La téléphonie sur IP est devenue un nouveau paradigme pour établir et transmettre les com- munications téléphoniques directement sur les réseaux IP de manière flexible et à faible coût. Toutefois, les services VoIP sont confrontés à plusieurs problèmes de sécurité qui sont soit hérités de la couche IP soit spécifiques au service lui-même. Une grande variété de mécanismes de protection sont disponibles pour y faire face. Cependant, ces services nécessitent des performances et une disponibilité du réseau élevées, et les mécanismes de protection peuvent nuire à ces perfor- mances. La gestion des risques offre de nouvelles perspectives à l'égard de cette problématique. Nos contributions portent sur l'application et l'automatisation de la gestion de risques dans les infrastructures VoIP selon trois axes. Le première axe porte sur l'automatisation du processus de gestion des risques dans un réseau VoIP d'entreprise. Dans ce cadre, nous avons développé un modèle pour évaluer les risques, un ensemble de contremesures progressives et des algorithmes de mitigation. Nous l'avons couplé à un système de détection d'anomalies basé sur les SVM et un mécanisme d'auto-configuration qui peut fournir un retour d'expérience sur l'efficacité des contremesures. Le deuxième axe concerne l'extension de notre stratégie dans les réseaux P2PSIP. Nous avons mis en place une solution adaptée à la nature distribuée des environnements pair- à-pair. Nous nous sommes aussi intéressés à l'architecture RELOAD et avons étudié comment traiter les attaques résiduelles à travers des mécanismes de confiance. Nous avons enfin étudié les services VoIP dans le cloud où nous proposons plusieurs stratégies pour le déploiement et l'application des contremesures.

Internetas bei jo paslaugos – neatskiriama šiuolaikinės visuomenės dalis. Šiuo metu sunkiai įsivaizduojamas yra kompanijų darbas – be elektroninės bankininkystės, deklaracijų, prekybos sistemų ir pan. Viena iš tokių paslaugų yra sparčiai besivystanti internetinė telefonija (IP telefonija) arba VoIP (Voice Over IP – angl.). Esminis ir pagrindinis šios komunikavimo paslaugos pranašumas prieš tradicines priemones – kaina. Tarptautiniai IP telefonijos pokalbių tarifai jau senokai pigumu viršija tradicinės telefonijos įkainius , o neretai šie pokalbiai yra visiškai nemokami. Nepaisant ekonomiškumo, IP telefonijos srityje kyla taip pat nemažai problemų. Kadangi balso informacija yra perduodama paketiniu būdu (SIP, H.323 protokolai) per IP tinklus ir neturi atskiro kanalo duomenų perdavimui, neretai susiduriama su tam tikromis kokybės problemomis. Svarbiausios jų – uždelsimas(vėlinimas) bei paketų praradimas. Šio darbo esmė – IP telefonijos kanalu perduoto balso įrašo kokybės tyrimas, originalaus bei perduoto įrašo kokybiniai skirtumai, todėl darbo problema yra – balso, perduoto IP telefonijos kanalais kokybės nuostoliai. Tyrimo metu bus matuojami kokybės nuostoliai. Šio darbo objektas – IP kanalu perduoto balso kokybė. Darbo tikslas – ištirti bei įvertinti IP telefonijos kanalais perduoto balso kokybę ir nustatyti, ar perduotas balso įrašas yra tinkamas naudoti balso atpažinimo sistemose. Uždaviniai tikslui pasiekti: • Išsiaiškinti IP telefonijos veikimo principus, technologijas; •... [toliau žr. visą tekstą] / VoIP (voice over IP protocol) is quickly growing and expanding technology. Calls and other services are much cheaper than traditional ones. Nevertheless it meets some quality problems such as packet delay and packet loss. The object of this paper is to explore voice transmitted by IP channels quality and to conclude about if that voice could be used in voice recognition systems. To complete this object, there are 4 main tasks: • Find out the technologies and working techniques of the VoIP. • Determine tools for quality analysis of the transmitted voice • Build an algorithm which could help to extract useful data from voice signal. • Summarize the results and conclude about ability to use transmitted voice in voice recognition systems. There was made 150 transmissions („Skype“ system was used) of the voice during the experiment and all of them was recorded. Fast Fourier transform was used to transform those recording to frequency scale. The quality loss was analyzed; original voice record specter was compared with transmitted voice record specter. The transmissions were divided into two parts. Transmissions was made in 2008 February (1st part) and 2008 May (2nd part). In the first part the results are almost the same, quality variations are stable (specter difference average is 1,44), all values are between 1,44 and 1,55 (specter difference average value). It was noticed that transmitted voice signal has large quality variations in the second part of the experiment. The... [to full text]

The information on the volume of traffic flowing between all possible origin and destination pairs in an IP network during a given period of time is generally referred to as traffic matrix (TM). This information, which is very important for various traffic engineering tasks, is very costly and difficult to obtain on large operational IP network, consequently it is often inferred from readily available link load measurements. In this thesis, we evaluated 5 TM estimation techniques, namely Tomogravity (TG), Entropy Maximization (EM), Quadratic Programming (QP), Linear Programming (LP) and Neural Network (NN) with gravity and worst-case bound (WCB) initial estimates. We found that the EM technique performed best, consistently, in most of our simulations and that the gravity model yielded better initial estimates than the WCB model. A hybrid of these techniques did not result in considerable decrease in estimation errors. We, however, achieved most significant reduction in errors by combining iterative proportionally-fitted estimates with the EM technique. Therefore, we propose this technique as a viable approach for estimating the traffic matrix of large-scale IP networks.

