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On-the-fly erasure coding for real-time video applications  [PDF]
Pierre-Ugo Tournoux,Emmanuel Lochin,Jerome Lacan,Amine Bouabdallah,Vincent Roca
Computer Science , 2009, DOI: 10.1109/TMM.2011.2126564
Abstract: This paper introduces a robust point-to-point transmission scheme: Tetrys, that relies on a novel on-the-fly erasure coding concept which reduces the delay for recovering lost data at the receiver side. In current erasure coding schemes, the packets that are not rebuilt at the receiver side are either lost or delayed by at least one RTT before transmission to the application. The present contribution aims at demonstrating that Tetrys coding scheme can fill the gap between real-time applications requirements and full reliability. Indeed, we show that in several cases, Tetrys can recover lost packets below one RTT over lossy and best-effort networks. We also show that Tetrys allows to enable full reliability without delay compromise and as a result: significantly improves the performance of time constrained applications. For instance, our evaluations present that video-conferencing applications obtain a PSNR gain up to 7dB compared to classic block-based erasure codes.
Erasure Coding and Congestion Control for Interactive Real-Time Communication  [PDF]
Pierre-Ugo Tournoux,Tuan Tran Thai,Emmanuel Lochin,Jerome Lacan,Vincent Roca
Computer Science , 2012,
Abstract: The use of real-time applications over the Internet is a challenging problem that the QoS epoch attempted to solve by proposing the DiffServ architecture. Today, the only existing service provided by the Internet is still best-effort. As a result, multimedia applications often perform on top of a transport layer that provides a variable sending rate. In an obvious manner, this variable sending rate is an issue for these applications with strong delay constraint. In a real-time context where retransmission can not be used to ensure reliability, video quality suffers from any packet losses. In this position paper, we discuss this problem and motivate why we want to bring out a certain class of erasure coding scheme inside multimedia congestion control protocols such as TFRC.
An Error Resilient Coding Scheme for Transmitting H. 264 Real-time Video Streaming Over 3G Networks
一种适合3G网络传输的H.264 实时视频容错编码

DU Jian-chao,WU Cheng-ke,WANG Yang-li,XIAO Song,

中国图象图形学报 , 2005,
Abstract: To deliver H.264 real-time video streaming over 3G networks,an error resilient coding scheme is presented in this paper.The system first divides original sequence into two sub-sequences to encode separately and then transmits the resulting two bitstreams over different channels so that the effect of burst channel error on image quality can be decreased.The frames which are not correctly decoded are predicted by bidirectional weighted interpolation of the frames from the other sub-stream.In order to restore the decoding process of the sub-stream suffering transmission error,the reference picture of the other sub-streams correctly decoded is selected as reference for the error sub-stream so that the error propagation can be eliminated.The simulation results show that the proposed method can efficiently combat bit error and packet loss in wireless channels and improve the image quality.The performance of the proposed method gains more than 0.6dB compared with the related approaches.
Reliable Multi-Path Routing Schemes for Real-Time Streaming  [PDF]
Emin Gabrielyan,Roger D. Hersch
Mathematics , 2006,
Abstract: In off-line streaming, packet level erasure resilient Forward Error Correction (FEC) codes rely on the unrestricted buffering time at the receiver. In real-time streaming, the extremely short playback buffering time makes FEC inefficient for protecting a single path communication against long link failures. It has been shown that one alternative path added to a single path route makes packet level FEC applicable even when the buffering time is limited. Further path diversity, however, increases the number of underlying links increasing the total link failure rate, requiring from the sender possibly more FEC packets. We introduce a scalar coefficient for rating a multi-path routing topology of any complexity. It is called Redundancy Overall Requirement (ROR) and is proportional to the total number of adaptive FEC packets required for protection of the communication. With the capillary routing algorithm, introduced in this paper we build thousands of multi-path routing patterns. By computing their ROR coefficients, we show that contrary to the expectations the overall requirement in FEC codes is reduced when the further diversity of dual-path routing is achieved by the capillary routing algorithm.
Online multipath convolutional coding for real-time transmission  [PDF]
Tuan Tran Thai,Emmanuel Lochin,Jerome Lacan
Computer Science , 2012,
Abstract: Most of multipath multimedia streaming proposals use Forward Error Correction (FEC) approach to protect from packet losses. However, FEC does not sustain well burst of losses even when packets from a given FEC block are spread over multiple paths. In this article, we propose an online multipath convolutional coding for real-time multipath streaming based on an on-the-fly coding scheme called Tetrys. We evaluate the benefits brought out by this coding scheme inside an existing FEC multipath load splitting proposal known as Encoded Multipath Streaming (EMS). We demonstrate that Tetrys consistently outperforms FEC in both uniform and burst losses with EMS scheme. We also propose a modification of the standard EMS algorithm that greatly improves the performance in terms of packet recovery. Finally, we analyze different spreading policies of the Tetrys redundancy traffic between available paths and observe that the longer propagation delay path should be preferably used to carry repair packets.
