Transmission efficace en temps réel de la voix sur réseaux ad hoc sans fil

La téléphonie mobile se démocratise et de nouveaux types de réseaux voient le jour, notamment les réseaux ad hoc. Sans focaliser exclusivement sur ces réseaux particuliers, le nombre de communications vocales effectuées chaque minute est en constante augmentation mais les réseaux sont encore souvent victimes d'erreurs de transmission. L'objectif de cette thèse porte sur l'utilisation de méthodes de codage en vue d'une transmission de la voix robuste face aux pertes de paquets, sur un réseau mobile et sans fil perturbé permettant le multichemin. La méthode envisagée prévoit l'utilisation d'un codage en descriptions multiples (MDC) appliqué à un flux de données issu d'un codec de parole bas débit, plus particulièrement l'AMR-WB (Adaptive Multi Rate - Wide Band). Parmi les paramètres encodés par l'AMR-WB, les coefficients de la prédiction linéaire sont calculés une fois par trame, contrairement aux autres paramètres qui sont calculés quatre fois. La problématique majeure réside dans la création adéquate de descriptions pour les paramètres de prédiction linéaire. La méthode retenue applique une quantification vectorielle conjuguée à quatre descriptions. Pour diminuer la complexité durant la recherche, le processus est épaulé d'un préclassificateur qui effectue une recherche localisée dans le dictionnaire complet selon la position d'un vecteur d'entrée. L'application du modèle de MDC à des signaux de parole montre que l'utilisation de quatre descriptions permet de meilleurs résultats lorsque le réseau est sujet à des pertes de paquets. Une optimisation de la communication entre le routage et le processus de création de descriptions mène à l'utilisation d'une méthode adaptative du codage en descriptions. Les travaux de cette thèse visaient la retranscription d'un signal de parole de qualité, avec une optimisation adéquate des ressources de stockage, de la complexité et des calculs. La méthode adaptative de MDC rencontre ces attentes et s'avère très robuste dans un contexte de perte de paquets

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

A General Framework for Analyzing, Characterizing, and Implementing Spectrally Modulated, Spectrally Encoded Signals

Fourth generation (4G) communications will support many capabilities while providing universal, high speed access. One potential enabler for these capabilities is software defined radio (SDR). When controlled by cognitive radio (CR) principles, the required waveform diversity is achieved via a synergistic union called CR-based SDR. Research is rapidly progressing in SDR hardware and software venues, but current CR-based SDR research lacks the theoretical foundation and analytic framework to permit efficient implementation. This limitation is addressed here by introducing a general framework for analyzing, characterizing, and implementing spectrally modulated, spectrally encoded (SMSE) signals within CR-based SDR architectures. Given orthogonal frequency division multiplexing (OFDM) is a 4G candidate signal, OFDM-based signals are collectively classified as SMSE since modulation and encoding are spectrally applied. The proposed framework provides analytic commonality and unification of SMSE signals. Applicability is first shown for candidate 4G signals, and resultant analytic expressions agree with published results. Implementability is then demonstrated in multiple coexistence scenarios via modeling and simulation to reinforce practical utility

VoLTE service implementation in EPS-IMS networks

Diplomová práce popisuje VoLTE službu, vývoj a nasazení LTE (zaváděcí fázi, skutečný LTE stav a výhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhraní atd.) Komunikace mezi uzly a funkce, rozhraní a protokoly jsou používány v průběhu signalizace (SIP SDP) a datový tok (RTCP RTP). Práce stručně popisuje základní toky hovorů, typy nosičů (GBR and N-GBR), a to vytvoření / mazaní nosičů během komunikace. Další část diplomové práce o implementaci volte, instalace a konfigurace IMS. Závěrečná část diplomové práce popisuje zkoušky sítě a, analýzu protokolu.The master's thesis describes VoLTE service, LTE evolution and deployment (deployment phases, actual LTE state and future perspectives etc.), EPC-IMS architecture (functional node description, interfaces etc.). Communications between nodes and functions, interfaces and protocols which are used during signaling (SIP-SDP) and data flow (RTCP RTP). Thesis briefly describe basic call flows, bearers types (GBR and N-GBR) and their establishment/delete during communication. The next part of master's thesis is about VoLTE implementation solutions, IMS installation and configuration. The final part of master's thesis describes the network and protocols tests, analyzes.

