Quality Differentiation with Source Shaping and Forward Error Correction

Forward Error Correction (FEC) is a commonly adopted mechanism to mitigate packet loss/bit error during real-time communication. An adaptive, Fuzzy based FEC algorithm to provide a robust video quality metric for multimedia transmission over wireless networks has been proposed to optimize the redundancy of the generated code words from a Reed-Solomon encoder and to save the bandwidth of the network channel. The scheme is based on probability estimations derived from the data loss rates related to the recovery mechanism at the client end. By applying the adaptive FEC, the server uses the reports to predict the next network loss rate using a curve-fitting technique to generate the optimized number of redundant packets to meet specific residual error rates at the client end. Simulation results in the cellular system show that the video quality is massively adapted to the optimized FEC codes based on the probability of packet loss and packet correlation in a wireless environment.

We devise a scheme which can provide reliable transport services in sensor networks and give an algorithm which minimizes the energy use of our scheme. We use a distributed sink where information arrives at the sink via multiple proxy nodes, called in this paper. The sender node uses forward error correction (FEC) erasure coding to encode each packet into multiple fragments and transmits the fragments to each of the prongs over a path which is disjoint from the paths to the other prongs. The erasure coding allows the sink to reconstruct the original packet even if some of the fragments are lost. We use a cross-layer design where higher network layers use information about packet loss and energy consumption to distribute the load in the network. We show that the source can distribute the load so that energy consumption is minimized. The optimization takes fault tolerance into account with a bound on the probability of packet loss. The extra fragments increase both the reliability and the energy in the network. However, we show with simulations that it is possible for the sensor to decrease energy use in the network by using the diversity available with multiple network paths.

In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.

Real-time audio-visual communication requires a packet-loss recovery mechanism due to inevitable packet loss characteristics of the transmission channel such as the Internet. Especially for interactive applications e.g. real-time audio-visual communication, the delay for the recovery and enhancement technology must be minimized. Among the low-delay packet-loss recovery techniques, Multiple Description coding (MDC) and Forward Error Correction (FEC) are widely used for their robustness and efficiency. This paper proposes three new MDC-FEC models to minimize the overall distortion subject to specific source coding rates and channel packet loss probabilities. The proposed solution would enable adaptive selection of an optimal packet-loss recovery mechanism based on real-time feedback of channel conditions. The results indicate that this approach leads to an optimal model for joint source-channel coding based on a combination of MDC and FEC.

The subject of research in the article is the processes of fast rerouting with the protection of the Quality of Service level in infocommunication networks. The aim of the work is the development of a mathematical model of Fast ReRouting with the protection of the Quality of Service level by the bandwidth and probability of packet loss for data traffic. The following tasks are solved in the article: development and research of a fast rerouting flow-based model with the protection of the Quality of Service level of data traffic in the infocommunication network. The following methods are used: the graph theory, the teletraffic theory, the queuing theory, and mathematical programming methods. The following results were obtained: the fast rerouting flow-based model was developed and investigated, which, due to the introduced protection conditions on indicators of bandwidth and packet loss probability, allows providing the Quality of Service along both the primary and backup multipath. Conclusions: Within the framework of the proposed flow-based model, the fast rerouting technological task was formulated as an optimization problem with the constraints of the conditions of implementation of the multipath routing strategy, conditions of flow conservation, and conditions of protection of the link, node, and level of Quality of Service in terms of bandwidth and packet loss probability. The application of this solution contributes to the optimal use of the available network resource while providing the specified level of Quality of Service in terms of bandwidth and probability of packet loss along both the primary and backup routes in the case of a failure of the infocommunication network elements. The proposed flow-based model can be used as the basis of the algorithmic software of existing routers and/or controllers of Software-Defined Networks, which are responsible for the calculation of the primary and backup paths in the fast rerouting of data traffic sensitive to such Quality of Service indicators as bandwidth and packet loss probability.

