Content-Aware Selective Retransmission Scheme in Heavy Loaded Wireless Networks (original) (raw)
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Retransmission-based error recovery is the simplest technique to minimize the overall packet loss ratio in order to increase the quality of the applications. Multimedia applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission can be also used for loss recovery in media applications, but the number of retransmissions is limited by the playout buffer and the recent network delay.
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In wireless networks, congestion control, alone, may not be enough to ensure good quality of multimedia streaming and efficient utilization of the network. Packet losses due to the high bit error rate not only degrade the multimedia quality, but render the current congestion control algorithms as inefficient: these algorithms back-off on every packet loss even when there is no congestion.
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In the past few years we have witnessed an explosive growth in the usage of media streaming applications. The newly appeared audio/video applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission-based error recovery is considered inappropriate for multimedia applications, because of its latency. However, this solution can be attractive because it requires minimal network bandwidth, processing cost and efficiently improves the quality. Despite its latency, retransmission can be used successfully in many cases, especially if playout buffering is employed. Only the successfully retransmitted packets will improve the quality parameters of the multimedia stream, therefore it is worth to examine which packets should be retransmitted. In this paper a source controlled selective retransmission algorithm is presented with a decision algorithm based on the actual RTT and sending rate determined by the TFRC. In our scheme the transmitter determines the playout delay caused by the playout buffer using the proposed Flood method. The needed information about the network congestion state and the network delay are provided by the TFRC (TCP Friendly Rate Control) algorithm. Our proposal does not need additional administration messages because the decision procedure and its inputs are at the transmitter. The obtained results show that significant quality improvement can be achieved with the proposed selective retransmission scheme.
Selective retransmission of MPEG video streams over IP networks
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Internet multimedia streaming is becoming increasingly popular to access multimedia information. Two major issues arise in spite of its popularity. First, limited bandwidth restricts high bit rate video transmission. Second, wireless transmission by its nature may introduce higher rate of errors during transmission. The frequent errors should cause deterioration of the quality of multimedia streams. In this paper we present
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Video-streaming services, such as Internet protocol television, promising the delivery of multimedia contents over wireless access networks to clients whenever and wherever, are becoming more and more popular. However, scarce radio resources, lossy characteristics of wireless links and high bandwidth demands pose the never-ending challenges for provisioning of real-time streaming services over wireless networks in a timely and reliable manner. Furthermore, a wireless channel may suffer from interference and multipath fading, which may cause random packet losses. In addition, wireless link layer does not provide a retransmission mechanism for multicast/broadcast traffic. This would significantly impact the clients’ quality of experience of streaming services. Traditional unicast retransmission solutions improve client’s quality, at the bandwidth expense, because every lost packet must be retransmitted separately. This chapter presents and practically evaluates a retransmission scheme...
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Efficient resource allocation is a key factor to improve the efficiency of video transmission over wireless channels. To increase the number of correctly received video frames at the decoder, it is desirable to reduce the video source rate while increasing error protection when the wireless channel is anticipated to be bad or when the receiver buffer is approaching starvation. In this study, we introduce a retransmission-based adaptive source-channel rate control scheme for video transmission over wireless packet networks. In this scheme, the level of adaptiveness is optimized to reduce the bandwidth requirement while guaranteeing delay and loss bounds. The proposed scheme has the advantage of providing closed-form expressions of the near-optimum parameters of the proposed model, which are then fed back to the transmitter to scale both the source and channel rates adaptively. Simulation and numerical investigations are carried out to verify the adequacy of the analysis and study the impact of the adaptive process on the continuity of the video playback process.
Broadband Communications Networks - Recent Advances and Lessons from Practice, 2018
Video-streaming services, such as Internet protocol television, promising the delivery of multimedia contents over wireless access networks to clients whenever and wherever, are becoming more and more popular. However, scarce radio resources, lossy characteristics of wireless links and high bandwidth demands pose the never-ending challenges for provisioning of real-time streaming services over wireless networks in a timely and reliable manner. Furthermore, a wireless channel may suffer from interference and multipath fading, which may cause random packet losses. In addition, wireless link layer does not provide a retransmission mechanism for multicast/broadcast traffic. This would significantly impact the clients' quality of experience of streaming services. Traditional unicast retransmission solutions improve client's quality, at the bandwidth expense, because every lost packet must be retransmitted separately. This chapter presents and practically evaluates a retransmission scheme for video-streaming services over last mile wireless networks. It is based on network coding techniques that increase the overall performance by means of reducing the number of physical transmissions, in comparison to traditional unicast retransmission approach, resulting in reduced bandwidth consumption. Thus, the Internet service providers can increase the number of clients over the same infrastructure or, alternatively, offer more services to the clients.
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In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine.