No / Over the past decade, wireless communications has experienced tremendous growth, and this growth is likely to multiply in the near future. The proliferation of mobile users and an ever increasing demand for multimedia services has resulted in greater capacity requirements. Radio frequency spectrum is scarce and cannot meet this ever increasing demand and the required Quality of Service (QoS) will no longer be achieved if efficient Radio Resource Management (RRM) solutions are not found. Conventional Radio Access Networks (RAN) have standalone Base Stations (BS) with capacity preconfigured for peak loads. These RANs have high call blocking and dropping rates since BSs resources cannot be shared. Cloud based RANs (C-RAN) have been proposed as a cost and energy efficient way of meeting high capacity demand of future wireless access networks by consolidating BSs to the cloud. Instead of relying on rejection of new call requests due to limited BS resources, C-RAN takes benefit of the cloud elasticity, which allows dynamic provisioning of cloud BS resources. This paper presents a novel C-RAN Call Admission Control (C-RAN CAC) to ensure Grade of Service (GoS) by improving blocking probability and improvement of call waiting times. Call blocking probability, call average waiting time and system utilization are used to evaluate the performance of the proposed CAC algorithm.

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In the next generation of networks we will begin to see the true convergence of voice, multimedia, and data traffic. This merging of various dedicated networks will occur both in the wired and wireless domains. Given the growth in the areas of wireless voice and data, we see that the combination of mobile and Internet communication constitutes the driving force behind the third-generation wireless system and makes the basis for the fourth-generation wireless system. For services like voice over IP over wireless (VoIPoW), the main challenge is to achieve QoS and spectrum efficiency. In order to support better QoS the IETF Mobile IP Working Group is discussing a number of enhancements to the base protocol to reduce the latency, packet loss and signaling overhead experienced during handoff. This support also includes both the call admission and the subsequent scheduling of packet transmissions. In this thesis, we will first survey the work done on issues related to QoS provisioning for wireless network and then will address bandwidth allocation problem in packet radio network with special emphasis on wireline to wireless internetworking zone. The main aim of the thesis is to evolve a strategy to reduce the call dropping probability by negotiating the QoS in those conditions when we do not have the sufficient resources (mainly bandwidth) to allocate. In order to achieve the QoS we have investigated the behaviour of the Real-time Voice traffic on a wireless link and its relation to the associated quality of service. This investigation opens a way for QoS negotiation, in a condition like during handoff, when the network is not able to sustain the negotiated bandwidth. The main results of this work are, that even with reduced bandwidth, quality for speech can be maintained at a reasonable level and this way the call dropping can be reduced. Such a scheme is useful in those conditions when we do not have the sufficient bandwidth to allocate like during a handoff of a mobile host from one cell to another. Moreover the bandwidth is a scarce resource in wireless domain so there should be an efficient call admission control policy. Many call admission control policies are proposed in the literature; here we propose a simple scheme for real-time traffic, specially speech, in a base station which increases the system throughput. In addition to above, we have also experimented with Cellular IP, one of the implementations of proposed micro-mobility architecture to provide faster handoff and seamless mobility in wired and wireless network.

Les systèmes de communications mobiles ont considérablement évolué durant ces dernières années. Ce fait a encouragé le déploiement de plusieurs systèmes ou réseaux cellulaires multi technologique. La qualité de service (QoS) offerte aux utilisateurs mobiles s’améliore d’un système à l’autre. Les systèmes de troisième génération (3G), comme l’UMTS, offrent une meilleure qualité de service par rapport à celle offerte par ceux de deuxième génération (2G), comme le GSM. Pour exemple, les systèmes de 3.5G (HSDPA) améliorent le débit du réseau de 3G sur le lien descendant afin de répondre aux exigences des nouveaux services. De plus, les réseaux de quatrième génération (4G), comme le WiMAX (IEEE802.16e) permettent, quant à eux, d’élargir la couverture cellulaire tout en offrant un débit supérieur. Cette nouvelle évolution améliore encore l’accessibilité aux services de l’Internet. La migration des services de l’Internet vers les réseaux mobiles, constitue un enjeu majeur de recherche en télécommunications. La thèse se divise en deux objectifs principaux ; le premier consiste à calculer et à évaluer les performances du système UMTS-WCDMA. Deux approches sont utilisées, l’approche exacte de l’analyse spectrale, et l’approche approximative de la perturbation singulière. Le deuxième objectif propose de nouveaux mécanismes de contrôle d’admission prenant en considération la mobilité intra et inter cellulaires des utilisateurs dans la gestion de la ressource radio. Ces mécanismes d’allocation de ressource concernent plus particulièrement les réseaux HSDPA et WiMAX. Ils se basent principalement sur les schémas de modulation et de codage adaptatif AMC (Adaptive Modulation and Coding) : la zone géographique est subdivisée en plusieurs régions concentriques. Avec nos mécanismes d’allocation des ressources, chaque appel accepté par le système voit son débit initial maintenu, et ce, indépendamment de la position de l’utilisateur dans la cellule / The mobile communication systems evolved considerably in the last few years. This fact encouraged the deployment of several systems or cellular networks in multi technological environment. The Quality of Service (QoS) offered to the mobile users improves from one system to another one. The systems of third generation (3G), like UMTS, offer a better quality of service compared to that offered by those of second generation (2G), like the GSM. For example, the systems of 3.5G (HSDPA) improve the throughput of the network of 3G in the downlink direction according to the requirements of the new services. Moreover, the networks of fourth generation (4G), like WiMax (IEEE802.16e), as for them, make it possible to widen the cover of the base station while offering a very important throughput in which supports the next generation applications or services already offered by the Internet. This new evolution still improves accessibility with the services of the Internet. The migration of the services of the Internet towards the mobile networks, constitutes a major stake of research in telecommunications. The thesis is divided into twomain aims. The first consists to compute the systemcapacity and to evaluate the performances of the UMTS-WCDMA system. Two approaches are used, one is of the spectral analysis, and other one is an approximation of the singular perturbation. The second goal of this thesis is to propose and develop the new mechanisms of call admission control (CAC) for HSDPA and IEEE802.16e networks based on the AMC (Adaptive Modulation and Coding) scheme : the geographical area of the cell is subdivided into several concentric regions. These CAC mechanisms maintain a same QoS (constant bit rate) for all users anywhere in the area of the cell and give priority to migrating or handoff calls by reserving the guard bandwidth than a new call in the intra cell and inter cell mobilities

The next generation wireless system (NGWS) has been conceived as a ubiquitous wireless environment. It integrates existing heterogeneous access networks, as well as future networks, and will offer high speed data, real-time applications (e.g. Voice over IP, videoconference ) and real-time multimedia (e.g. real-time audio and video) support with a certain Quality of Service (QoS) level to mobile users. It is required that the mobile nodes have the capability of selecting services that are offered by each provider and determining the best path through the various networks. Efficient radio resource management (RRM) is one of the key issues required to support global roaming of the mobile users among different network architectures of the NGWS and a precise call admission control (CAC) scheme satisfies the requirements of high network utilization, cost reduction, minimum handover latency and high-level QoS of all the connections. This thesis is going to describe an adaptive class-based CAC algorithm, which is expected to prioritize the arriving channel resource requests, based on user¿s classification and channel allocation policy. The proposed CAC algorithm couples with Fuzzy Logic (FL) and Pre-emptive Resume (PR) theories to manage and improve the performance of the integrated wireless network system. The novel algorithm is assessed using a mathematical analytic method to measure the performance by evaluating the handover dropping probability and the system utilization.

One of the enabling technologies for Long Term Evolution (LTE) deployments is the femtocell technology. By having femtocells deployed indoors and closer to the user, high data rate services can be provided efficiently. These femtocells are expected to be depolyed in large numbers which raises many technical challenges including the handover management. In fact, managing handovers in femtocell environments, with the conventional manual adjustment techniques, is almost impossible to keep pace with in such a rapidly growing femtocell environment. Therefore, doing this automatically by implementing Self Organizing Network (SON) use cases becomes a necessity rather than an option. However, having multiple SON use cases operating simultaneously with a shared objective could cause them to interact either negatively or positively. In both cases, designing a suitable coordination policy is critical in solving negative conflicts and building upon positive benefits. In this work, we focus on studying the interactions between three self optimization use cases aiming at improving the overall handover procedure in LTE femtocell networks. These self optimization use cases are handover, Call Admission Control (CAC) and load balancing. We develop a comprehensive, unified LTE compliant evaluation environment. This environment is extendable to other radio access technologies including LTE-Advanced (LTE-A), and can also be used to study other SON use cases. Various recommendations made by main bodies in the area of femtocells are considered including the Small Cell Forum, the Next Generation Mobile Networks (NGMN) alliance and the 3rd Generation Partnership Project (3GPP). Additionally, traffic sources are simulated in compliance with these recommendations and evaluation methodologies. We study the interaction between three representative handover related self optimization schemes. We start by testing these schemes separately, in order to make sure that they meet their individual goals, and then their mutual interactions when operating simultaneously. Based on these experiments, we recommend several guidelines that can help mobile network operators and researchers in designing better coordination policies. / Thesis (Master, Electrical & Computer Engineering) -- Queen's University, 2012-12-05 22:35:27.538

