Some Results on the Statistics of Delay Terms in SR ARQ on Markov Channels

Abstract-In this paper we explore the packet delay statistics of a Selective Repeat ARQ scheme on a Discrete Time Markov Channel with non-instantaneous round trip delay. In particular, we are interested in obtaining considerations about the queueing delay of the process and also possible comparisons between different delay components. For this reason, we analyze in detail the impact of system parameters, such as the packet arrival rate and the packet error probability, on the terms which constitute the overall delay. Finally, we explore the connection of these numerical evaluations with the QoS requirements connected to delay for multimedia traffic

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

Studies on the performance of some ARQ schemes

This thesis consists of a summary part and seven published articles. All the articles are about performance analysis of ARQ schemes. Two of the publications study the performance of an ARQ scheme with packet combining, called the EARQ (extended ARQ) scheme. In the packet combining algorithm, the bitwise modulo-2 sum of two erroneous copies of a packet is computed to locate the errors. The packet combining algorithm involves a straightforward search procedure, the computational complexity of which easily becomes prohibitive. As a solution to this, a modified scheme is proposed, where the search procedure is attempted only when there are at most Nmax 1s at the output of the modulo-2 adder. In one article, time diversity was utilized, whereas space diversity reception was considered in the other work. The remaining five publications study the throughput performance of adaptive selective-repeat and go-back-N ARQ schemes, where the switching between the transmission modes is done based on the simple algorithm proposed by Y.-D. Yao in 1995. In this method, α contiguous NACKs or β contiguous ACKs indicate changes from 'good' to 'bad' or from 'bad' to 'good' channel conditions, respectively. The numbers α and β are the two design parameters of the adaptive scheme. The time-varying forward channel is modelled by two-state Markov chains, known as Gilbert-Elliott channel models. The states are characterized by bit error rates, packet error rates or fading parameters. The performance of the adaptive ARQ scheme is measured by its average throughput over all states of the system model, which is a Markov chain. A useful upper bound for the achievable average throughput is provided by the performance of an (assumed) ideal adaptive scheme which is always in the 'correct' transmission mode. The optimization of α and β is done based on minimizing the mean-square distance between the actual and the ideal performance curves. Methods of optimizing the packet size(s) used in the adaptive selective-repeat scheme are also proposed.reviewe

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

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

Network Science for IoT

The research work presented in this thesis is based on the concept and defintion of network that can spread in several and different real world contexts. Indeed, we can refer to a network in a telecommunications sense considering a collection of transmitters, receivers, and communication channels that send or are used to send information to one another. However, as a matter of fact, in nature there are other several examples of networks: the human brain is one of them. The relationship between the actors in Hollywood can be studied in terms of network as well, a generic social community can be compared to a network, eco-systems are networks of species. The recent Network Science aims at studying all these systems using a set of common mathematical methods. In the following of the thesis, we will focus on some of well known telecommunications networks issues using standard telecommunications procedures to address them, with relevant reference to video flow transmissions and management of electric vehicles networks. At the same time, different models aiming at reach the same goals in contexts that may differ from a telecommunications setup can be used. In more details, we will evaluate queueing systems, jamming problems, groups recognition in networks, and mobile computing using game theoretic approaches. It is worth noting that this aspect can be also seen in a reverse order. Indeed, we will discuss how standard telecommunications analysis can be used to investigate on problems not directly related to a telecommunications background. In particular, one of our future purposes is to investigate on the brain connectivity that is raising significant interest in the recent scientific society

Network coding for computer networking

Conventional communication networks route data packets in a store-and-forward mode. A router buffers received packets and forwards them intact towards their intended destination. Network Coding (NC), however, generalises this method by allowing the router to perform algebraic operations on the packets before forwarding them. The purpose of NC is to improve the network performance to achieve its maximum capacity also known as max-flow min-cut bound. NC has become very well established in the field of information theory, however, practical implementations in real-world networks is yet to be explored. In this thesis, new implementations of NC are brought forward. The effect of NC on flow error control protocols and queuing over computer networks is investigated by establishing and designing a mathematical and simulation framework. One goal of such investigation is to understand how NC technique can reduce the number of packets required to acknowledge the reception of those sent over the network while error-control schemes are employed. Another goal is to control the network queuing stability by reducing the number of packets required to convey a set of information. A custom-built simulator based on SimEvents® has been developed in order to model several scenarios within this approach. The work in this thesis is divided into two key parts. The objective of the first part is to study the performance of communication networks employing error control protocols when NC is adopted. In particular, two main Automatic Repeat reQuest (ARQ) schemes are invoked, namely the Stop-and-Wait (SW) and Selective Repeat (SR) ARQ. Results show that in unicast point-to point communication, the proposed NC scheme offers an increase in the throughput over traditional SW ARQ between 2.5% and 50.5% at each link, with negligible decoding delay. Additionally, in a Butterfly network, SR ARQ employing NC achieves a throughput gain between 22% and 44% over traditional SR ARQ when the number of incoming links to the intermediate node varies between 2 and 5. Moreover, in an extended Butterfly network, NC offered a throughput increase of up to 48% under an error-free scenario and 50% in the presence of errors. Despite the extensive research on synchronous NC performance in various fields, little has been said about its queuing behaviour. One assumption is that packets are served following a Poisson distribution. The packets from different streams are coded prior to being served and then exit through only one stream. This study determines the arrival distribution that coded packets follow at the serving node. In general this leads to study general queuing systems of type G/M/1. Hence, the objective of the second part of this study is twofold. The study aims to determine the distribution of the coded packets and estimate the waiting time faced by coded packets before their complete serving process. Results show that NC brings a new solution for queuing stability as evidenced by the small waiting time the coded packets spend in the intermediate node queue before serving. This work is further enhanced by studying the server utilization in traditional routing and NC scenarios. NC-based M/M/1 with finite capacity K is also analysed to investigate packet loss probability for both scenarios. Based on the results achieved, the utilization of NC in error-prone and long propagation delay networks is recommended. Additionally, since the work provides an insightful prediction of particular networks queuing behaviour, employing synchronous NC can bring a solution for systems’ stability with packet-controlled sources and limited input buffers

