Cet exposé analyse les perspectives large bande des réseaux PMR, à travers l'évaluation du candidat LTE, et la proposition d'une possible évolution du codage canal, la solution brevetée des codes turbo à protection non uniforme. Une première étude dans le chapitre 2 se concentre sur l'analyse multi-couche et l'identification des problèmes clé des communications de voix et de vidéo sur un réseau LTE professionnel. Les capacités voix et vidéo sont estimées pour les liens montant et descendant de la transmission LTE, et l'efficacité spectrale de la voix en lien descendant est comparée à celle de PMR et GSM. Ce chapitre souligne certains points clé de l'évolution de LTE. S'ils étaient pas résolus par la suite, LTE se verrait perdre de sa crédibilité en tant que candidat à l'évolution de la PMR. Une telle caractéristique clé des réseaux PMR est le codage canal à protection non uniforme, qui pourrait être adapté au système LTE pour une évolution aux contraintes de la sécurité publique. Le chapitre 3 introduit cette proposition d'évolution, qui a été brevetée: les turbo codes à protection non uniforme intégrée. Nous proposons une nouvelle approche pour le codage canal à protection non uniforme à travers les codes turbo progressives hiérarchiques. Les configurations parallèles et séries sont analysées. Les mécanismes de protection non uniformes sont intégrés dans la structure de l'encodeur même à travers l'insertion progressif et hiérarchique de nouvelles données de l'utilisateur. Le turbo décodage est modifié pour exploiter de façon optimale l'insertion progressive de données dans le processus d'encodage et estimer hiérarchiquement ces données. Les propriétés des structures parallèles et séries sont analysées à l'aide d'une analogie aux codes pilotes, ainsi qu'en regardant de plus près leurs caractéristiques de poids de codage. Le taux de transmission virtuel et les représentations des graphs factor fournissent une meilleure compréhension de ces propriétés. Les gains de codage sont évalués à l'aide de simulations numériques, en supposant des canaux de transmission radio statiques et dynamiques, et en utilisant des codes de référence. Enfin, dans le chapitre 4, l'idée breveté du code turbo parallal progressif et hiérarchique (PPHTC) est évaluée sur la plateforme LTE. Une description détaillée de l'architecture des bearers de LTE est donnée, et ses conséquences sont discutées. Le nouveau codage canal est inséré et évalué sur cette plateforme, et ses performances sont comparées avec des schémas de transmission typique à LTE. L'analyse de la qualité de la voix aide à conclure sur l'efficacité de la solution proposée dans un système de transmission réel. Pourtant, même si cette dernière donne les meilleurs résultats, d'avantage d'optimisations devraient être envisagées pour obtenir des gains améliorés et mieux exploiter le potentiel du codage proposé. L'exposé se conclut dans le chapitre 5 et une courte discussion présente les futures perspectives de recherche

Modern speech communication technologies expose users to perceptual quality degradations that were not experienced earlier with conventional telephone systems. Since perceived speech quality is a major contributor to the end user's perception of quality of service, speech quality estimation has become an important research field. In this dissertation, perceptual quality estimators are proposed for several emerging speech communication applications, in particular for i) wireless communications with noise suppression capabilities, ii) wireless-VoIP communications, iii) far-field hands-free speech communications, and iv) text-to-speech systems. First, a general-purpose speech quality estimator is proposed based on statistical models of normative speech behaviour and on innovative techniques to detect multiple signal distortions. The estimators do not depend on a clean reference signal hence are termed ``blind." Quality meters are then distributed along the network chain to allow for both quality degradations and quality enhancements to be handled. In order to improve estimation performance for wireless communications, statistical models of noise-suppressed speech are also incorporated. Next, a hybrid signal-and-link-parametric quality estimation paradigm is proposed for emerging wireless-VoIP communications. The algorithm uses VoIP connection parameters to estimate a base quality representative of the packet switching network. Signal-based distortions are then detected and quantified in order to adjust the base quality accordingly. The proposed hybrid methodology is shown to overcome the limitations of existing pure signal-based and pure link parametric algorithms. Temporal dynamics information is then investigated for quality diagnosis for hands-free speech communications. A spectro-temporal signal representation, where speech and reverberation tail components are shown to be separable, is used for blind characterization of room acoustics. In particular, estimators of reverberation time, direct-to-reverberation energy ratio, and reverberant speech quality are developed. Lastly, perceptual quality estimation for text-to-speech systems is addressed. Text- and speaker-independent hidden Markov models, trained on naturally produced speech, are used to capture normative spectral-temporal information. Deviations from the models, computed by means of a log-likelihood measure, are shown to be reliable indicators of multiple quality attributes including naturalness, fluency, and intelligibility. / Thesis (Ph.D, Electrical & Computer Engineering) -- Queen's University, 2008-12-22 14:54:49.28

In this dissertation, four approaches to improve Voice over Internet Protocol (VoIP) security is proposed. The first two approaches are aimed at encrypting/decrypting and authenticating VoIP packets, whereas the last two approaches are aimed at key exchange and user authentication. For the first contribution, a reconfigurable, high throughput hardware implementation for the different block cipher operational modes is proposed. The proposed architecture is unified: and it combines multiple related functions on the same architecture. In other words, it has the ability to encrypt/decrypt a plaintext/ciphertext efficiently using different operational modes. Moreover, it has the ability to ensure data integrity using different operational modes. The proposed architecture is tested using the most widely used block ciphers: DES, TDES, AES-128 AES-192, AES-256, and IDEA. The proposed architecture implementation i, ;anal z d and evaluated in comparing it against other iniplenientaticls. Eta, the second contribution, a high speed, deep-pipelined architecture for AES algorithm based on the composite field approach targeting VoIP applications is proposed. A new algorithm for finding the isomorphic mapping matrix to work for any irreducible polynomial, not only the primitive polynomials, is proposed. Moreover, the modified algorithm is used to find the optimum matrix that gives the minimum delay. The matrix is then used to implement the SubBytes/InvSubBytes transformation using composite fields, which in turn allows Its to design a very high speed deep-pipelined architecture. As a result of using the optimized matrix, a processing throughput of 49.401 Gbps is achieved, which is twice as fast as the fastest design introduced before. Another feature of this architecture is the separation of the encryption circuit from the decryption circuit to allow concurrent encryption and decryption, which facilitates full duplex encryption/decryption for VolP applications. For the third contribution. a high speed. low area ALU to perform field operations required for cryptographic applications is proposed. Although the proposed architecture- works for any cryptographic application, an ECC implementation for VoIP applications is targeted. A processor array design space exploration for GF(2m) multiplier is conducted, fins exploration results in different processor array configurations. Among these configurations, the fastest one is chosen since VolP applications are targeted. The multiplier architecture is then modified to work as a squarer. Based on the multiplier architecture, a unified architecture to calculate addition, multiplication, squaring, and inversion is proposed. The overall area is optimized by using three type's ''1 processing elements instead NI using a . e!;'tdeir processing element everywhere. NIST-recommended irreducible polynomials is used. which makes our deign secure and more suitable for cryptographic applications. The proposed architecture is implemented for GF(2 163). GF(2 283) and GF(2 571) on a Xilinx XC2V 4000-6 device to verify the proposed architecture and measure its performance. A maximum frequency of 261 MHz is achieved- which allows the architecture to calculate GE(2 163) multiplication in 640 ns and inversion in 1-40.357. As a fourth contribution. a high speed ECC architecture based on a high-radix scalar multiplication is proposed. This architecture is optimized for VoIP applications First. a new high-radix scalar multiplication algorithm is proposed. Then. a merged double-and-add elliptic curve ALU based on the proposed algorithm is designed. The merged double-and-add ALU combines point doubling and adding operations on one architecture. which in turn reduces the critic-al path delay. The ECC' processor utilizes the previously proposed field ALU. which implements Addition. squaring. multiplication. and division over GF(2m) A maximum frequency of 253 MHz is achieved. which allows the architecture to calculate GF(2 163) scalar multiplication for radix 2 8 in 9 u.s. At a minimum our results for GF(2 163). show a speedup ranging from 1.5 to 326 times in comparison to previous FPGA implementations and a speedup ranging from 1.1 to .5.6 times in comparison to previous ASPIC implementations.

