Source Packets Research Articles

A simple coded ARQ for satellite broadcasting

We introduce a novel packet retransmission technique which improves the efficiency of automatic retransmission query (ARQ) protocols in the context of satellite broadcast/multicast systems. The proposed coded ARQ technique, similarly to fountain coding, performs transmission of redundant packets, which are made by linear combinations of the packets composing the source block. Differently from fountain codes, the packets for the linear combinations are selected on the basis of the retransmission requests coming from the user terminals. The selection is performed in a way that, at the terminals, the source packets can be recovered iteratively by means of simple back-substitutions. This work aims at providing a simple and efficient alternative to reliable multicast protocols based on erasure correction coding techniques.

Random linear network coding with ladder-shaped global coding matrix for robust video transmission

Robust video multicast in erasure networks using network coding (NC) to reduce the impact of packet loss is studied in this paper. In our proposed solution, random linear network coding (RLNC) is adopted at intermediate nodes of the network. RLNC linearly combines a group of packets by randomly selecting weighting coefficients on a finite field, and the loss of an RLNC-coded packet is equivalent to the loss of one constraint in a linear system of equations required for RLNC decoding. Unless the global coding coefficient matrix, or simply called the global coding matrix (GCM), is of full rank, a receive node cannot reconstruct all source packets. To address this rank deficiency problem, we propose to construct a special-structured GCM, called the ladder-shaped GCM (LGCM), for layered H.264/SVC (scalable video coding) video multicast. The ladder shape of the sparse coding matrix is maintained throughout the RLNC process to achieve two objectives: (1) to enable partial decoding of a block; and (2) to provide unequal erasure protection for H.264/SVC priority layers. Furthermore, quality degradation is minimized by optimizing the amount of redundancy assigned to each layer, and graceful quality degradation is achieved by error concealment (EC). Simulation results are given to demonstrate the superior performance of the proposed RLNC–LGCM scheme over the traditional RLNC with a generalGCM.

Secure Network Coding against Wiretapping and Byzantine Attacks

In wireless networks, an attacker can tune a receiver and tap the communication between two nodes. Whether or not some meaningful information is obtained by tapping a wireless connection depends on the transmission scheme. In this paper, we design some secure network coding by combining information-theoretic approaches with cryptographic approaches. It ensures that the wiretapper cannot get any meaningful information no matter how many channels are wiretapped. In addition, if each source packet is augmented with a hash symbol which is computed from a simple nonlinear polynomial function of the data symbols, then the probability of detecting the modification is very high.

An Adaptive Packet and Block Length Forward Error Correction for Video Streaming Over Wireless Networks

Video streaming over wireless networks is a challenging task due to its high error rate. Forward error correction (FEC) is a popular mechanism to recover lost packets for video streaming. Conventional FEC mechanisms use a whole redundant packet to recover the error source packet, when the packet error occurs with only a few bit errors inside. In this paper, we propose an Adaptive packet and block length FEC (APB-FEC) control mechanism. In order to overcome the high bit error rate, a small packet length reduces the packet error rate and a large FEC block length will enhance the recovery performance. Our proposed APB-FEC can obtain better recovery performance than conventional FEC mechanisms. Hence, APB-FEC can also reduce retransmission overhead. Using extensive emulations, we validate the efficiency of APB-FEC mechanism for video streaming over wireless networks.

Decentralized Coding Algorithms for Distributed Storage in Wireless Sensor Networks

We consider large-scale wireless sensor networks with n nodes, out of which k are in possession, (e.g., have sensed or collected in some other way) k information packets. In the scenarios in which network nodes are vulnerable because of, for example, limited energy or a hostile environment, it is desirable to disseminate the acquired information throughout the network so that each of the n nodes stores one (possibly coded) packet so that the original k source packets can be recovered, locally and in a computationally simple way from any k(1 + ¿) nodes for some small ¿ > 0. We develop decentralized Fountain codes based algorithms to solve this problem. Unlike all previously developed schemes, our algorithms are truly distributed, that is, nodes do not know n, k or connectivity in the network, except in their own neighborhoods, and they do not maintain any routing tables.