QoS Scheduling in IEEE 802.16 Broadband Wireless Access Networks

With the exploding increase of mobile users and the release of new wireless applications, the high bandwidth requirement has been taking as a main concern for the design and development of the wireless techniques. There is no doubt that broadband wireless access with the support of heterogeneous kinds of applications is the trend in the next generation wireless networks. As a promising broadband wireless access standard, IEEE 802.16 has attracted extensive attentions from both industry and academia due to its high data rate and the inherent media access control (MAC) mechanism, which takes the service differentiation and quality of service (QoS) provisioning into account. To achieve service differentiation and QoS satisfaction for heterogenous applications is a very complicated issue. It refers to many fields, such as connection admission control (CAC), congestion control, routing algorithm, MAC protocol, and scheduling scheme. Among these fields, packet scheduling plays one of the most important roles in fulfilling service differentiation and QoS provisioning. It decides the order of packet transmissions, and provides mechanisms for the resource allocation and multiplexing at the packet level to ensure that different types of applications meet their service requirements and the network maintains a high resource utilization. In this thesis, we focus on the packet scheduling for difficult types of services in IEEE 802.16 networks, where unicast and mulitcast scheduling are investigated. For unicast scheduling, two types of services are considered: non-real-time polling service (nrtPS) and best effort (BE) service. We propose a flexible and efficient resource allocation and scheduling framework for nrtPS applications to achieve a tradeoff between the delivery delay and resource utilization, where automatic repeat request (ARQ) mechanisms and the adaptive modulation and coding (AMC) technique are jointly considered. For BE service, considering the heterogeneity of subscriber stations (SSs) in IEEE 802.16 networks, we propose the weighted proportional fairness scheduling scheme to achieve the flexible scheduling and resource allocation among SSs based on their traffic demands/patterns. For multicast scheduling, a cooperative multicast scheduling is proposed to achieve high throughput and reliable transmission. By using the two-phase transmission model to exploit the spatial diversity gain in the multicast scenario, the proposed scheduling scheme can significantly improve the throughput not only for all multicast groups, but also for each group member. Analytical models are developed to investigate the performance of the proposed schemes in terms of some important performance measurements, such as throughput, resource utilization, and service probability. Extensive simulations are conducted to illustrate the efficient of the proposed schemes and the accuracy of the analytical models. The research work should provide meaningful guidelines for the system design and the selection of operational parameters, such as the number of TV channels supported by the network, the achieved video quality of each SS in the network, and the setting of weights for SSs under different BE traffic demands

Enabling Technologies for Ultra-Reliable and Low Latency Communications: From PHY and MAC Layer Perspectives

© 1998-2012 IEEE. Future 5th generation networks are expected to enable three key services-enhanced mobile broadband, massive machine type communications and ultra-reliable and low latency communications (URLLC). As per the 3rd generation partnership project URLLC requirements, it is expected that the reliability of one transmission of a 32 byte packet will be at least 99.999% and the latency will be at most 1 ms. This unprecedented level of reliability and latency will yield various new applications, such as smart grids, industrial automation and intelligent transport systems. In this survey we present potential future URLLC applications, and summarize the corresponding reliability and latency requirements. We provide a comprehensive discussion on physical (PHY) and medium access control (MAC) layer techniques that enable URLLC, addressing both licensed and unlicensed bands. This paper evaluates the relevant PHY and MAC techniques for their ability to improve the reliability and reduce the latency. We identify that enabling long-term evolution to coexist in the unlicensed spectrum is also a potential enabler of URLLC in the unlicensed band, and provide numerical evaluations. Lastly, this paper discusses the potential future research directions and challenges in achieving the URLLC requirements