The issues involved in Internet telephony (Voice over Internet Protocol (VoIP) device) can be best understood by actually implementing a VoIP device and studying its performance. In this regard, an Internet telephony device, providing full duplex voice communication over internet, and a user interface program have been developed. In the process, a number of implementation issues came into focus, which we have touched upon in this thesis. Transport layer network protocols are discussed in the concept of real time streaming applications and Real Time Protocol (RTP) is modified to use as transport layer protocol in developed VoIP device. Adaptive playout buffering algorithms are studied and compared with each other by trace driven simulation experiments with objective measures. A method to solve clock synchronization problem in streaming internet applications is presented. One way and round trip delay measurement functionalities are added to the VoIP device, so that device can be used to investigate the network characteristics.

High-performance networking (HPN) is of significance today in order to enable next-generation applications using wired and wireless networks. Some of the examples of HPN include low-latency industrial sensing, monitoring and automation using Wireless Sensor Networks (WSNs). HPN however requires protocol optimization at many layers of the open system interface (OSI) network model in order to meet the stringent performance constraints of the given applications. Furthermore, these protocols need to be impervious to denial of service (DoS) and distributed DoS (DDoS) attacks. Some of the key performance aspects of HPN are low point-to-point and end-to-end latency, high reliability of transmitted frames and performance predictability under various network load situations. This work focuses on two discrete issues in designing protocols for HPN applications. The first research issue looks at the Medium Access Control (MAC) layer of the OSI network model for designing of MAC protocols that provide low-latency and high reliability for point-to-point communication under a WSN. Existing standards in this area are governed by IEEE 802.15.4 specification which defines protocols for MAC and PHY layers for short-range, low bit-rate, and low-cost wireless networks. However, the IEEE 802.15.4 specification is inefficient in terms of latency and reliability performance and, as a result, is unable to meet the stringent operational requirements as defined by counterpart wired sensor networks. Work presented under current research issue describes new MAC protocols that are able to show low-latency transmission performance under strict timing constants for power limited WSNs. This enhancement of the MAC protocols is named extended GTS (XGTS) contained under extended CFP (ECFP) and is published under the IEEE's 802.15.4e standard. The second research issue focuses on the application layer of the OSI network model to design protocols that enhance the robustness of the text based protocols to various traffic inputs. The purpose of this is to increase the reliability of the given text based application layer protocol under a varied load. Session Initiation Protocol (SIP) is used as a case study and the work aims to build algorithms that ensure that SIP can continue to function under specific traffic conditions, which would otherwise deem the protocol useless due to DoS and DDoS attacks. Proposed algorithms investigate techniques that enhance the robustness of the SIP against parsing attacks without performing a deep parse of the protocol data unit (PDU). The desired effect of this is to reduce the time spent in parsing the SIP messages at a SIP router and as a result increase the number of SIP messages processed per unit time at a SIP router.

Dissertação (mestrado)—Universidade de Brasília, Instituto de Ciências Exatas, Departamento de Ciência da Computação, Mestrado Profissional em Computação Aplicada, 2015. / Submitted by Albânia Cézar de Melo (albania@bce.unb.br) on 2016-01-28T11:53:24Z No. of bitstreams: 1 2015_ClaudioXavierOliveira.pdf: 6916242 bytes, checksum: 945503a644d7ced55dd003d0dad373a3 (MD5) / Approved for entry into archive by Raquel Viana(raquelviana@bce.unb.br) on 2016-01-28T18:44:42Z (GMT) No. of bitstreams: 1 2015_ClaudioXavierOliveira.pdf: 6916242 bytes, checksum: 945503a644d7ced55dd003d0dad373a3 (MD5) / Made available in DSpace on 2016-01-28T18:44:42Z (GMT). No. of bitstreams: 1 2015_ClaudioXavierOliveira.pdf: 6916242 bytes, checksum: 945503a644d7ced55dd003d0dad373a3 (MD5) / Nos últimos anos, temos observado um crescimento de uso e evolução das redes padrão IEEE 802.11. Atualmente, a Universidade de Brasília conta com uma infraestrutura de WLAN (do Inglês, Wireless Local Area Network) para atender os