Sub-packet forward error correction mechanism for video streaming over wireless networks

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.

Streaming media on the Internet can be unreliable. Services such as audio-on-demand drastically increase the loads on networks; therefore, new, robust, and highly efficient coding algorithms are necessary. One method overlooked to date, which can work alongside existing audio compression schemes, is that which takes into account the semantics and natural repetition of music. Similarity detection within polyphonic audio has presented problematic challenges within the field of music information retrieval. One approach to deal with bursty errors is to use self-similarity to replace missing segments. Many existing systems exist based on packet loss and replacement on a network level, but none attempt repairs of large dropouts of 5 seconds or more. Music exhibits standard structures that can be used as a forward error correction (FEC) mechanism. FEC is an area that addresses the issue of packet loss with the onus of repair placed as much as possible on the listener's device. We have developed a server--client-based framework (SoFI) for automatic detection and replacement of large packet losses on wireless networks when receiving time-dependent streamed audio. Whenever dropouts occur, SoFI swaps audio presented to the listener between a live stream and previous sections of the audio stored locally. Objective and subjective evaluations of SoFI where subjects were presented with other simulated approaches to audio repair together with simulations of replacements including varying lengths of time in the repair give positive results.

In response to time-varying network loads, the proposed adaptive bandwidth-constrained media-specific forward error correction (FEC) mechanism for voice over IP (VoIP) attempts to maximise the perceived voice quality at the receiver by switching between coding algorithms while maintaining a constant source bit rate. Its inherent fixed bandwidth nature alloys the worries of network overloading caused by additional redundancies. Besides that, during low network traffic, the mechanism adopts a better speech coding scheme with a lower level of redundancy to achieve better voice quality and to reduce the overall latency. Typical voice traffic accumulated through Internet measurements is characterised by using the 2-state Markovian (Gilbert) loss model. To evaluate the overall performance of the adaptive mechanism, the cumulative distribution function (CDF) of perceived voice quality is adopted. Preliminary results clearly show that the proposed adaptive mechanism produces a consistent and enhanced voice reception subject to the primary-encoder limitation while maintaining a constant bandwidth.

Bandwidth aggregation is a key research issue in integrating heterogeneous wireless networks, since it can substantially increase the throughput and reliability for enhancing streaming video quality. However, the burst loss in the unreliable wireless channels is a severely challenging problem which significantly degrades the effectiveness of bandwidth aggregation. Previous studies attack such a critical problem by reactively increasing the Forward Error Correction (FEC) redundancy. Nonetheless, the increased FEC redundancy will impose higher loads on networks and lead to larger end-to-end delays of video streaming. In this paper, we propose a novel Loss Tolerant Bandwidth Aggregation approach (LTBA), which proactively leverages the channel diversity in heterogeneous wireless networks to overcome the burst loss. First, we allocate the FEC packets according to the ‘loss-free’ bandwidth of each wireless network to the multihomed client. Second, we deliberately insert intervals between the FEC packets' departures while still respecting the delay constraint. The proposed LTBA is able to reduce the consecutive packet loss under the burst loss assumption. We carry out the analysis for proving that the proposed LTBA outperforms the existing ‘back-to-back’ transmission schemes in minimizing end-to-end video distortion based on the Gilbert path model and continuous time Markov chain. We conduct the performance evaluation in the Exata and emulation results show that LTBA outperforms existing bandwidth aggregation schemes in handling the burst channel loss as well as in improving the video PSNR (Peak Signal-to-Noise Ratio).

Unmanned aerial vehicles (UAV) are rising in popularity together with video applications for both military and civilian use. Because of that, it is necessary to address a set of challenges related to the device movement, scarce resources as well as high error rates, making evident the need for an adaptive mechanism to strengthen video transmissions. Adaptive Forward Error Correction (FEC) techniques are known to be suitable to enhance the Quality of Experience (QoE) of video transmitted over error-prone wireless networks with high mobility. This book chapter proposes an adaptive video-aware FEC mechanism that uses motion vectors details to improve real-time UAV video transmissions, providing both higher user experience and better usage of resources. The benefits and drawbacks of the proposed mechanism along with the related work are analysed and put up for test through simulations and evaluated using QoE metrics.