O crescente aumento da demanda de tráfego nas redes celulares vem aumentando a necessidade de uma melhor utilização dos recursos do sistema, já que sua expansão é custosa. Nas estações rádio base (ERB), a disponibilidade de largura de faixa de freqüências é limitada e desta maneira, em uma rede de comunicação móvel celular, o controle de admissão de chamadas exerce grande influência no desempenho do sistema, pois determina a utilização de banda das ERBs e se uma determinada quantidade de recursos (banda) será alocado ou não para uma determinada chamada. O desempenho da rede pode ser atrelado a determinados parâmetros, como a probabilidade de bloqueio de novas chamadas, probabilidade de bloqueio de chamadas handoff e a utilização de banda da rede. Este trabalho propõe um controle de admissão de chamadas que, no atendimento de uma chamada, faz o empréstimo de banda de chamadas em andamento na célula no caso de banda insuficiente. O sistema adota um mecanismo heurístico que determina a banda disponível para novas chamadas conforme os valores de certos parâmetros do sistema. O empréstimo de banda é realizado em chamadas em andamento nas células até níveis mínimos estabelecidos para cada tipo de chamada, que se diferenciam pelas necessidades de banda de cada uma. O algoritmo foi aplicado às bandas e características de uma rede de terceira geração (3G), que possui chamadas de voz, videoconferência, interação multimídia, e-mail, downloads e transferência de arquivos e a uma rede GSM/GPRS (global system for mobile communications/ general packet radio service), que possui chamadas de voz e de dados. Os resultados mostram melhorias na probabilidade de bloqueio de novas chamadas, probabilidade de bloqueio de handoff e na utilização de banda do sistema. / The recent growth in traffic loads in cellular networks has seen the need for a better use of system resources as its expansion is expensive. In the base transceiver station (BTS), the bandwidth availability is limited. Thus, in cellular networks the call admission control greatly influences the system performance because it determines the bandwidth use of the BTSs and if an amount of resources will or will not be allocated to a call. The network performance can be evaluated by parameters such as blocking probability of new calls, dropping probability of handoff calls and bandwidth use. This work proposes a call admission control that carries out the bandwidth borrowing when a call arrives and there is not enough bandwidth. The system makes use of a heuristic mechanism that determines the available bandwidth for the new calls according to some parameter values of the system. The bandwidth borrowing is applied to the cell ongoing calls until the minimum levels for each type are met. The algorithm was applied to the bandwidths and characteristics of a third generation cellular network, which supports voice calls, videoconference, multimedia interaction, e-mails, downloads and file transfers. It was also applied to a GSM/GPRS (global system for mobile communications/ general packet radio service), which supports voice and data calls. The results show improvements in the blocking probability of new calls, dropping probability of handoff calls and in the bandwidth use of the system.