usuários de seus campi. Por meio dessa infraestrutura, a Universidade vem buscando alterativas para melhor atender a seus usuários, preferencialmente sem custos adicionais. Uma das alternativas é a implantação do serviço de voz sobre a rede sem fio da Instituição, em que os usuários utilizarão seus dispositivos móveis na comunicação, dispondo da facilidade de mobilidade. Ao se utilizar serviços de voz sobre redes sem fio, é necessário que parâmetros como atraso, variação de atraso e perda de pacotes estejam de acordo com recomendações para uma boa qualidade nas chamadas telefônicas. Além disso, prover mobilidade em redes padrão IEEE 802.11 envolve a transição do usuário móvel entre pontos de acesso, sendo um dos fatores de degradação da qualidade auditiva da voz. A solução de rede sem fio da UnB dispõe de tecnologias, como ClientMatch e os padrões IEEE 802.11k e 802.11r, que auxiliam o usuário móvel no processo de transição. Nesse contexto, o trabalho em questão buscou avaliar o impacto da tecnologia ClientMatch e dos padrões IEEE 802.11k e 802.11r no suporte à mobilidade e tráfego de voz, aferindo a qualidade auditiva da voz e o desempenho dos principais parâmetros de QoS (do Inglês, Quality of Service) em uma rede VoIP (do Inglês, Voice over IP). Mediante os experimentos realizados foi possível verificar que a tecnologia ClientMatch – Sticky Client não está voltada para suporte a mobilidade, apresentando baixo índice de atuação. As tecnologias IEEE 802.11k e 802.11r não produziram os resultados esperados nas avaliações realizadas. Mesmo com baixos índices de atuação da tecnologia ClientMatch – Sticky Client e resultados inesperados das soluções 802.11k e 802.11r, a rede WLAN da UnB permite um bom desempenho para o tráfego de voz mediante mobilidade, com níveis aceitáveis de qualidade da voz, devendo-se atentar para o processo de transição entre pontos de acesso, ClientMatch – Band Steering e o distanciamento entre os pontos de acesso. Além dos experimentos que buscaram avaliar o impacto da tecnologia ClientMatch e dos padrões IEEE 802.11k e 802.11r no suporte à mobilidade e tráfego de voz, foram realizados testes com CODECs (do Inglês, Coder–Decoder) mediante mobilidade do cliente móvel. O codificador–decodificador G711 foi o que apresentou melhores resultados nos experimentos executados com relação a qualidade auditiva da voz. / In recent years, we have observed an increase of use and evolution of the IEEE 802.11 based-networks. Currently, the University of Brasília (UnB) owns and operate a WLAN (Wireless Local Area Network) infrastructure to serve users of its campuses. Through this infrastructure, UnB is seeking alternatives to better serve its users, preferably without increasing its costs. An alternative is the implementation of voice over the wireless network of the Institution, where users utilize their mobile devices for communication while providing mobility. When using voice over wireless networks, parameters such as delay, delay variation and packet loss must comply with the recommendations for reasonable phone calls quality. Furthermore, providing mobility in IEEE 802.11 networks involves the transition of the mobile user between access points and this very process is one of the key factors of voice quality degradation. The UNB wireless networking solution encompasses technologies such as ClientMatch, IEEE 802.11r and 802.11k standards that assist the mobile user in the transition process. In this context, this work evaluates the impact of the ClientMatch technology, IEEE 802.11r and 802.11k standards in supporting mobility and voice traffic, while evaluating the audio quality, performance of the main QoS (Quality of Service) parameters in a VoIP network (Voice over IP). The conducted experiments showed that the ClientMatch – Sticky Client technology is not tailored for mobility support, presenting low rate of action, while the IEEE 802.11r and 802.11k technologies have not produced the expected results. Even with the low rates of operation of ClientMatch – Sticky Client technology and unexpected results of the solutions 802.11k and 802.11r, the UnB WLAN infrastructure provided good performance for voice traffic, even under moderate mobility, attaining acceptable levels of voice quality; nevertheless, the process of transition between access points, ClientMatch – Band Steering and the distance between access points may impact in the results. In addition to the experiments that seek to evaluate the impact of technology ClientMatch and IEEE 802.11r and 802.11k standards in supporting mobility and voice traffic, tests were performed with CODECs (Coder - Decoder) by mobility of mobile client. The encoder – decoder G711 showed the best results in the experiments performed with respect to the voice quality.