This paper introduces Adaptive and Reliable Forward Error Correction (AR_FEC), a novel mechanism that is designed to ensure reliable and efficient real-time video delivery for Unmanned Aerial Vehicles (UAVs). UAVs communicate using wireless channels that are inherently unreliable. AR_FEC is an application-layer solution that is inspired by edge computing. It incorporates several elements, specifically, adaptive Forward Error Correction (FEC), dynamic video quality, and Unequal Error Protection (UEP). AR_FEC dynamically adjusts the number of redundancy frames per video Group of Picture (GoP) based on packet loss rate, recent average round-trip times, and cross-layer information such as bandwidth usage and network traffic load. Moreover, AR_FEC reacts to congestion via a dynamic video quality algorithm. By adaptively determining suitable redundancy for the video I-frames and adjusting video quality, significant improvements are achieved. Comprehensive simulations demonstrate that the proposed AR_FEC protocol offers excellent performance for the number of delivered frames, latency, throughput, and delivery of frames, within the delay constraints for real-time multimedia streaming from UAVs.

Optical packet-switching (OPS) networks remain an emerging technology, as exemplified by the feasibility challenges of optical RAM. Nevertheless, OPS networks possess significant potential advantages, including a high data transfer rate and low power consumption due to the elimination of optical–electrical–optical conversion, compared to conventional electronic packet-switching networks, flexibility in handling traffic fluctuations due to packet-switching principles, and low latency as they do not require overhead for communication path establishment. However, effective data recovery from non-negligible packet loss due to optical packet collisions remains a critical challenge in OPS networks. Packet-level forward error correction (FEC) is regarded as an effective data recovery method for lossy networks, such as OPS ones, because it does not rely on packet-by-packet feedback, which could be lost, and avoids the delays associated with feedback and retransmission. On the other hand, FEC requires the transmission of additional packets beyond the original data, increasing the computational load associated with encoding and decoding. In this paper, we propose a data transfer method based on erasure coding, which is useful not only in OPS networks but also in other network environments suffering from severe packet loss due to collisions or signal degradation, such as optical circuit switching networks and wireless networks. This method improves effective data transfer throughput by limiting the number of transmitted packets and reducing the computational overhead of the encoding and decoding processes. This improvement is achieved through the appropriate configuration of the distribution of XOR operation target packets that constitute XOR packets, thereby restricting the number of symbols encoded in each XOR operation and introducing cumulative feedback of which symbols were already recovered at the receiver. To validate the proposed method, we conducted not only simulation evaluations but also what we believe to be the world’s first packet transmission demonstration experiment using packet-level erasure coding over an actual OPS network testbed, taking actual computational overhead into account. The experimental results show that, in high transmission rate environments, the proposed method achieves up to a 188.7% increase in effective data transfer throughput compared to the benchmark LT-Codes-based packet-level FEC transmission method while reducing the processing time on the sender side by up to 15.6% and on the receiver side by up to 39.4%. In addition, the number of packets required for data decoding can be reduced by up to 63.3%.

This paper describes the design and implementation of a multimedia home gateway for threescreen television (3STV) service. The proposed in-home wireless network uses scalable video coding (SVC) and unequal error protection with Raptor forward error correction (FEC) for maximizing the quality of experience (QoE) over the variable-bandwidth, error-prone wireless network. The gateway incorporates (a) dynamic SVC layerswitching, which enables the server to perform selecting appropriate layers from SVC bitstreams, (b) adaptive Raptor FEC, which controls the overhead of Raptor FEC according to packet loss rate (PLR), (c) an efficient combination of (a) and (b), and (d) slice group-based selective streaming in the overall gateway architecture. The paper explains the home gateway architecture as well as experiments for performance evaluation as compared to that of traditional SVC streaming. In the experiments conducted, gains in video quality vary from 2 to 5dB in peak signal-to-noise ratio (PSNR), with corresponding subjective improvements. Overall reductions of bit rate at the input to the home gateway vary from 28% to 36% <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sup> .

Long Term Evolution (LTE) systems have been specified and designed to accommodate small, high performance, power-efficient, end-user devices. The evolved Multimedia Broadcast/Multicast Service (e-MBMS) feature is introduced by the 3rd Generation Partnership Project (3GPP) as a complement to the existing MBMS service in order to accommodate multicast groups that are interested in receiving the same data. MBMS service is provided by MBMS over a Single Frequency Network (MBSFN) and/or Point-To-Multipoint (PTM) transmission methods. One of the challenges of MBMS is the complete error recovery of the transmitted files, a matter of great importance since the distribution of binary data must result in 100% error-free download. To fulfill this tight requirement, Forward Error Correction (FEC) mechanism has been proposed by 3GPP. In this work, we investigate the reliable multicasting by introducing a transmission method that combines the advantages of MBSFN and PTM transmission methods. We compare several FEC-based file recovery methods and evaluate them against various network parameters in a realistic simulation environment. The comparison is based on a cost-oriented analysis of MBMS service that takes into account the transmission cost over all the interfaces and nodes of the LTE architecture. The simulation results are performed with the aid of a new simulation tool and show that the performance of the file repair schemes depend on the network configuration.

