noise ratio (PSNR) in various channel environments, compared to the previous algorithm(Energy based Cooperative Relaying, ECR) without considering the metric of error propagation.The proposed LCR algorithm minimizes video quality degradation from packet loss using both the channel information of relaying node and the amount of layer based error propagation in scalable video.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.521-530
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• 1989

Broadband ATM(Asynchronous Transfer Mode) networking techniques based on lightwave technology and high speed integrated circuits appear to be the choice of transport technology for broadband ISDN. Among other problems, the issue of video transport over broadband packet(ATM) networks still requries further investigation. In this paper, the problems of transporting video signals over a broaband packet network are investigated together with possible solutions. In particular, clock recovery packet loss compensation and transport technique based on hierarchical video coding scheme are described in detail. This would allow efficient bandwidth sharing and minimum degradation in video quality.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.900-903
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• 2002

This paper proposes the use of wavelet packet transform in a transform based video coding scheme, which is mainly used in low/very low bit-rate video coding schemes i.e. H.263 standard. In the experiments, the discrete cosine transform in the video coding scheme is replaced by the wavelet packet transform, and the improved performance in term of peak signal to noise ratio is measured and compared with the results obtained from the coding scheme implementing the ordinary wavelet transform. The experimental results show an impressive improvement obtained from the use of wavelet packet transform.