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

Traditional Forward Error Correction (FEC) mechanisms can be divided into Packet level FEC (PFEC) mechanisms and Byte level FEC (BFEC) mechanisms. The PFEC mechanism of recovering from errors in a source packet requires an entire FEC redundant packet even though the error involves a few bit errors. The recovery capability of the BFEC mechanism is only half of the FEC redundancy. Accordingly, an adaptive Sub-Packet FEC (SPFEC) mechanism is proposed in this paper to improve the quality of video streaming data over wireless networks, simultaneously enhancing the recovery performance and reducing the end-to-end delay jitter. The SPFEC mechanism divides a packet into n sub-packets by means of the concept of a virtual packet. The SPFEC mechanism uses a checksum in each sub-packet to identify the position of the error sub-packet. Simulation experiments show the adaptive SPFEC mechanism achieves high recovery performance and low end-to-end delay jitter. The SPFEC mechanism outperforms traditional FEC mechanism in terms of packet loss rate and video Peak Signal-to-Noise Ratio (PSNR). SPFEC offers an alternative for improved efficiency video streaming that will be of interest to the designers of the next generation environments.

Robust video multicast with joint network coding and video interleaving

In this work, we propose a cross-layer solution to robust video multicast in erasure networks based on random linear network coding (RLNC) in the network layer and video interleaving (VI) in the application layer, and call it the joint RLNC-VI scheme. In the RLNC implementation, we partition one video coding unit (VCU) into several priority levels using scalable properties of H.264/SVC video. Packets from the same priority level of several VCUs form one RLNC generation, and unequal protection is applied to different generations. RLNC provides redundancy for video packets in the network layer and has proved to be useful in a multicast environment. Then, we propose a new packet-level interleaving scheme, called the RLNC-facilitated interleaving scheme, where each received packet corresponds to a new constraint on source packets. As a result, it can facilitate the RLNC decoding at the destination node. Furthermore, we study the problem of optimal interleaving design, which selects the optimal interleaving degree and the optimal redundancy of each generation. The tradeoff between delay and received video quality due to the choice of different VCUs is also examined. It is shown by simulation results that the proposed RLNC-VI scheme outperforms the pure RLNC method for robust video multicast in erasure networks. This can be explained by two reasons. First, the VI scheme distributes the impact of the loss (or erasure) of one VCU into partial data loss over multiple neighboring VCUs. Second, the original video content can be easily recovered with spatial/temporal error concealment (EC) in the joint RLNC-VI scheme.

Doped fountain coding for minimum delay data collection in circular networks

This paper studies decentralized, Fountain and network-coding based strategies for facilitating data collection in circular wireless sensor networks, which rely on the stochastic diversity of data storage. The goal is to allow for a reduced delay collection by a data collector who accesses the network at a random position and random time. Data dissemination is performed by a set of relays which form a circular route to exchange source packets. The storage nodes within the transmission range of the route's relays linearly combine and store overheard relay transmissions using random decentralized strategies. An intelligent data collector first collects a minimum set of coded packets from a subset of storage nodes in its proximity, which might be sufficient for recovering the original packets and, by using a message-passing decoder, attempts recovering all original source packets from this set. Whenever the decoder stalls, the source packet which restarts decoding is polled/doped from its original source node. The random-walk-based analysis of the decoding/doping process furnishes the collection delay analysis with a prediction on the number of required doped packets. The number of doped packets can be surprisingly small when employed with an Ideal Soliton code degree distribution and, hence, the doping strategy may have the least collection delay when the density of source nodes is sufficiently large. Furthermore, we demonstrate that network coding makes dissemination more efficient at the expense of a larger collection delay. Not surprisingly, a circular network allows for a significantly more (analytically and otherwise) tractable strategies relative to a network whose model is a random geometric graph.

Channel coding and decoding in a relay system operated with physical-layer network coding

This paper investigates link-by-link channel-coded PNC (physical layer network coding), in which a critical process at the relay is to transform the superimposed channel-coded packets received from the two end nodes (plus noise), Y <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">3</sub> = X <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> + X <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> +W <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">3</sub> , to the network-coded combination of the source packets, S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> oplus S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> . This is in contrast to the traditional multiple-access problem, in which the goal is to obtain both S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> and S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> explicitly at the relay node. Trying to obtain S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> and S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> explicitly is an overkill if we are only interested in S1oplusS <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> . In this paper, we refer to the transformation Y <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">3</sub> rarr S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> oplus S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> as the channel-decoding- network-coding process (CNC) in that it involves both channel decoding and network coding operations. This paper shows that if we adopt the repeat accumulate (RA) channel code at the two end nodes, then there is a compatible decoder at the relay that can perform the transformation Y <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">3</sub> rarr S <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sub> oplusS <sub xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sub> efficiently. Specifically, we redesign the belief propagation decoding algorithm of the RA code for traditional point-to-point channel to suit the need of the PNC multiple-access channel. Simulation results show that our new scheme outperforms the previously proposed schemes significantly in terms of BER without added complexity.