Made available in DSpace on 2015-04-11T14:03:17Z (GMT). No. of bitstreams: 1 DISSERTACAO CLAUDIA SUZANY.pdf: 727638 bytes, checksum: 99ca8c710111acef29c2891b5586f81b (MD5) Previous issue date: 2008-11-12 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / This thesis presents a prototype of admission control for VoIP calls in wireless mesh networks, called CAC-RM, using the IEEE 802.11g protocol. This admission control goal is to avoid that incoming VoIP data flows consume excessive resources and cause degradation in the established ones. The admission control proposed in our approach only accepts new VoIP calls if the inprogress calls quality do not become degraded. The main features of our approach are: (i) an estimated ocupation time at the channel with VoIP traffic and with traffic without quality requirements, so called BE (Best effort); (ii) the resources reservation for VoIP traffic; (iii) an integration of our prototype to the OLSR proactive routing protocol (sending control messages with information about the occupancy node time with BE and VoIP traffic); (iv) an intra-flow interference estimate with admission control integrated; (v) regulation for BE traffic; and (vi) prioritization to sending and receiving VoIP traffic. The prototype was initially evaluated by simulations of a wireless mesh network in a scenario where all nodes are neighbours, considering two variations: the first one has only VoIP data flows and the second has VoIP and BE data flows. We measured the packet loss and the delays of VoIP calls and evaluated the MOS (Mean Opinion Score) measure indicating the user s satisfaction. In the first scenario, without CAC-RM, when the number of VOIP calls increases, their quality are decreased (indicated by the MOS value measured). Using the proposed calls admission control some calls were rejected and the gains is about 70% in the MOS value measured. Besides there was a reduction up to 82% and 96%, respectively, in packets losses and delays. In the second variation the proposed CAC to avoid reduction to about 29% of VoIP with quality over 3.5, and reduces too the damagethe packet VoIP losses uppon 86 %. In other scenario using multiple hops, we used VoIP traffic and hop count, ETX and ML metrics. When the network is saturated, there are no calls with enough VoIP quality. But it s possible to get satisfatory VoIP calls rejecting some calls as done by the proposed CAC. This gives a gain of MOS and reduces to 75 % the packet loss. In both scenarios, the proposed VoIP call admission control has provem efficient avoiding admission of new VoIP calls would degrade the quality of already established connections and ensuring that BE traffic does not damage VoIP packets on the wireless mesh network. / Esta dissertação apresenta um controle de admissão para chamadas VoIP em Rede Mesh sem fio, denominado CAC-RM,que utiliza o protocolo IEEE 802.11g. Seu uso visa prevenir que a chegada de novas chamadas VoIP na rede consuma excessivamente recursos dos nós e causem degradação nos fluxos já estabelecidos, devido ao congestionamento do meio sem fio. Em nossa pesquisa, o controle de admissão proposto só aceitará novas chamadas VoIP caso a qualidade das chamadas em andamento não se torne insatisafatória. As principais características de nossa proposta são: (i) a estimativa do tempo de ocupação do meio com tráfego VoIP e tráfego que não necessita de requisitos de qualidade, denominado BE (Best effort); (ii) a reserva de recursos para tráfego VoIP; (iii) a integração do controle de admissão ao protocolo de roteamento pró-ativo OLSR (usando o envio de mensagens de controle com informações sobre o tempo de ocupação do nó com tráfego BE e VoIP); (iv) a estimativa sobre a interferência intra-fluxo integrada ao controle de admissão; (v) regulagem do tráfego BE; e (vi) a priorização do envio e recebimento do tráfego VoIP. Inicialmente, o controle de admissão proposto foi avaliado a partir de simulações de uma rede mesh sem fio em um cenário em que todos os nós são vizinhos e considerando duas variações: na primeira há somente tráfego VoIP e na segunda, tráfego VoIP e BE. Foram medidos e avaliados a perda e atrasos de pacotes das chamadas VoIP, assim como os valores do MOS (Mean Opinion Score), que indica a satisfação do usuário. No cenário em que os nós vizinhos só trafegam VoIP e não é utilizado o CAC-RM, à medida que a quantidade de chamadas VoIP cresce, sua qualidade é reduzida até que se torna inviável o uso do VoIP na rede. Com o uso do CAC-RM, obteve-se ganhos de até 70% no valor do MOS medido a partir da rejeição de algumas chamadas. Além disso, houve redução de até 82% e 96%, respectivamente, nas perdas de pacotes e atrasos. Já no cenário em que há tráfego VoIP e BE, o CAC proposto conseguiu evitar redução de, aproximadamenre, 29% das chamadas com qualidade, além de conseguir reduzir em até 86% as perdas de pacotes VoIP. No cenário com múltiplos saltos foram transmitidas chamadas VoIP na presença das métricas de roteamento Contagem de salto, ETX e ML. Sem o uso do CAC-RM e com tráfego intenso de dados, não há chamadas VoIP com qualidade satisfatória quando são necessários 02 saltos. A partir da rejeição de algumas chamadas feitas pelo CAC proposto, é possível obter chamadas VoIP com qualidade, contabilizando ganho no MOS, com redução de até 75% nas perdas de pacote. Em ambos os cenários simulados, o controle de admissão de chamadas proposto, mostrou-se eficiente ao evitar que a entrada de novas chamadas VoIP comprometesse o desempenho de chamadas já estabelecidas e que a existência de tráfego BE inviabilizasse o uso do VoIP na rede mesh sem fio.

