Performance of Forward Error Correction in Transport Protocol at Intrasegment Level

In this paper we evaluate the performance of forward error correction (FEC) techniques in the presence of four wave mixing (FWM) and dispersion induced inter-symbol interference (ISI). A comparison with the detuning technique [1] is also presented, for the case of FWM penalty mitigation. FEC has been widely used in long haul submarine optical transmission systems, both repeated [2] and unrepeated [3] and its performance in the presence of polarization mode dispersion (PMD) has also been investigated [4,5]. The influence of non-linearities on optical systems that use FEC has recently been reported [6].

The performance of forward error correction (FEC) based on Reed-Solomon coding is analyzed experimentally for a burst-mode upstream channel within a passive optical network (PON) testbed. During this analysis, the upstream FEC performance is impaired by inducing correlated and localized errors within the burst through the injection of optical transients. These transients emulate the optical signal variation associated with the add-and-drop events of wavelength channels within a long-reach optical link based on a chain of erbium doped fiber amplifiers (EDFAs). The robustness of the FEC has been analyzed by measuring the post-FEC bit error rate (BER) as a function of the amplitude of the emulated transients and their delay with respect to the transmitted bursts. A margin of approximately 4dB is demonstrated for the transient amplitude before the FEC degradation. Furthermore, while the post-FEC BER is strongly degraded by the emulated transients, the pre-FEC BER stays below the FEC threshold, demonstrating the importance of measuring the post-FEC BER in order to correctly characterize the FEC performance in PON upstream channels affected by optical transients.

This paper describes an advanced decoding scheme for ATSC Systems. To improve the reception under dynamic propagation conditions an advanced VSB (A-VSB) was proposed. One of the new features in this transmission system is the improved performance of forward error correction. To use the full capacity of this redundancy information, an optimum decoding algorithm is essential. Most receivers use a soft-output Viterbi algorithm as convolutional decoding which offers the opportunity of adapting likelihood paths easily. Not only Reed-Solomon corrected packets might be fed back to the Viterbi decoder and can be used as soft-input information. Even transport stream sequences, which appear periodically, could be useful for an optimum convolutional decoder. This paper shows an example, how transport stream information might be used as additional forward error correction. Finally, results of the improvements are shown as bit-error probability in AWGN channel at the convolutional decoders output.

TCP as a reliable transport protocol is tuned to operate efficiently within traditional networks. Within wireless communication scenarios, TCP's congestion avoidance mechanism is falsely triggered by higher bit error probabilities of wireless links and disruptions due to handoffs within cellular communication networks. To separate wired and wireless subpaths of a hybrid communication path the indirect transport protocol model has been introduced. It subdivides the end-to-end transport connection into several subconnections, operating over wired subpaths and over wireless subpaths. Thus, the negative effects of the wireless subpath do not influence the transport connection operating over the wired part of the network. Interconnection of transport connections is realized by so called transport gateways. Due to the potential mobility of wireless attached stations, the migration of transport gateways to network nodes located nearby the current location of the mobile stations is needed. The migration approach presented by Bakre and Badrinath (1995) interrupts transport communication by up to 1.4 seconds. Our approach reduces communication pauses to 75 milliseconds. This is achieved by the so called delayed migration approach, that enables transport layer communication while state information of the transport protocol is transferred to the new gateway. To support the delayed migration approach modifications of the Mobile IP proposal are required. These modifications and performance measurements are presented in this paper.