Radio Communications

In the last decades the restless evolution of information and communication technologies (ICT) brought to a deep transformation of our habits. The growth of the Internet and the advances in hardware and software implementations modified our way to communicate and to share information. In this book, an overview of the major issues faced today by researchers in the field of radio communications is given through 35 high quality chapters written by specialists working in universities and research centers all over the world. Various aspects will be deeply discussed: channel modeling, beamforming, multiple antennas, cooperative networks, opportunistic scheduling, advanced admission control, handover management, systems performance assessment, routing issues in mobility conditions, localization, web security. Advanced techniques for the radio resource management will be discussed both in single and multiple radio technologies; either in infrastructure, mesh or ad hoc networks

Recent Advances in Signal Processing

The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity

Planning and dynamic spectrum management in heterogeneous mobile networks with QoE optimization

The radio and network planning and optimisation are continuous processes that do not end after the network has been launched. To achieve the best trade-offs, especially between quality and costs, operators make use of several coverage and capacity enhancement methods. The research from this thesis proposes methods such as the implementation of cell zooming and Relay Stations (RSs) with dynamic sleep modes and Carrier Aggregation (CA) for coverage and capacity enhancements. Initially, a survey is presented on ubiquitous mesh networks implementation scenarios and an updated characterization of requirements for services and applications is proposed. The performance targets for the key parameters, delay, delay variation, information loss and throughput have been addressed for all types of services. Furthermore, with the increased competition, mobile operator’s success does not only depend on how good the offered Quality of Service (QoS) is, but also if it meets the end user’s expectations, i.e., Quality of Experience (QoE). In this context, a model for the mapping between QoS parameters and QoE has been proposed for multimedia traffic. The planning and optimization of fixed Worldwide Interoperability for Microwave Access (WiMAX) networks with RSs in conjunction with cell zooming has been addressed. The challenging case of a propagation measurement-based scenario in the hilly region of Covilhã has been considered. A cost/revenue function has been developed by taking into account the cost of building and maintaining the infrastructure with the use of RSs. This part of the work also investigates the energy efficiency and economic implications of the use of power saving modes for RSs in conjunction with cell zooming. Assuming that the RSs can be switched-off or zoomed out to zero in periods when the traffic exchange is low, such as nights and weekends, it has been shown that energy consumption may be reduced whereas cellular coverage and capacity, as well as economic performance may be improved. An integrated Common Radio Resource Management (iCRRM) entity is proposed that implements inter-band CA by performing scheduling between two Long Term Evolution – Advanced (LTE-A) Component Carriers (CCs). Considering the bandwidths available in Portugal, the 800 MHz and 2.6 GHz CCs have been considered whilst mobile video traffic is addressed. Through extensive simulations it has been found that the proposed multi-band schedulers overcome the capacity of LTE systems without CA. Result shown a clear improvement of the QoS, QoE and economic trade-off with CA

Understanding the acoustic implications of digital transmission on fricatives

The aim of this thesis is to provide a better understanding of the acoustic implications of digital transmission on fricatives relevant across research fields. This is motivated by the increasing amount of digital transmitted speech across the world, and the limited knowledge on the effects of digital transmission on consonants. The thesis investigates the fricatives /f/, /θ/, /s/, /ʃ/, /z/, /ð/ and [fj]. Fricatives were expected to be particularly affected by codec compression because of their noise-like and aperiodic structure, which might be mistaken for noise by the codecs. The thesis investigates the effects of the AMR-WB-, Opus-, and MP3 codec using three different bitrates and in live transmission. The acoustic implications were measured as the first four spectral moments, peak frequency, and via spectrographic analysis. These measures were compared between baseline uncompressed WAV files and each of the codec compressed versions. This resulted in three studies. The first two are in controlled conditions i.e. the WAV files are codec compressed via a computer, whereas the third study is live with the speech transmitted between two mobile phones with and without background noise. The findings indicate significant effects of the codec compressions on the spectral measures with segment, codec and bitrate dependent tendencies. The live transmission and background noise generally produced larger effects than the controlled conditions. Intensity played a key role in the magnitude of the effects of the codec compressions and live transmission. This has implications when using codec compressed speech as data, but especially in socio- and forensic phonetics with possible diffusion of sound changes and speaker comparisons. In addition, the results have implications beyond linguistics e.g. in psychology, where clarity of speech plays a role in perceived charisma, and in hearing aid and cochlear implant technology, which both approach speech digitally and incorporate noise reduction