Training overhead for decoding random linear network codes in wireless networks

We consider multicast communications from a single source to multiple destinations through a wireless network with unreliable links. Random linear network coding achieves the min-cut flow capacity; however, additional overhead is needed for end-to-end error protection and to communicate the network coding matrix to each destination. We present a joint coding and training scheme in which training bits are appended to each source packet, and the channel code is applied across both the training and data. This scheme allows each destination to decode jointly the network coding matrix along with the data without knowledge of the network topology. It also balances the reliability of communicating the network coding matrices with the reliability of data detection. The throughput for this scheme, accounting for overhead, is characterized as a function of the packet size, channel properties (error and erasure statistics), number of independent messages, and field size. We also compare the performance with that obtained by individual channel coding of training and data. Numerical results are presented for a grid network that illustrate the reduction in throughput due to overhead.

Flexible forward error correction codes with application to partial media data recovery

Conventionally, linear block codes designed for packet erasure correction are targeted to recover all the lost source packets per block, when the fraction of lost data is smaller than the redundancy overhead. However, these codes fail to recover any lost packets, if the number of erasures just exceeds the limit for full recovery capability, while it can still be beneficial to recover part of the symbols. In addition, common linear block codes are not well suited for unequal error protection, since different block codes with different rates must be allocated for each priority class separately. These two problems motivate the design of more flexible forward error correction (FEC) codes for media streaming applications. We first review the performance of short and long linear block codes. Long block codes generally offer better error correction capabilities, but at the price of higher complexity and larger coding delay. Short block codes can be more appropriate in media streaming applications that require smooth performance degradation when the channel loss rate increases. We study a new class of linear block codes using sparse generator matrices that permit to optimize the performance of short block codes for partial recovery of the lost packets. In addition, the proposed codes are extended to the design of unequal erasure protection solutions. Simulations of practical video streaming scenarios demonstrate that the flexible sparse codes offer a promising solution with interesting error correction capabilities and small variance in the residual loss rate. They typically represent an effective trade-off between short block codes with limited flexibility, and long block codes with delay penalties.

Automatic Request for Cooperation (ARC) and Relay Selection for Wireless Networks

Recently, there has been growing interest in the design of wireless cooperative protocol to achieve higher diversity-multiplexing tradeoff among single antenna devices. We propose an automatic request for cooperation (ARC) scheme for wireless networks which can achieve higher order diversity by selecting the best relay. In this scheme, a source transmits a data packet towards a destination and a group of relays. The destination tries to decode the information from the source and if the detection is correct the process will stop. Otherwise, the destination transmits an ARC towards the relays. We utilize this ARC signal for selecting the best relay from the set of relays that have successfully decoded the source packet. The selected relay generates and transmits redundant information for the source packet. The destination combines the two packets received from the source and the best relay to improve the reliability of the packet. We analyze the packet error rate, spectral efficiency and diversity-multiplexing tradeoff of our proposal and compare them with some existing protocols. Analysis shows that our proposal can achieve higher diversity multiplexing tradeoff than conventional cooperative protocols.

A family of concatenated network codes for improved performance with generations

Random network coding can be viewed as a single block code applied to all source packets. To manage the concomitant high coding complexity, source packets can be partitioned into generations; block coding is then performed on each set. To reach a better performance-complexity tradeoff, we propose a novel concatenated network code which mixes generations while retaining the desirable properties of generation-based coding. Focusing on the code's erasure performance, we show that the probability of successfully decoding a generation on erasure channels can increase substantially for any erasure rate. Using both analysis (for small networks) and simulations (for larger networks), we show how the code's parameters can be tuned to extract best performance. As a result, the probability of failing to decode a generation is reduced by nearly one order of magnitude.

Byzantine Modification Detection in Multicast Networks With Random Network Coding

<para xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> An information-theoretic approach for detecting Byzantine or adversarial modifications in networks employing random linear network coding is described. Each exogenous source packet is augmented with a flexible number of hash symbols that are obtained as a polynomial function of the data symbols. This approach depends only on the adversary not knowing the random coding coefficients of all other packets received by the sink nodes when designing its adversarial packets. We show how the detection probability varies with the overhead (ratio of hash to data symbols), coding field size, and the amount of information unknown to the adversary about the random code. </para>

Towards universal power efficient scheduling in Gaussian channels

In this paper, we propose a framework for designing power efficient schedulers for transmitting bursty traffic sources over Gaussian wireless channels that provides deterministic and statistical guarantees on absolute delays experienced by the source packets. The proposed schedulers compute the transmission rate and power using temporal water-filling techniques without any knowledge of the arrival traffic statistics. The schedulers reduce the average transmission power substantially (55% in some scenarios) for small increases in delay. The framework allows us to design schedulers that artfully tradeoff the performance with the complexity of computing the schedulers. We also introduce an iterative process to compute a lower bound on the transmit power of any scheduler that provides absolute delay guarantees. The utility of having accurate traffic predictors is demonstrated; specifically, we show that a perfect one step predictor achieves near optimal performances for small delay bounds. The proposed schedulers and iterative method of computing the lower bound are also shown to provide statistical guarantees on packet delays.