Consistent data delivery in constrained networks such as wireless sensor networks (WSN), which has a significant role in Internet of Things (IoT), is vital due to the high probability of packet loss, not only because of the unreliable channel but also from link and buffer congestions. Thus, a reliable transport protocol should be used. Data caching at intermediate nodes is one way of improving the reliability performance of transport protocols in WSNs. However, it is not yet known up to what extent data caching alone can alleviate congestion in WSNs. This study investigates and analyses how a cache-based transport protocol, that has no congestion control mechanism and has a fixed transmission window, performs under congestion states. The transport protocol, called DTSN, is evaluated under link and buffer congestion states and the optimal performance is determined in terms of goodput, transmission cost and cache hit metrics. The results of our analysis reveal that intermediate caching was able to perform significantly during congestions in WSN. However, the amount of traffic being sent and intermediate node's cache memory should be taken into account. The optimum caching performance during link contentions is reached at higher transmission window and cache size values, while for buffer congestions, it is achieved at lower transmission window but higher cache size values, respectively. These values obtained can guarantee optimum usage of cache while ensuring congestion avoidance. In addition, the correlation that is established can serve as a basis for developing congestion window management for transport protocols in WSN that employ data caching.

Research into a transport protocol for high-speed networks is one of the most progressive areas in the field of computer communications. In high-speed networks using optical fiber as a transmission medium that gives the bandwidth from a hundred Mbits/s to Gbits/s, the roundtrip (propagation) delay and bandwidth product becomes large. It is well known that for an increase in this product, the throughput performance of the conventional error control procedures such as the go-back-N-based-ARQ scheme does degrade significantly. Therefore, a new protocol based on a selective-repeat ARQ scheme must be developed. The authors propose the stop-and-selective repeat (SSR) ARQ protocol and analyze the throughput performance of it. The throughput formula is exactly derived in terms of the roundtrip delay, the frame error rate, and the window size of from one to two roundtrip delays. The window size needed to offer the adequate throughput greater than 95% is shown.

In Japan, the next generation of Integrated Services Digital Broadcasting — Terrestrial (ISDB-T) system is being developed for transmission of large-volume content services such as ultra high definition television (UHDTV). The system uses low density parity check (LDPC) code as the forward error correction (FEC) to improve its robustness. In particular, the performance of the system's FEC should be evaluated not only in additive white Gaussian noise (AWGN) environments, but also in multipath environments. Moreover, it would be desirable for the new FEC to have better performance than the previous one even in multipath environments. Previous research evaluated the transmission performance of LDPC code by using computer simulations of the propagation model. The actual propagation characteristics were not evaluated. In this paper, we describe the transmission performance of convolutional coded orthogonal frequency division multiplexing (CC-OFDM) and LDPC coded OFDM (LDPC-OFDM) in computer simulations using the same OFDM frame structure and propagation channels as measured in an outdoor experiment conducted in an urban area. We show that LDPC code outperforms CC in multipath environments and how the transmission performance depends on the code rate. In addition, we analyze the correlation between transmission performance and propagation channels.

Digital Video Broadcasting-Terrestrial second generation (DVB-T2) systems are increasingly deployed for high quality terrestrial broadcasting. However, low Signal-to-Noise Ratio (SNR) conditions in urban and fringe coverage areas pose significant challenges, resulting in elevated Bit Error Rates (BER) and degraded service reliability. Forward Error Correction (FEC) techniques, particularly Low-Density Parity-Check (LDPC) and Bose Chaudhuri Hocquenghem (BCH) codes, play a crucial role in mitigating transmission errors and ensuring robust communication. This study investigates a systematic approach to optimizing LDPC and BCH FEC schemes to enhance transmission reliability and minimize BER under low-SNR conditions in DVB-T2 networks. Using simulation-based analyses, the study evaluates multiple LDPC code rates (e.g., 1/2, 2/3, 3/4) in combination with BCH codeword lengths and error-correcting capabilities. Results indicate that an optimized LDPC (64800, 32400) configuration combined with a BCH (255, 239) code significantly reduces BER, achieving approximately 10⁻⁵ at an SNR of 2 dB, compared to a BER of 10⁻³ using standard DVB-T2 FEC configurations. Additionally, iterative decoding algorithms for LDPC were fine-tuned to balance convergence speed and error correction performance, demonstrating a 25% reduction in decoding iterations without compromising reliability. The analysis also explores trade-offs between redundancy, throughput and latency, highlighting the practical considerations for real-world deployment. The findings underscore that strategic optimization of LDPC/BCH FEC parameters is critical for maintaining broadcast quality in low-SNR scenarios. By enhancing error resilience and reducing BER, the proposed approach improves signal integrity, supports consistent reception in fringe areas and maximizes spectrum efficiency. This work provides a foundation for adaptive FEC strategies in future DVB T2 systems, enabling broadcasters to deliver reliable high-definition content even under challenging channel conditions.