High frame-corruption is widely observed in Long Range Wide Area Networks (LoRaWAN) due to the coexistence with other networks in ISM bands and an Aloha-like MAC layer. LoRa's Forward Error Correction (FEC) mechanism is often insufficient to retrieve corrupted data. In fact, real-life measurements show that at least one-fourth of received transmissions are corrupted. When more frames are dropped, LoRa nodes usually switch over to higher spreading factors (SF), thus increasing transmission times and increasing the required energy. This paper introduces ReDCoS, a novel coding technique at the application layer that improves recovery of corrupted LoRa frames, thus reducing the overall transmission time and energy invested by LoRa nodes by several-fold. ReDCoS utilizes lightweight coding techniques to pre-encode the transmitted data. Therefore, the inbuilt Cyclic Redundancy Check (CRC) that follows is computed based on an already encoded data. At the receiver, we use both the CRC and the coded data to recover data from a corrupted frame beyond the built-in Error Correcting Code (ECC). We compare the performance of ReDCoS to (i) the standard FEC of vanilla-LoRaWAN, and to (ii) Reed Solomon (RS) coding applied as ECC to the data of LoRaWAN. The results indicated a 54x and 13.5x improvement of decoding ratio, respectively, when 20 data symbols were sent. Furthermore, we evaluated ReDCoS on-field using LoRa SX1261 transceivers showing that it outperformed RS-coding by factor of at least 2x (and up to 6x) in terms of the decoding ratio while consuming 38.5% less energy per correctly received transmission.

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.

To provide the broadcast services with Internet Protocol (IP)‐based networks, one needs to digitise the content, fill up the packets and send them. This chapter describes the basic concepts and techniques for video compression that are commonly used in delivering high‐definition IP television (HD IPTV) services. Multiprotocol label switching (MPLS) is a mechanism to direct packets on the shortest path using 20 bit 'labels' rather than the longer IP‐based network addresses. MPLS networks help IPTV services satisfy quality of services (QoS) requirements. The chapter describes how the various protocols and the MPLS network can compensate for the lack of QoS guarantee in IP networks. To provide scalable on‐demand media streaming, many other mechanisms such as periodic broadcast protocols patching and bandwidth skimming have been proposed. Among these, the chapter explores the details of an approach that combines the forward error correction (FEC) and retransmission mechanism.

The established success of wireless communications in terms of high data rates provision is undoubtedly owed to several combinations of different mechanisms. Such a combination is the well-known forward error correction (FEC) mechanism that combines automatic-repeat-request (ARQ) protocol and error correction capabilities. Furthermore, in the last decade, a cross-layer design provides the combination of different mechanisms across the layers and thus the performance gain is increased in wireless communications over fading channels. In this paper, we present a practical cross-layer design implementation that combines adaptive modulation and coding (AMC) and hybrid ARQ at the physical and data link layer by using different error correction schemes. Our implementation is based on a cross-layer design of AMC and HARQ. We perform simulations with different channel coding and modulation schemes that are implemented in broadband wireless access networks as IEEE 802.16 and 3GPP HSPA and thus the results obtained show the performance achieved by this practical cross-layer implementation in terms of spectral efficiency in two main broadband wireless access networks available into the market.

Congestion in ATM communications is a significant issue as it can have a dramatic effect on critical or real-time data. Forward error correction (FEC) can be used to decrease or eliminate this effect. Conventional FEC has a uniform error correction capability, which can result in poor bandwidth utilization. Adaptive FEC is more suitable because it allows a different quality of service for each data type. In this paper a versatile adaptive FEC technique is proposed. It is shown that this technique can decrease the effect of congestion and increase the channel utilization.

Multi-hop wireless networks (MWNs) provide an important infrastructure for ubiquitous multimedia content access. However, the accumulated packet losses greatly decrease the quality of multimedia services, particularly video streaming services. In order to overcome packet losses in MWNs, in this paper we propose a proportional integral derivative control-based forward error correction (PID-FEC) mechanism to improve the quality of video streaming services. Our proposed method adopts the cross-layer approach, and leverages the functionalities of different network layers. The automatic repeat request (ARQ) on the media access control (MAC) layer is used as an indicator of packet losses. With the packet loss information, the redundancy rates are adaptively regulated based on the PID control algorithm. Experimental results, presented herein, show that the PID-FEC achieves a better quality of video streaming, as well as a higher FEC efficiency, as compared with conventional FEC schemes, over a variety of network conditions.