Online Versus Offline Rate in Streaming Codes for Variable-Size Messages

Providing high quality-of-service for live communication is a pervasive challenge which is plagued by packet losses during transmission. Streaming codes are a class of erasure codes specifically designed for such low-latency streaming communication settings. We consider the recently proposed setting of streaming codes under variable-size messages which reflects the requirements of applications such as live video streaming. In practice, streaming codes often need to operate in an "online" setting where the sizes of the future messages are unknown. Yet, previously studied upper bounds on the rate apply to "offline" coding schemes with access to all (including future) message sizes. In this paper, we evaluate whether the optimal offline rate is a feasible goal for online streaming codes when communicating over a burst-only packet loss channel. We identify two broad parameter regimes where, perhaps surprisingly, online streaming codes can, in fact, match the optimal offline rate. For both of these settings, we present rate-optimal online code constructions. For all remaining parameter settings, we establish that it is impossible for online coding schemes to attain the optimal offline rate.Comment: 16 pages, 2 figures, this is an extended version of the IEEE ISIT 2020 paper with the same titl

Efficient and Effective Schemes for Streaming Media Delivery

The rapid expansion of the Internet and the increasingly wide deployment of wireless networks provide opportunities to deliver streaming media content to users at anywhere, anytime. To ensure good user experience, it is important to battle adversary effects, such as delay, loss and jitter. In this thesis, we first study efficient loss recovery schemes, which require pure XOR operations. In particular, we propose a novel scheme capable of recovering up to 3 packet losses, and it has the lowest complexity among all known schemes. We also propose an efficient algorithm for array codes decoding, which achieves significant throughput gain and energy savings over conventional codes. We believe these schemes are applicable to streaming applications, especially in wireless environments. We then study quality adaptation schemes for client buffer management. Our control-theoretic approach results in an efficient online rate control algorithm with analytically tractable performance. Extensive experimental results show that three goals are achieved: fast startup, continuous playback in the face of severe congestion, and maximal quality and smoothness over the entire streaming session. The scheme is later extended to streaming with limited quality levels, which is then directly applicable to existing systems

First-Passage Time and Large-Deviation Analysis for Erasure Channels with Memory

This article considers the performance of digital communication systems transmitting messages over finite-state erasure channels with memory. Information bits are protected from channel erasures using error-correcting codes; successful receptions of codewords are acknowledged at the source through instantaneous feedback. The primary focus of this research is on delay-sensitive applications, codes with finite block lengths and, necessarily, non-vanishing probabilities of decoding failure. The contribution of this article is twofold. A methodology to compute the distribution of the time required to empty a buffer is introduced. Based on this distribution, the mean hitting time to an empty queue and delay-violation probabilities for specific thresholds can be computed explicitly. The proposed techniques apply to situations where the transmit buffer contains a predetermined number of information bits at the onset of the data transfer. Furthermore, as additional performance criteria, large deviation principles are obtained for the empirical mean service time and the average packet-transmission time associated with the communication process. This rigorous framework yields a pragmatic methodology to select code rate and block length for the communication unit as functions of the service requirements. Examples motivated by practical systems are provided to further illustrate the applicability of these techniques.Comment: To appear in IEEE Transactions on Information Theor

Optimization of protection techniques based on FEC codes for the transmission of multimedia packetized streams