Performance and adaptability of the transport protocol are among the most critical parts of the TCP/IP protocol stack since these factors are directly related to the user's interaction and application experience. In this work we analyze specific problems of a reliable transport protocol's performance caused by rapidly changing topology in MANETs in addition to classical transport layer challenges. Then we propose the Wireless Transport Protocol (WTP) - a new lightweight, bandwidth-efficient, and reliable cross-layer transport protocol capable of adjusting its performance to suit highly dynamic environments, such as MANETs. Experimental results show that simplifying the protocol reduces its processing, while cross-layer optimization enables more precise adjustments to network and connection fluctuations, hence increasing transport protocol efficiency by up to 9% under different conditions as compared to standard TCP protocol.

The efficiency of special purpose radio-communications depends on transport layer protocols. Many factors have an impact on special purpose radio-communications efficiency, as well as TCP and UDP protocols. This paper presents the transport protocols throughput analysis in the HF and VHF radio-communications versus RTT and packet error rate probability. This paper considers a posibility of the application of TCP and UDP protocols in special purpose radiocommunications. The following factors will be considered: RTT, Round Trip Time, data packet size, window size and bit error rate. It can be seen from the analysis which transport protocol is the most suitable for special purpose radio-networks.

This chapter presents a distributed program that implements the reliable transport service (Chap. 20) over an LRD (loss, reorder, duplication) channel, as illustrated in Fig. 21.1. In computer networking, such a program is referred to as a transport protocol. Following this terminology, we refer to the system at each address as a “tp” system, where “tp” is short for transport protocol. We refer to the messages exchanged between tp users as “data blocks”, and reserve the term “messages” for the messages exchanged between the tp systems; the former would be sent inside the latter.

A lot of research on mobile ad hoc networks has focused on the development of routing and transport protocols. Specifically, this research is focused on the performance of TCP over wireless mobile ad hoc networks. Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly due to congestion. However, networks in wireless links perform poorly and suffer from losses due to reasons such as bit error and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms resulting in degraded end-to-end performance in wireless systems. The paper sets out the basis for research into the design and development of a new algorithm that is integrated with Mobile IPv6 in order to be implemented in future needs, and thus improve the performance of TCP in such networks. It is proposed to use OMNET++ as the simulation tool to compare the performance of TCP using different approaches with that of the general TCP.

This paper discusses the design and performance of a forward error correction (FEC) code classification system that is used to determine the size of an unknown codeword from a stream of bits. The classification system is a deep neural network that is trained and tested on half rate low density parity check (LDPC) codes. Tests were performed on streams of codewords using codes of up to 250 different sizes. The CNN based classifier performs very well.

BitTorrent is probably the most famous file-sharing protocol used in the Internet currently. It represents more than half of the P2P traffic. Various applications are using BitTorrent-like protocols to deliver the resource and implement techniques to perform a reliable data transmission. Forward Error Correction (FEC) is an efficient mechanism used for this goal. This paper proposes a performance evaluation of FEC implemented on BitTorrent protocol. A simulation framework has been developed to evaluate the improvement depending on many factors like the leeches/seeds number and capacities, the network nature (homogeneous or heterogeneous), the resource size, and the FEC redundancy ratio. The completion time metric shows that FEC is a method that accelerates the data access in some specific network configurations. On the contrary, this technique can also disrupt the system in some cases since it introduces an overhead.

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.