This tutorial paper provides a foundation for integrating probabilistic shaping (PS) and forward error correction (FEC). A layered PS-FEC architecture consisting of a PS encoder and an FEC encoder is introduced, of which probabilistic amplitude shaping is a practical instance. Achievable PS encoding rates and achievable FEC decoding rates are derived using information-theoretic arguments. The developed tools are applied to the design and performance assessment of optical transponders based on measurements from optical transmission experiments.

With the explosive growth of video applications over the packet switched networks, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. In our previous work, we proposed a framework with a receiver driven protocol to coordinate simultaneous video streaming from multiple senders to a single receiver in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. The receiver-driven protocol employs two algorithms: the rate allocation and packet partition. The rate allocation algorithm determines the sending rate for each sender while the packet partition algorithm ensures no sender sends the same packets, and at the same time, minimizes the probability of late packets. We extend the rate allocation scheme to be used with forward error correction (FEC) in order to minimize the probability of packet loss in a bursty loss environment such as one due to network congestion. Using both simulations and actual Internet experiments, we demonstrate the effectiveness of our rate allocation scheme in reducing packet loss, and hence, achieving higher visual quality for the streamed video.

Online video transmissions over wireless networks are rising in popularity and have already become part of our daily life. In the meantime, it is necessary to address a number of challenges ranging from the scarce resources, time-varying, and high error rates, to the fluctuating bandwidth, unveiling the need for an adaptive mechanism to ensure a good video transmission. Adaptive Forward Error Correction (FEC) techniques with Quality of Experience (QoE) assurance are appropriate to deliver QoE-aware video data to wireless users in dynamic and high error rates networks. This paper proposes an adaptive Video-aware FEC and Fuzzy Logic-based mechanism to shield realtime video transmissions against packet loss in wireless networks, improving both user experience and the usage of resources. The benefits and drawbacks of the proposed mechanism compared with exiting work are demonstrated through simulations and assessed with QoE metrics.

The growth in the power and connectivity of the Internet has sparked an even larger growth in streaming media. The sheer number of possible users and applications at any point in time raises the probability of streaming multimedia flows encountering congestion. To overcome short-term congestion and avoid long-term congestion collapse, there is a growing consensus that Internet applications must be TCP-Friendly, with proposed approaches to detect and punish non-TCP friendly flows. Unlike TCP, new TCP-friendly streaming media protocols refrain from retransmissions to avoid delay and jitter, but they are susceptible to quality degradation from packet loss. While multimedia applications can tolerate some data loss, excessive packet loss during congestion yields unacceptable media quality. Since video encoding involves interframe dependencies to achieve high compression rates, the random dropping of packets by routers can seriously degrade video quality. For example, as little as 3% MPEG packet loss can cause 30% of the frames to be undecodable. Streaming media flows often utilize lower latency repair approaches, such as Forward Error Correction (FEC), in conjunction with TCP-Friendly protocols to deliver streaming applications over the Internet. However, FEC requires redundant repair data to be added to the original video stream. Current approaches use either apriori, static FEC choices or adapt FEC to perceived packet loss on the network without regard to TCP-Friendly data rate constraints. When a streaming video operates within TCP-Friendly bitrate limits, adding FEC will reduce the effective transmission rate of the original video content.

An expanding proportion of voice traffic is being carried by packet networks. Speech quality can be impaired in qualitatively new ways in packet networks when packets are lost or the spacing between them is distorted. Three parameters that characterize the performance of packet networks were examined for their relative impact on speech quality as judged by human observers: network delay or latency, packet loss, and packet delay variation or “jitter.” We manipulated these variables via a network emulator made available by NIST. This report summarizes five laboratory experiments that examined the variables in a variety of experimental procedures for presenting and judging speech. The experiments agreed in showing that the relative importance of the variables for affecting speech quality was, in decreasing order: packet loss, jitter, delay. The effect on speech quality of 200 ms of network delay was shown to be equivalent to the effect of one percentage point of packet loss. Many consumers also traded off some speech quality for a free, added feature, unified messaging.

