Type II ARQ schemes with multiple copy decoding for mobile communications

Error control coding improves reliability and efficiency of wireless communication systems. This research delves into the latest advancements in error control coding schemes for wireless communication systems, with a specific focus on their application within the domain of Low Power Wide Area Networks (LPWANs) such as Narrowband Internet of Things (NB-IoT) systems, LoRa energy-efficient hardware design, random access control systems, and wireless sensor networks. In the context of LPWANs, particularly NB-IoT systems, we investigate the adaptive coding and modulation (ACM) in NB-IoT systems, which adapts coding rates and modulation techniques to changing channel circumstances. Comprehensive analysis of the literature review shows that the proposed approaches are more energy efficient and less error-prone than fixed coding and modulation schemes. Using Cell Design Methodology (CDM), Single Error Correction (SEC) codes are optimized for energy efficiency. Power and energy efficiency have improved in standard CMOS and CDM logic systems. This impacts IoT devices, memory storage, and security systems. Next, Forward Error Correction (FEC) is examined for ALOHA-based random-access control systems wherein packet loss rate and spectral efficiency metrics are mathematically calculated. FEC is beneficial, especially when time and frequency synchronization diverge. Monte Carlo simulations studied in the literature shows that FEC improves satellite communication systems. This review establishes benefits of FEC, notably in satellite communications systems. In addition, Wireless Sensor Networks (WSNs) error control techniques and network reliability and performance are examined along with Convolutional, Turbo, Low density parity check codes(LDPC), and Polar error correction coding (ECC) methods. We learn about their benefits, drawbacks, and implementation challenges by assessing their efficacy, efficiency, and applicability in wireless sensor networks (WSN) applications. This study further highlights advancements of energy-efficient error control coding when, integrated into network designs alongwith the trade-offs between efficacy and intricacy to help improve wireless communication networks. Finally, this study summarizes current error control coding approaches for wireless communication systems. We provide insights into improving reliability, energy conservation, and efficacy by examining ACM, CDM, FEC, and error correction coding (ECC) across varied domains including LPWAN and IoT technologies with earnest hope to facilitate researchers, engineers, and professionals working in the field of error control coding in wireless communication.

Today, Forward Error Correcting (FEC) codes are mainly implemented in hardware, and many believe that their complexity prohibits their software implementation. This paper presents in detail how the performances of a software implementantion can be significantly improved. Different levels of optimization which are independent of the working environment are presented and discussed. The coding throughput of 100 Mbps on an UltraSparc 1 shows that FEC codes can be easily added to multimedia applications without requiring dedicated hardware support. As a case study, we use FEC codes to protect AAL5-PDUs from cell losses in ATM networks.

Forward error correction (FEC) coding, modulation formats like differential phase shift keying (DPSK) and adaptive electronic equalization to mitigate, e.g. polarization mode dispersion (PMD) are some of the methods applied in current direct-detection optical communication systems, which are limited to a spectral efficiency of 1 bit/s/Hz. Coherent single-carrier QPSK systems with polarization multiplexing (PolMUX) allows to attain an spectral efficiency of up to 4 bit/s/Hz. However, to guarantee a bit error rate (BER) <; 10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-15</sup> adaptive multiple-input multiple-output (MIMO) equalization as well as outer FEC coding has to be included which results in less than 4 bit/s/Hz in commercial 40 Gbit/s systems. From wireline and wireless systems it is well known that larger symbol alphabets in combination with FEC coding (denoted as coded modulation) and orthogonal frequency division multiplexing (OFDM) can significantly enhance the spectral efficiency. However, in order to exploit the channel capacity of the fiber the modulation itself needs to be capacity achieving. In this contribution we apply capacity achieving constellations in an optical OFDM system and jointly design mapping and low-density parity-check (LDPC) codes for an iterative decoding scheme, namely bit-interleaved coded modulation with iterative decoding (BICM-ID). The authors expect this coding scheme to be most appropriate for future optical transponders to utilize long haul communications close to channel capacity.

We propose an explicit construction of full-diversity space-time block codes, under the constraint of an error correction capability. Furthermore, these codes are constructed in order to be suitable for a serial concatenation with an outer linear forward error correcting (FEC) code. We apply the binary rank criterion, and we use the threaded layering technique and an inner linear FEC code to define a space-time error-correcting code. When serially concatenated with an outer linear FEC code, a product code can be built at the receiver, and adapted iterative receiver structures can be applied. An optimized hybrid structure mixing MMSE turbo equalization and turbo product code decoding is proposed. It yields reduced complexity and enhanced performance compared to previous existing structures.