Esta tesis presenta dos modelos novedosos de arquitecturas basadas en esquemas FEC con el fin de proteger flujos de paquetes con contenido multimedial, para comunicaciones en tiempo real y en canales donde las pérdidas se producen en ráfagas. El objetivo de estos diseños ha sido maximizar la eficiencia de los códigos FEC considerados. Por un lado, el primer modelo busca alcanzar un menor coste computacional para los códigos de Reed- Solomon, ya que su conocida capacidad de recuperación para todo tipo de canales necesita un coste computacional elevado. Por otro lado, en el caso de los códigos LDPC, se ha perseguido aumentar la capacidad de recuperación de estos códigos operando en canales con errores en ráfagas, teniendo en cuenta que los códigos LDPC no están directamente diseñados para este tipo de entorno. El modelo aplicado a los códigos de Reed-Solomon se denomina inter-packet symbol approach. Este esquema consiste en una estructura alternativa que asocia los bits de los símbolos del código en distintos paquetes. Esta característica permite aprovechar de forma mejor la capacidad de recuperación de los códigos de Reed-Solomon frente a pérdidas de paquetes en ráfagas. Las prestaciones de este esquema han sido estudiadas en términos de tiempo de codificación/decodificación versus capacidad de recuperación y han sido comparados con otros esquemas propuestos en literatura. El análisis teórico ha demostrado que el enfoque propuesto permite la utilización de Campos de Galois de menor dimensión con respecto a otras soluciones. Esto se traduce en una disminución del tiempo de codificación/decodificación requerido, mientras que mantiene una capacidad de recuperación comparable. Aunque la utilización de los códigos LDPC está típicamente orientada hacía canales con errores uniformemente distribuidos (canales sin memoria) y para bloques de información largos, esta tesis surgiere el uso de este tipo de códigos FEC a nivel de aplicación, para canales con pérdidas en ráfagas y para entornos de comunicación de tiempo real, es decir, con una latencia de transmisión muy baja. Para satisfacer estas limitaciones, la configuración apropiada de los parámetros de un código LDPC ha sido determinada usando bloques de información pequeños y adaptando el código FEC de modo que sea capaz de recuperar paquetes perdidos en canales con errores en ráfagas. Para ello, primeramente se ha diseñado un algoritmo que realiza una estimación de las capacidades de recuperación del código LDPC para un canal con pérdidas en ráfagas. Una vez caracterizado el código, se ha diseñado un segundo algoritmo que optimiza la estructura del código en términos de capacidad de recuperación para las características especificas del canal con memoria, generado una versión modificada del código LDPC, adaptada al canal con perdidas en ráfagas. Finalmente, los dos esquemas FEC propuestos, han sido evaluado experimentalmente en entornos de simulación usando canales con errores en ráfagas y se han comparado con otras soluciones y esquemas ya existentes. ABSTRACT This thesis presents two enhanced FEC-based schemes to protect real-time packetized multimedia streams in bursty channels. The objective of these novel architectures has been the optimization of existing FEC codes, that is, Reed-Solomon codes and LDPC codes. On the one hand, the optimization is focused on the achievement of a lower computational cost for Reed-Solomon codes, since their well known robust recovery capability against any type of losses needs a high complexity. On the other hand, in the case of LDPC codes, the optimization is addressed to increase the recovery capabilities for a bursty channel, since they are not specifically designed for the scenario considered in this thesis. The scheme based on Reed-Solomon codes is called inter-packet symbol approach, and it consists in an alternative bit structure that allocates each symbol of a Reed- Solomon code in several media packets. This characteristic permits to exploit better the recovery capability of Reed-Solomon codes against bursty packet losses. The performance of this scheme has been studied in terms of encoding/decoding time versus recovery capability, and compared with other proposed schemes in the literature. The theoretical analysis has shown that the proposed approach allows the use of a lower size of the Galois Fields compared to other solutions. This lower size results in a decrease of the required encoding/decoding time while keeping a comparable recovery capability. Although the use of LDPC codes is typically addressed for channels where losses are uniformly distributed (memoryless channels) and for large information blocks, this thesis suggests the use of this type of FEC codes at the application layer, in bursty channels and for real-time scenario, where low transmission latency is requested. To fulfill these constraints, the appropriate configuration parameters of an LDPC scheme have been determined using small blocks of information and adapting the FEC code to be capable of recovering packet losses in bursty environments. This purpose is achieved in two steps. The first step is performed by an algorithm that estimates the recovery capability if a given LDPC code in a burst packet loss network. The second step is the optimization of the code: an algorithm optimizes the code structure in terms of recovery capability against the specific behavior of the channel with memory, generating a burst oriented version of the considered LDPC code. Finally, for both proposed FEC schemes, experimental results have been carried out in a simulated transmission channel to assess the performances of the schemes and compared to several other schemes

A hybrid packet loss recovery technique in wireless ad hoc networks

TCP utilization in wireless networks poses certain problems due to its inability to distinguish packet losses caused by congestion from those caused by frequent wireless errors, leading to degraded network performance. To avoid these problems and to minimize the effect of intensive channel contention in wireless networks, this work presents a new Hybrid ARQ technique for reliable and efficient packets transfer in static wireless ad hoc network. It is a combination of recent FEC based Raptor coding technique with ARQ based selective retransmission method, which outperforms purely ARQ based method. In contrast to most Hybrid ARQ techniques, which usually employ a byte level FEC, we mostly use packet level FEC in our simulations for the data transfer, on top of less frequent ARQ to recover the residual errors. Existing packet level FEC methods are mostly based on simple parity check codes or Reed Solomon codes with erasure decoding; in this work we use the recent raptor codes. We also introduce the notion of adaptive redundancy which helps to achieve better average network performance and to further improve the redundancy efficiency

ENSURE: A Time Sensitive Transport Protocol to Achieve Reliability Over Wireless in Petrochemical Plants

As society becomes more reliant on the resources extracted in petroleum refinement the production demand for petrochemical plants increases. A key element is producing efficiently while maintaining safety through constant monitoring of equipment feedback. Currently, temperature and flow sensors are deployed at various points of production and 10/100 Ethernet cable is installed to connect them to a master control unit. This comes at a great monetary cost, not only at the time of implementation but also when repairs are required. The capability to provide plant wide wireless networks would both decrease investment cost and downtime needed for repairs. However, the current state of wireless networks does not provide any guarantee of reliability, which is critical to the industry. When factoring in the need for real-time information, network reliability further decreases. This work presents the design and development of a series of transport layer protocols (coined ENSURE) to provide time-sensitive reliability. More specifically three versions were developed to meet specific needs of the data being sent. ENSURE 1.0 addresses reliability, 2.0 enforces a time limit and the final version, 3.0, provides a balance of the two. A network engineer can set each specific area of the plant to use a different version of ENSURE based network performance needs for the data it produces. The end result being a plant wide wireless network that performs in a timely and reliable fashion