In future broadband networks based on asynchronous transfer mode (ATM) transmission and switching technology, the dominant kind of impairment is expected to be packet (ATM cell) losses. The use of forward error correction (FEC) codes has been proposed as a way to overcome these losses. However, the maximum length of a lost packet string that a packet loss recovery scheme can be designed to recover highly depends upon the temporal behavior between the adjacent packet loss bursts. Thus, a realistic model for describing the packet loss process is needed. The Gilbert model proposed for such purpose turns out to be an unrealistic model, since it fails to consider some important characteristics during the packet loss process. A more realistic model is advanced in this paper.

In this paper, we present the design of a new packet loss concealment (PLC) algorithm based on Dynamic Forward Error Correction (FEC) to improve speech quality degradation caused by packet losses. It is to be implemented for the standard ITU-T AMR-WD G.722.2 codec. The performance proposed scheme is compared with the original method embedded in the standard ITU-T G.772.2. To evaluate its performance, the perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD) tests under a variety of packet loss rates are used and the obtained results prove that the proposed PLC based Dynamic FEC is better than the concealment method embedded in the ITU-T G.722.2 for speakers (female and male). The performance measures confirm that our PLC method based Dynamic FEC is superior than the one embedded in the standard ITU-T G.722.2.

We report the implementation experience and experimental evaluation of a staggered adaptive forward error correction (FEC) system for video multicast over wireless LANs. In the system, the parity packets generated by a cross-packet FEC code are transmitted at a time delay from the original video packets, i.e. staggercasting video stream and FEC stream in different multicast groups. The delay provides temporal diversity to improve the robustness of video multicast, especially to enable the clients to correct burst packet loss using FEC and to achieve seamless handoff. A wireless client dynamically joins the FEC multicast groups based upon its channel conditions and handoff events. We have implemented the system including the streaming server and client proxy. A novel software architecture is designed to integrate the FEC functionality in the clients without requirement for changing the existing video player software. We conduct extensive experiments to investigate the impact of FEC overhead and the delay between the video stream and FEC stream to the video quality under different interference levels and mobile handoff durations. The efficacy of staggered adaptive FEC system on improving video multicast quality is demonstrated in real system implementation.

This paper presents a comparison of end-to-end recovery and control methods for the ITU-T G.729 CS-ACELP codec operating at 8 Kbps. We have tested Media independent forward error correction (FEC) parity 3, Media independent FEC parity 4 and Media specific FEC combined with queue management methods namely Drop Tail, Predictive Loss Pattern (PLoP) and Random Early Detection (RED). The performance measures using perceptual evaluation of speech quality (PESQ) tests under various packet loss conditions prove that Media independent FEC parity 3 combined with RED algorithm give better results.

Real-time audio over a best-effort network, such as the Internet, frequently suffers from packet loss. To mitigate the impact of such packet loss, several research efforts and implementation studies advocate the use of forward error correction (FEC) coding. Although these prior works have pioneered promising and novel applications of to Internet audio, they do not definitively demonstrate the advantages of because they do not evaluate aggregate performance that results from multiplexing many like flows. We build on previous landmark works with a systematic study of for packet audio that characterizes the aggregate performance across all audio sources in the network. We refine the novel but ad hoc coding techniques proposed by Hardman, Sasse, Handley and Watson (see Proc. INET, 1995) into a formal framework that we call processing-based FEC (SFEC) and use our framework to more rigorously evaluate the relative merits of this approach. Through extensive simulation, we evaluate the scalability of SFEC for packet audio-i.e., the ability for a coding algorithm to improve aggregate performance when used by all sources in the network-and find that optimal signal quality is achieved when sources react to network congestion not by blindly adding FEC, but rather by adding in a controlled fashion that simultaneously constrains the source-coding rate. As a result, packet loss is mitigated without introducing more congestion, thus admitting a more scalable and effective approach than successively adding redundancy to a constant bit-rate source. While this result may seem intuitive, it has not been previously suggested in the context of Internet audio, and until now, has not been systematically studied.