Thanks to the large buffering time of off-line streaming applications, erasure resilient forward error correction (FEC) codes can improve the reliability of communication particularly well. However real-time streaming puts hard restrictions on the buffer size making FEC inefficient for combating long link failures on single path routes. Path diversity is orthogonal to buffering and permits real-time streaming to also benefit from application of FEC. We introduce a capillary routing algorithm offering layer by layer a wide range of multi-path routing topologies starting from a simple solution and evolving toward reliable routing patterns with highly developed path diversity. The friendliness of a multi-path routing pattern is rated by the overall amount of FEC redundancy required for combating the non-simultaneous failures of all links of the communication footprint. We rated the friendliness of a dozen of capillary routing layers, built on several hundreds of network samples obtained from a random walk mobile ad-hoc network (MANET). The overall requirement in redundant FEC codes decreases substantially as the spreading of the routing grows

This document generally describes how to use Forward Error Correction (FEC) codes to efficiently provide and/or augment reliability for data transport. primary focus of this document is the application of FEC codes to one-to-many reliable data transport using IP multicast. This document describes what information is needed to identify a specific FEC code, what information needs to be communicated out-of-band to use the FEC code, and what information is needed in data packets to identify the encoding symbols they carry. procedures for specifying FEC codes and registering them with the Internet Assigned Numbers Authority (IANA) are also described. This document should be read in conjunction with and uses the terminology of the companion document titled, The Use of Forward Error Correction (FEC) in Reliable Multicast.

Chapter 17 - Error-Control Coding Techniques and Applications

Due to the increasing demand of services and content, Quality of Service (QoS) has become an important aspect in digital broadcasting industry especially with the implementation of Digital Terrestrial Television (DTT) in Malaysia. Thus, the performance of DVB-T2 in Malaysia should be investigated because there are areas which are not in the coverage due to variety of terrain, where some areas received weak DVB-T2 signals. Apart from geological and topographic factors in Malaysia, the selection of DVB-T2 parameters is also an important factor in determining signal strength and coverage This research has proposed to study the performance analysis of DVB-T2 propagation for fixed reception in Malaysia's central region. The Field Strength Measurement (FSM) was set up to evaluate the QoS of DVB-T2 signal such as Field Strength (E), Modulation Error Ratio (MER), and Carrier to Noise Ratio (C/N) at specific test points across Malaysia's central region. The FSM was simulated at the designated test points using CHIRplus_BC software to evaluate the performance of Single Frequency Network (SFN) at three locations of transmitter. The DVB-T2 Modulation and Forward Error Correction (FEC) code rate parameters were simulated in order to evaluate the improvement in terms of coverage area. It can be concluded that the coverage of DTT is reduced with the increase in the distance of the measurement area to the transmitter. The finding also shows that the field strength value obtained from FSM and simulation was comparable. Furthermore, from the simulation results. it can be perceived that the coverage area across the central region of Malaysia could be improved by optimizing the modulation and FEC code rate. However, there is a trade-off between the signal coverage and channel capacity. The greater the signal coverage the less the channel capacity.

Rate-adaptive optical transmission techniques adjust information bit rate based on transmission distance and other factors affecting signal quality. These techniques enable increased bit rates over shorter links, while enabling transmission over longer links when regeneration is not available. They are likely to become more important with increasing network traffic and a continuing evolution toward optically switched mesh networks, which make signal quality more variable. We propose a rate-adaptive scheme using variable-rate forward error correction (FEC) codes and variable constellations with a fixed symbol rate, quantifying how achievable bit rates vary with distance. The scheme uses serially concatenated Reed-Solomon codes and an inner repetition code to vary the code rate, combined with singlecarrier polarization-multiplexed M-ary quadrature amplitude modulation (PM-M-QAM) with variable M and digital coherent detection. A rate adaptation algorithm uses the signal-to-noise ratio (SNR) or the FEC decoder input bit-error ratio (BER) estimated by a receiver to determine the FEC code rate and constellation size that maximizes the information bit rate while satisfying a target FEC decoder output BER and an SNR margin, yielding a peak rate of 200 Gbit/s in a nominal 50-GHz channel bandwidth. We simulate single-channel transmission through a long-haul fiber system incorporating numerous optical switches, evaluating the impact of fiber nonlinearity and bandwidth narrowing. With zero SNR margin, we achieve bit rates of 200/100/50 Gbit/s over distances of 650/2000/3000 km. Compared to an ideal coding scheme, the proposed scheme exhibits a performance gap ranging from about 6.4 dB at 650 km to 7.5 dB at 5000 km.

This paper investigates the transmission of H.264 /AVC video in the 3GPP Multimedia Broadcast/Multicast Streaming service (MBMS). Application-layer forward error correction (FEC) codes are used to combat transmission errors in the radio access network. In this FEC protection scheme, the media RTP stream is organized into source blocks spanning many RTP packets, over which FEC repair packets are generated. This paper proposes a novel method for unequal error protection that is applicable in MBMS. The method reduces the expected tune-in delay when a new user joins into a broadcast. It is based on four steps. First, temporally scalable H.264 /AVC streams are coded including reference and non-reference pictures or sub-sequences. Second, the constituent pictures of a group of pictures (GOP) are grouped according to their temporal scalability layer. Third, the interleaved packetization mode of RFC3984 is used to transmit the groups in ascending order of relevance for decoding. As an example, the non-reference pictures of a GOP are sent earlier than the reference pictures of the GOP. Fourth, each group is considered a source block for FEC coding and the strength of the FEC is selected according to its importance. Simulations show that the proposed method improves the quality of the received video stream and decreases the expected tune-in delay.