Fault-Tolerant Real-Time Streaming with FEC thanks to Capillary Multi-Path Routing  [PDF]
Emin Gabrielyan
Computer Science , 2006,
Abstract: Erasure resilient FEC codes in off-line packetized streaming rely on time diversity. This requires unrestricted buffering time at the receiver. In real-time streaming the playback buffering time must be very short. Path diversity is an orthogonal strategy. However, the large number of long paths increases the number of underlying links and consecutively the overall link failure rate. This may increase the overall requirement in redundant FEC packets for combating the link failures. We introduce the Redundancy Overall Requirement (ROR) metric, a routing coefficient specifying the total number of FEC packets required for compensation of all underlying link failures. We present a capillary routing algorithm for constructing layer by layer steadily diversifying multi-path routing patterns. By measuring the ROR coefficients of a dozen of routing layers on hundreds of network samples, we show that the number of required FEC packets decreases substantially when the path diversity is increased by the capillary routing construction algorithm.
Transcoder Migration For Real Time Video Streaming Systems  [PDF]
Paul Farrow,Martin Reed,Maciej Glowiak,Joe Mambretti
Computer Science , 2015,
Abstract: The increase in real time ultra-high definition video services presents a challenging issue to current network infrastructures, because of its high bandwidth usage, which saturate network links. The required bandwidth is related to strict QoS requirements for digital media. There are systems in place currently to help reduce these problems, such as transcoders and application layer multicasting. However, these approaches are limited because they are usually implemented as static resources. In contrast, by using the OpenFlow based system presented in this paper, it is possible to provide a more effective approach using dynamic resources - by both optimally placing transcoders in the network, as well as by migrating them to different locations while the streaming is taking place. This migration mechanism provides a near seamless switchover with minimal interruption to the clients.
Adaptive Network Coding for Scheduling Real-time Traffic with Hard Deadlines  [PDF]
Lei Yang,Yalin Evren Sagduyu,Jason Hongjun Li,Junshan Zhang
Computer Science , 2012,
Abstract: We study adaptive network coding (NC) for scheduling real-time traffic over a single-hop wireless network. To meet the hard deadlines of real-time traffic, it is critical to strike a balance between maximizing the throughput and minimizing the risk that the entire block of coded packets may not be decodable by the deadline. Thus motivated, we explore adaptive NC, where the block size is adapted based on the remaining time to the deadline, by casting this sequential block size adaptation problem as a finite-horizon Markov decision process. One interesting finding is that the optimal block size and its corresponding action space monotonically decrease as the deadline approaches, and the optimal block size is bounded by the "greedy" block size. These unique structures make it possible to narrow down the search space of dynamic programming, building on which we develop a monotonicity-based backward induction algorithm (MBIA) that can solve for the optimal block size in polynomial time. Since channel erasure probabilities would be time-varying in a mobile network, we further develop a joint real-time scheduling and channel learning scheme with adaptive NC that can adapt to channel dynamics. We also generalize the analysis to multiple flows with hard deadlines and long-term delivery ratio constraints, devise a low-complexity online scheduling algorithm integrated with the MBIA, and then establish its asymptotical throughput-optimality. In addition to analysis and simulation results, we perform high fidelity wireless emulation tests with real radio transmissions to demonstrate the feasibility of the MBIA in finding the optimal block size in real time.
On Real Time Coding with Limited Lookahead  [PDF]
Himanshu Asnani,Tsachy Weissman
Mathematics , 2011,
Abstract: A real time coding system with lookahead consists of a memoryless source, a memoryless channel, an encoder, which encodes the source symbols sequentially with knowledge of future source symbols upto a fixed finite lookahead, d, with or without feedback of the past channel output symbols and a decoder, which sequentially constructs the source symbols using the channel output. The objective is to minimize the expected per-symbol distortion. For a fixed finite lookahead d>=1 we invoke the theory of controlled markov chains to obtain an average cost optimality equation (ACOE), the solution of which, denoted by D(d), is the minimum expected per-symbol distortion. With increasing d, D(d) bridges the gap between causal encoding, d=0, where symbol by symbol encoding-decoding is optimal and the infinite lookahead case, d=\infty, where Shannon Theoretic arguments show that separation is optimal. We extend the analysis to a system with finite state decoders, with or without noise-free feedback. For a Bernoulli source and binary symmetric channel, under hamming loss, we compute the optimal distortion for various source and channel parameters, and thus obtain computable bounds on D(d). We also identify regions of source and channel parameters where symbol by symbol encoding-decoding is suboptimal. Finally, we demonstrate the wide applicability of our approach by applying it in additional coding scenarios, such as the case where the sequential decoder can take cost constrained actions affecting the quality or availability of side information about the source.
Streaming-Codes for Multicast over Burst Erasure Channels  [PDF]
Ahmed Badr,Devin Lui,Ashish Khisti
Mathematics , 2013,
Abstract: We study the capacity limits of real-time streaming over burst-erasure channels. A stream of source packets must be sequentially encoded and the resulting channel packets must be transmitted over a two-receiver burst-erasure broadcast channel. The source packets must be sequentially reconstructed at each receiver with a possibly different reconstruction deadline. We study the associated capacity as a function of burst-lengths and delays at the two receivers. We establish that the operation of the system can be divided into two main regimes: a low-delay regime and a large-delay regime. We fully characterize the capacity in the large delay regime. The key to this characterization is an inherent slackness in the delay of one of the receivers. At every point in this regime we can reduce the delay of at-least one of the users until a certain critical value and thus it suffices to obtain code constructions for certain critical delays. We partially characterize the capacity in the low-delay regime. Our capacity results involve code constructions and converse techniques that appear to be novel. We also provide a rigorous information theoretic converse theorem in the point-to-point setting which was studied by Martinian in an earlier work.
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