271 research outputs found

    Performance of cognitive stop-and-wait hybrid automatic repeat request in the face of imperfect sensing

    No full text
    The cognitive radio (CR) paradigm has the potential of improving the exploitation of the electromagnetic spectrum by detecting instantaneously unoccupied spectrum slots allocated to primary users (PUs). In order to support the process of spectrum reuse, we consider a CR scheme, which senses and opportunistically accesses a PU's spectrum for communication between a pair of nodes relying on the stop-and-wait hybrid automatic repeat request (SW-HARQ) protocol. This arrangement is represented by the cognitive SW-HARQ (CSW-HARQ), where the availability/unavailability of the PU's channel is modeled as a two-state Markov chain having OFF and ON states, respectively. Once the cognitive user (CU) finds that the PU's channel is available (i.e., in the OFF state), the CU transmits data over the PU channel's spectrum, while relying on the principles of SW-HARQ. We investigate both the throughput and the delay of CSW-HARQ, with a special emphasis on the impact of the various system parameters involved in the scenarios of both perfect and imperfect spectrum sensing. Furthermore, we analyze both the throughput as well as the average packet delay and end-to-end packet delay of the CSW-HARQ system. We propose a pair of analytical approaches: 1) the probability-based and 2) the discrete time Markov chain-based. Closed-form expressions are derived for both the throughput and the delay under the perfect and imperfect sensing environments that are validated by simulation. We demonstrate that the activity of PUs, the transmission reliability of the CU, and the sensing environment have a significant impact on both the throughput and the delay of the CR system

    Data Transmission in the Presence of Limited Channel State Information Feedback

    Get PDF

    Cross-layer performance control of wireless channels using active local profiles

    Get PDF
    To optimize performance of applications running over wireless channels state-of-the-art wireless access technologies incorporate a number of channel adaptation mechanisms. While these mechanisms are expected to operate jointly providing the best possible performance for current wireless channel and traffic conditions, their joint effect is often difficult to predict. To control functionality of various channel adaptation mechanisms a new cross-layer performance optimization system is sought. This system should be responsible for exchange of control information between different layers and further optimization of wireless channel performance. In this paper design of the cross-layer performance control system for wireless access technologies with dynamic adaptation of protocol parameters at different layers of the protocol stack is proposed. Functionalities of components of the system are isolated and described in detail. To determine the range of protocol parameters providing the best possible performance for a wide range of channel and arrival statistics the proposed system is analytically analyzed. Particularly, probability distribution functions of the number of lost frames and delay of a frame as functions of first- and second-order wireless channel and arrival statistics, automatic repeat request, forward error correction functionality, protocol data unit size at different layers are derived. Numerical examples illustrating performance of the whole system and its elements are provided. Obtained results demonstrate that the proposed system provide significant performance gains compared to static configuration of protocols

    ARQ protocol for joint source and channel coding and its applications

    Get PDF
    Shannon\u27s separation theorem states that for transmission over noisy channels, approaching channel capacity is possible with the separation of source and channel coding. Practically, the situation is different. Infinite size blocks are needed to achieve this theoretical limit. Also, time-varying channels require a different approach. This leads to many approaches for source and channel coding. This dissertation will address a joint source and channel coding that suits Automatic Repeat Request (ARQ) application and applies it to packet switching networks. Following aspects of the proposed joint source and channel coding approach will be presented: The design of the proposed joint source and channel coding scheme. The approach is based on a variable length coding scheme which adapts the arithmetic coding process for joint source and channel coding. The protocol using this joint source and channel coding scheme in communication systems. The error recovery technique of the proposed scheme is presented. The application of the scheme and protocol. The design is applied to wireless TCP network and real-time video transmissions. The coding scheme embeds the redundancy needed for error detection in source coding stage. The self-synchronization property of lossless compression is utilized by decoder to detect channel errors. With this approach, error detection may be delayed. The delay in detection is referred to as error propagation distance. This work analyzes the distribution of error propagation distance. The error recovery technique of this joint source and channel coding for ARQ (JARQ) protocol is analyzed. Throughput is studied using signal flow graph for both independent channel and nonindependent channels. A packet combining technique is presented which utilizes the non-uniform distribution of error propagation distance to increase the throughput. The proposed scheme may be applied to many areas. In particular, two applications are discussed. A TCP/JARQ protocol stack is introduced and the coordination between TCP and JARQ layers is discussed to maximize system performance. By limiting the number of retransmission, the proposed scheme is applied to real-time transmission to meet timing requirement

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

    Full text link
    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

    Analysis of discrete-time queueing systems with multidimensional state space

    Get PDF

    Analysis of Error Control and Congestion Control Protocols

    Get PDF
    This thesis presents an analysis of a class of error control and congestion control protocols used in computer networks. We address two kinds of packet errors: (a) independent errors and (b) congestion-dependent errors. Our performance measure is the expected time and the standard deviation of the time to transmit a large message, consisting of N packets. The analysis of error control protocols. Assuming independent packet errors gives an insight on how the error control protocols should really work if buffer overflows are minimal. Some pertinent results on the performance of go-back-n, selective repeat, blast with full retransmission on error (BFRE) and a variant of BFRE, the Optimal BFRE that we propose, are obtained. We then analyze error control protocols in the presence of congestion-dependent errors. We study the selective repeat and go-back-n protocols and find that irrespective of retransmission strategy, the expected time as well as the standard deviation of the time to transmit N packets increases sharply the face of heavy congestion. However, if the congestion level is low, the two retransmission strategies perform similarly. We conclude that congestion control is a far more important issue when errors are caused by congestion. We next study the performance of a queue with dynamically changing input rates that are based on implicit or explicit feedback. This is motivated by recent proposals for adaptive congestion control algorithms where the sender\u27s window size is adjusted based on perceived congestion level of a bottleneck node. We develop a Fokker-Planck approximation for a simplified system; yet it is powerful enough to answer the important questions regarding stability, convergence (or oscillations), fairness and the significant effect that delayed feedback plays on performance. Specifically, we find that, in the absence of feedback delay, a linear increase/exponential decrease rate control algorithm is provably stable and fair. Delayed feedback, however, introduces cyclic behavior. This last result not only concurs with some recent simulation studies, it also expounds quantitatively on the real causes behind them
    corecore