The growing demands on high-speed data transmission over long distances require optimized communication systems with bandwidth efficient modulation techniques like Quadrature Amplitude Modulation (QAM). However, the various design parameters of these QAM based systems conflict each other and the complex interdependence makes it difficult and time consuming to appropriately tune the parameters towards an optimal solution for a given set of constraints. Therefore, throughout this paper we analyze various conflicting design parameters of QAM and derive objective optimized trade-offs. More specifically, we study in detail the effects and interdependencies of parameters in the matched filter, the Forward Error Correction (FEC) code pair and the modulation order and span the design and solution space through experimental and numerical results. We discuss the observed effects and show how a selection of parameters can be concluded under various constraints. Furthermore, a pareto optimality search on our design space points out the best combinations of modulation order, filter and FEC design parameters for maximizing the performance under various Signal-to-Noise Ratio (SNR) conditions of the channel. For instance, if the channel is constrained by a SNR of 10dB and the tolerable Bit Error Rate (BER) is 10−3, then 16-QAM with a required 2/3 FEC yields 33.3% higher performance than 64-QAM with a required 1/3 FEC. Thus, the outcomes of this paper are a comprehensive elaboration on the cross-effects of conflicting design parameters and a set of optimized solutions under various constraints.

The area of digital communications has undergone a significant transformation in the last twenty years due mostly to the discovery of iterative forward error correction (FEC) codes (such as turbo codes) and iterative demodulation techniques (such as turbo equalization). Waveforms developed for the High Frequency (HF) band typically place an interleaver between the FEC scheme and the transmitted symbols. The purpose of this interleaver is to decorrelate the errors caused by slow fading multipath channels encountered on HF. If no interleaver is used, most standard EEC schemes become ineffective (e.g. convolutional codes). Although HF waveforms were not designed with the notion of iterative demodulation techniques in mind, it turns out that these techniques can be applied to single-carrier HF waveforms which utilize FEC and interleaving by iteratively exchanging soft information between the equalizer and the FEC decoder. This paper will investigate the benefits of applying iterative demodulation techniques to the 3200 and 9600 bit per second waveforms defined in STANAG 4539 as a function of the Doppler spread and multipath spread encountered on HF channels.

Modified Packet Combining (MPC) scheme is an error correction scheme, which can correct double or, higher bit errors. In some earlier work, a scheme known as Packet Reversed Packet Combining (PRPC) scheme that will correct error if occurs at same bit location of erroneous copies, was studied. Both PRPC and MPC scheme are believed to offer higher throughput and better error correction capability at high bit error rate (BER) and large packet size only. Exact analysis was performed to substantiate this claim of higher throughput and better error correction capability in high as well as in low bit error rate (BER) for large and medium packet size. To combat high bit error rate some researches proposed to use Error Correction Code (ECC) in place of Error Detection Code (EDC) in ARQ scheme. But no conclusive result is found to justify the idea in terms of throughput. The current investigation is an attempt to explore the basic gap. In this paper, we investigate the exact analysis of both PRPC and MPC in a Modified way over Conventional PRPC, and report the findings thereof. We propose a combined protocol of Modified PRPC and MPC, and find that it will offer higher throughput and better error correction capability at high and low BER with medium and large packet size as we have used MPC (Modified Packet Combining) scheme with PRPC (Packet Reverse Packet Combining) scheme in a Modified way by combining BEC (Backward Error Correction) with FEC (Forward Error Correction) Codes for Mobile Communication and Networks.

In this work we present a complete set of techniques devoted to offer robustness against frame losses in distributed speech recognition over packet-switched networks. The proposed scheme is composed of tree techniques, two of them are applied at the sender and the last one in the recognizer itself. On one hand, a media-specific forward error correction (FEC) technique is used to allow the recovery of information within the bursts. On the other hand, a recognizer-based technique well known by its remarkable ability to reduce the effects of long consecutive frame losses during recognition, the weighted Viterbi algorithm (WVA), is used to handle the additional information introduced by FEC codes. Moreover, a double stream strategy whereby interleaving can be applied along with FEC codes without any delay increase, is also applied. The application of interleaving allows to reduce the perceived burst length at the receiver, further improving the recognition performance. As a result, the proposed scheme can provide an acceptable performance even under extremely adverse channel conditions.

We consider a transmission of a delay-sensitive data stream (e.g., a video call) from a single source to a single destination. The reliability of this transmission may suffer from bursty packet losses - the predominant type of failures in today's Internet. An effective and well studied solution to this problem is to protect the data by a Forward Error Correction (FEC) code and send the FEC packets over multiple paths. In this paper, we show that the loss rate of such a classic multipath FEC scheme can often be significantly reduced, while keeping the total transmission rate and delay unchanged. Our key observation is that the propagation times on the available paths often significantly differ, usually by 10-100ms. We propose to exploit these differences by appropriate packet scheduling that we call `Spread'. We evaluate our solution with a precise, analytical formulation and trace-driven simulations. Our studies show that Spread substantially outperforms the state-of-the-art solutions. It typically achieves two- to five-fold improvement (reduction) in the effective loss rate. Or conversely, keeping the same level of effective loss rate, Spread significantly decreases the observed delays and helps fighting the delay jitter.