Near-capacity MIMOs using iterative detection

In this thesis, Multiple-Input Multiple-Output (MIMO) techniques designed for transmission over narrowband Rayleigh fading channels are investigated. Specifically, in order to providea diversity gain while eliminating the complexity of MIMO channel estimation, a Differential Space-Time Spreading (DSTS) scheme is designed that employs non-coherent detection. Additionally, in order to maximise the coding advantage of DSTS, it is combined with Sphere Packing (SP) modulation. The related capacity analysis shows that the DSTS-SP scheme exhibits a higher capacity than its counterpart dispensing with SP. Furthermore, in order to attain additional performance gains, the DSTS system invokes iterative detection, where the outer code is constituted by a Recursive Systematic Convolutional (RSC) code, while the inner code is a SP demapper in one of the prototype systems investigated, while the other scheme employs a Unity Rate Code (URC) as its inner code in order to eliminate the error floor exhibited by the system dispensing with URC. EXIT charts are used to analyse the convergence behaviour of the iteratively detected schemes and a novel technique is proposed for computing the maximum achievable rate of the system based on EXIT charts. Explicitly, the four-antenna-aided DSTSSP system employing no URC precoding attains a coding gain of 12 dB at a BER of 10-5 and performs within 1.82 dB from the maximum achievable rate limit. By contrast, the URC aidedprecoded system operates within 0.92 dB from the same limit.On the other hand, in order to maximise the DSTS system’s throughput, an adaptive DSTSSP scheme is proposed that exploits the advantages of differential encoding, iterative decoding as well as SP modulation. The achievable integrity and bit rate enhancements of the system are determined by the following factors: the specific MIMO configuration used for transmitting data from the four antennas, the spreading factor used and the RSC encoder’s code rate.Additionally, multi-functional MIMO techniques are designed to provide diversity gains, multiplexing gains and beamforming gains by combining the benefits of space-time codes, VBLASTand beamforming. First, a system employing Nt=4 transmit Antenna Arrays (AA) with LAA number of elements per AA and Nr=4 receive antennas is proposed, which is referred to as a Layered Steered Space-Time Code (LSSTC). Three iteratively detected near-capacity LSSTC-SP receiver structures are proposed, which differ in the number of inner iterations employed between the inner decoder and the SP demapper as well as in the choice of the outer code, which is either an RSC code or an Irregular Convolutional Code (IrCC). The three systems are capable of operating within 0.9, 0.4 and 0.6 dB from the maximum achievable rate limit of the system. A comparison between the three iteratively-detected schemes reveals that a carefully designed two-stage iterative detection scheme is capable of operating sufficiently close to capacity at a lower complexity, when compared to a three-stage system employing a RSC or a two-stage system using an IrCC as an outer code. On the other hand, in order to allow the LSSTC scheme to employ less receive antennas than transmit antennas, while still accommodating multiple users, a Layered Steered Space-Time Spreading (LSSTS) scheme is proposed that combines the benefits of space-time spreading, V-BLAST, beamforming and generalised MC DS-CDMA. Furthermore, iteratively detected LSSTS schemes are presented and an LLR post-processing technique is proposed in order to improve the attainable performance of the iteratively detected LSSTS system.Finally, a distributed turbo coding scheme is proposed that combines the benefits of turbo coding and cooperative communication, where iterative detection is employed by exchanging extrinsic information between the decoders of different single-antenna-aided users. Specifically, the effect of the errors induced in the first phase of cooperation, where the two users exchange their data, on the performance of the uplink in studied, while considering different fading channel characteristics