O crescente aumento da demanda de tráfego nas redes celulares vem aumentando a necessidade de uma melhor utilização dos recursos do sistema, já que sua expansão é custosa. Nas estações rádio base (ERB), a disponibilidade de largura de faixa de freqüências é limitada e desta maneira, em uma rede de comunicação móvel celular, o controle de admissão de chamadas exerce grande influência no desempenho do sistema, pois determina a utilização de banda das ERBs e se uma determinada quantidade de recursos (banda) será alocado ou não para uma determinada chamada. O desempenho da rede pode ser atrelado a determinados parâmetros, como a probabilidade de bloqueio de novas chamadas, probabilidade de bloqueio de chamadas handoff e a utilização de banda da rede. Este trabalho propõe um controle de admissão de chamadas que, no atendimento de uma chamada, faz o empréstimo de banda de chamadas em andamento na célula no caso de banda insuficiente. O sistema adota um mecanismo heurístico que determina a banda disponível para novas chamadas conforme os valores de certos parâmetros do sistema. O empréstimo de banda é realizado em chamadas em andamento nas células até níveis mínimos estabelecidos para cada tipo de chamada, que se diferenciam pelas necessidades de banda de cada uma. O algoritmo foi aplicado às bandas e características de uma rede de terceira geração (3G), que possui chamadas de voz, videoconferência, interação multimídia, e-mail, downloads e transferência de arquivos e a uma rede GSM/GPRS (global system for mobile communications/ general packet radio service), que possui chamadas de voz e de dados. Os resultados mostram melhorias na probabilidade de bloqueio de novas chamadas, probabilidade de bloqueio de handoff e na utilização de banda do sistema. / The recent growth in traffic loads in cellular networks has seen the need for a better use of system resources as its expansion is expensive. In the base transceiver station (BTS), the bandwidth availability is limited. Thus, in cellular networks the call admission control greatly influences the system performance because it determines the bandwidth use of the BTSs and if an amount of resources will or will not be allocated to a call. The network performance can be evaluated by parameters such as blocking probability of new calls, dropping probability of handoff calls and bandwidth use. This work proposes a call admission control that carries out the bandwidth borrowing when a call arrives and there is not enough bandwidth. The system makes use of a heuristic mechanism that determines the available bandwidth for the new calls according to some parameter values of the system. The bandwidth borrowing is applied to the cell ongoing calls until the minimum levels for each type are met. The algorithm was applied to the bandwidths and characteristics of a third generation cellular network, which supports voice calls, videoconference, multimedia interaction, e-mails, downloads and file transfers. It was also applied to a GSM/GPRS (global system for mobile communications/ general packet radio service), which supports voice and data calls. The results show improvements in the blocking probability of new calls, dropping probability of handoff calls and in the bandwidth use of the system.

In IEEE 802.11 WLANs, Link Adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bitrates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. In this thesis, a VoIP codec adaptation algorithm is proposed as a solution, based on a cross-layer feedback from RTCP packets and the MAC layer, which can adapt the codecs of active calls to adjust them to the multirate scenario. A combination of this algorithm with a call admission control mechanism is also studied. The results show an important improvement in terms of the QoS of the already active flows as also in the total hotspot's capacity. Additionally, by defining a new Grade of Service related parameter, the Q-Factor, which captures the trade-off between dropping and blocking ratio and perceived speech quality, the codec adaptation algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements. Finally, a new QoS-enabled AP, which implements these enhancements is designed. / En las redes inalámbricas del estándar IEEE 802.11, los mecanismos de adaptación de enlace que eligen la tasa de transmisión de cada nodo, pueden provocar variaciones aleatorias e inesperadas en la capacidad efectiva del canal. Cuando estos cambios son hacia tasas de transmisión mas bajas, los flujos inelásticos, tales como los de VoIP, pueden de repente sufrir congestión, lo que se traduce en aumento de retrasos y pérdidas de paquetes. En esa tesis, se propone un algoritmo de adaptación de codificadores de voz como solución, basado en técnicas multinivel (cross-layer) que combinan el uso de información de diferentes capas, como los paquetes RTCP y la capa MAC, y que puede adaptar los codecs de las llamadas activas para ajustarlos al escenario "multi-rate". Adicionalmente, la combinación de este algoritmo con un mecanismo de control de admisión de llamadas (CAC) se ha estudiado. Los resultados muestran una importante mejora en términos de QoS de los flujos activos como también en la capacidad total del hotspot. Además, mediante la definición de un nuevo factor, el Q-Factor, que puede captar la compensación entre la tasa de corte y de bloqueo de llamadas y de la calidad percibida por esas, el algoritmo de adaptación de codecs se puede ajustar para lograr la máxima capacidad sin penalizar severamente ninguna de esas variables y así satisfacer los requisitos técnicos de calidad y los usuarios. Por último, un nuevo punto de acceso (AP) habilitado para ofrecer calidad de servicio, ha sido diseñado que lleva a cabo estas mejoras.