Mobile telephony technology like GSM made portable telephony a possibility. The arising and development of the Internet made a revolutionary change to communication and interchange of information. Bluetooth wireless technology revolutionizes personal connectivity by providing freedom from wired connections. Combining these technologies together brings the concept of Mobile@Home of Ericsson. Mobile@Home is a fixed-mobile convergence concept using the fixed network to carry present and future mobile services (e.g. voice, video, mail and Internet access) all the way to the home or office. By combining the high bandwidth of the fixed access network with the wireless technology of Bluetooth, Mobile@Home makes it possible to deliver high bandwidth to the mobile phone. Mobile@Home requires a Bluetooth enabled mobile phone and a Bluetooth enabled HBS (Home Base Station), placed at the home or office. By means of fast IP access (ADSL, cable modem etc.) the HBS connects into the standard mobile core network through a HBSC (Home Base Station Controller). The purpose of this thesis is the generation of simulated traffic between the HBS and HBSC and to analyze its behavior. This primary involves generation of signaling through an internal protocol, provided by Ericsson, for management and call control, and generation of GSM EFR (Enhanced Full Rate) voice streams over the RTP (Real Time Protocol) protocol. The simulation will consist of both the HBS and MS (Mobile Station). A set of HBS: s with attached MS will call one another through the HBSC. In this assignment only the GSM signaling will be considered because of time and scope limitations. The goal is to validate the RTP traffic generated towards the HBSC. Parameters like packet loss, packet delay and erroneous packets will be analyzed.

Videoconferencing is one of the most common telepresence methods today and educational videos is rising in popularity among distance learners. Traditional videoconferencing is unable to convey gestures and mutual eye contact between participants. This study aim to propose a Virtual Reality telepresence solution using game engines. A literature study confirmed the effectiveness achieved in VR is comparable to the effectiveness in face-to-face meetings. The suggested solution implements whiteboard functionality from a real-life perspective, confirming it is possible to include new functionality and directly transfer old functionality to the VR system from the communication systems today. The system was evaluated based on the response time, packet loss, bandwidth, frame rate and through user tests. The evaluation shows it is possible to design a telepresence system with VR capable of passing the Turing Test for Telepresence. The participants of the user tests did not experience discomfort and they were positively inclined to the telepresence system. Though, discomfort may emerge if the VR character is used with a common office workstation. Future studies in this topic would involve modifications of the third person camera, making the head's rotation follow the direction of the camera view and implementing movable eye pupils on the VR character using the upcoming eye-tracking accessory.

Mobile telecommunications belong to one of the fields which still evolves and it is expected that in the future they are going to offer great benefit to the whole world. This work focuses on current and future development of mobile Internet connection in the Czech Republic. At first the technologies used to provide mobile Internet connection are described and compared and then the situation (in terms of technologies and mobile operators) in the Czech market is outlined and its possible evolution in the near future is presented. Furthermore, the tarifs of mobile operators are compared with those in place two years ago. The strengths and weaknesses of the current mobile Internet connection are evaluated based on the conducted questionnaire survey. The latest results are then analyzed and compared with the results from a previous survey conducted in 2007 as part of a bachelor's thesis, which has been further extended in this mater's thesis.