Efficient allocation of packet-level forward error correction in video streaming over the Internet

Forward error correction (FEC) improves the quality of compressed video transmitted through a lossy network for real-time applications such as video streaming. The FEC techniques are generally applied on source video packets at the frame level. In this paper, we propose a technique where the FEC is applied on source packets at the group-of-pictures (GoP) level assuming an MPEG-like compression scheme. We derive analytically an estimate of the average playable frame rate for a given packet loss probability. Our analysis over a range of network conditions indicates that in most practical network conditions, the proposed technique provides a larger playable frame rate compared to the frame-level FEC technique. This analysis results are also validated by video streaming simulations conducted on the NS-2 network simulator.

Progressive Transmission of Images Over Fading Channels Using Rate-Compatible LDPC Codes

In this paper, we propose a combined source/channel coding scheme for transmission of images over fading channels. The proposed scheme employs rate-compatible low-density parity-check codes along with embedded image coders such as JPEG2000 and set partitioning in hierarchical trees (SPIHT). The assignment of channel coding rates to source packets is performed by a fast trellis-based algorithm. We examine the performance of the proposed scheme over correlated and uncorrelated Rayleigh flat-fading channels with and without side information. Simulation results for the expected peak signal-to-noise ratio of reconstructed images, which are within 1 dB of the capacity upper bound over a wide range of channel signal-to-noise ratios, show considerable improvement compared to existing results under similar conditions. We also study the sensitivity of the proposed scheme in the presence of channel estimation error at the transmitter and demonstrate that under most conditions our scheme is more robust compared to existing schemes.

Low complexity adaptive error control for receiver‐driven layered video multicast

A novel error control algorithm for robust video transmission is presented in this paper. In the algorithm, both the source packets and parity packets are delivered in layers. Each receiver, in accordance with its available bandwidth and packet loss rate, employs the optimal policy for subscribing source layers and parity layers minimizing the average distortion of the reconstructed frames. The algorithm is able to perform fast policy adaptation for location‐ and time‐varying network environments. The implementation of the fast adaptive algorithm is based on a novel table look‐up process, in which the tables are constructed using a dynamic programming technique. Our experiments show that the algorithm is an effective alternative for video broadcast applications where the robust transmissions over the heterogeneous network environment are desired.

Wavelet packet based analysis of sound fields in rooms using coincident microphone arrays

This paper presents a passive analysis method for determining the spatio-temporal characteristics of sound fields in small rooms. The analysis finds an approximate directional reflectogram (ADR) which reveals the approximate arrival directions, time delays and amplitudes of the direct sound and early reflections without using a special or known sound source. A coincident microphone array is used to obtain directional recordings. The recordings are analysed by wavelet packet decomposition to determine the direction of the sound source and select wavelet packet coefficients to reconstruct the estimate of the direct sound. ADR is then computed via deconvolution using this estimate. Experiments have been carried out using synthesized recordings that were obtained from actual room impulse responses measured in two rooms for various source locations. The method estimates the source direction with a mean absolute error of about 7°. Calculated ADRs provide a good estimate of the time delays and arrival directions of acoustical reflections, whereas the amplitudes differ slightly.

Dynamic Neural-Based Buffer Management for Queuing Systems With Self-Similar Characteristics

Buffer management in queuing systems plays an important role in addressing the tradeoff between efficiency measured in terms of overall packet loss and fairness measured in terms of individual source packet loss. Complete partitioning (CP) of a buffer with the best fairness characteristic and complete sharing (CS) of a buffer with the best efficiency characteristic are at the opposite ends of the spectrum of buffer management techniques. Dynamic partitioning buffer management techniques aim at addressing the tradeoff between efficiency and fairness. Ease of implementation is the key issue when determining the practicality of a dynamic buffer management technique. In this paper, two novel dynamic buffer management techniques for queuing systems accommodating self-similar traffic patterns are introduced. The techniques take advantage of the adaptive learning power of perceptron neural networks when applied to arriving traffic patterns of queuing systems. Relying on the water-filling approach, our proposed techniques are capable of coping with the tradeoff between packet loss and fairness issues. Computer simulations reveal that both of the proposed techniques enjoy great efficiency and fairness characteristics as well as ease of implementation.