The combination of infrastructure-to-vehicle and vehicle-to-vehicle communications, namely the multi-hop Vehicular Communications Network (VCN) , appears as a promising solution for the ubiquitous access to IP services in vehicular environments. In this thesis, we address the challenges of multi-hop VCN, and investigate the seamless provision of IP services over such network. Three different schemes are proposed and analyzed. First, we study the limitations of current standards for the provision of IP services, such as 802.11p/WAVE, and propose a framework that enables multi-hop communications and a robust IP mobility mechanism over WAVE. An accurate analytical model is developed to evaluate the throughput performance, and to determine the feasibility of the deployment of IP-based services in 802.11p/WAVE networks. Next, the IP mobility support is extended to asymmetric multi-hop VCN. The proposed IP mobility and routing mechanisms react to the asymmetric links, and also employ geographic location and road traffic information to enable predictive handovers. Moreover, since multi-hop communications suffer from security threats, it ensures that all mobility signalling is authenticated among the participant vehicles. Last, we extend our study to a heterogeneous multi-hop VCN, and propose a hybrid scheme that allows for the on-going IP sessions to be transferred along the heterogeneous communications system. The proposed global IP mobility scheme focuses on urban vehicular scenarios, and enables seamless communications for in-vehicle networks, commuters, and pedestrians. The overall performance of IP applications over multi-hop VCN are improved substantially by the proposed schemes. This is demonstrated by means of analytical evaluations, as well as extensive simulations that are carried out in realistic highway and urban vehicular scenarios. More importantly, we believe that our dissertation provides useful analytical tools, for evaluating the throughput and delay performance of IP applications in multi-hop vehicular environments. In addition, we provide a set of practical and efficient solutions for the seamless support of IP tra c along the heterogeneous and multi-hop vehicular network, which will help on achieving ubiquitous drive-thru Internet, and infotainment traffic access in both urban and highway scenarios.

IP multicasting is a method for transmitting the same information to multiple receivers over IP networks. Reliability issue of multicasting contains the challenges for detection and recovery of packet losses and ordered delivery of the entire data. In this work, existing reliable multicast protocols are classified into three main groups, namely tree based, NACK-only and router assisted, and a representative protocol for each group is selected to demonstrate the advantages and disadvantages of the corresponding approaches. The selected protocols are SRM, PGM and RMTP. Performance characteristics of these protocols are empirically evaluated by using simulation results. Network Simulator-2 (ns2), a discrete event simulator is used for the implementation and simulation of the selected protocols. The contributions of the thesis are twofold, i.e. the extension of the ns library with an open source implementation of RMTP which did not exist earlier and the evaluation of the selected protocols by investigating performance metrics like distribution delay and recovery latency with respect to varying multicast group size, network diameter, link loss rate, etc.

Despite the persistent change and growth that characterizes the Internet, the Transmission Control Protocol (TCP) still dominates at the transport layer, carrying more than 90\% of the global traffic. Despite its astonishing success, it has been observed that TCP can cause poor end-to-end performance, especially for large transfers and in network paths with high bandwidth-delay product. In this thesis, we focus on mechanisms that can address key problems in TCP performance, without any modification in the protocol itself. This evolutionary approach is important in practice, as the deployment of clean-slate transport protocols in the Internet has been proved to be extremely difficult. Specifically, we identify a number of TCP-related problems that can cause poor end-to-end performance. These problems include poorly dimensioned socket buffer sizes at the end-hosts, suboptimal buffer sizing at routers and switches, and congestion unresponsive TCP traffic aggregates. We propose solutions that can address these issues, without any modification to TCP. <br> <br> In network paths with significant available bandwidth, increasing the TCP window till observing loss can result in much lower throughput than the path's available bandwidth. We show that changes in TCP are {em not required} to utilize all the available bandwidth, and propose the application-layer SOcket Buffer Auto-Sizing (SOBAS) mechanism to achieve this goal. SOBAS relies on run-time estimation of the round trip time (RTT) and receive rate, and limits its socket buffer size when the receive rate approaches the path's available bandwidth. In a congested network, SOBAS does not limit its socket buffer size. Our experiment results show that SOBAS improves TCP throughput in uncongested network without hurting TCP performance in congested networks. <br> <br> Improper router buffer sizing can also result in poor TCP throughput. Previous research in router buffer sizing focused on network performance metrics such as link utilization or loss rate. Instead, we focus on the impact of buffer sizing on end-to-end TCP performance. We find that the router buffer size that optimizes TCP throughput is largely determined by the link's output to input capacity ratio. If that ratio is larger than one, the loss rate drops exponentially with the buffer size and the optimal buffer size is close to zero. Otherwise, if the output to input capacity ratio is lower than one, the loss rate follows a power-law reduction with the buffer size and significant buffering is needed. The amount of buffering required in this case depends on whether most flows end in the slow-start phase or in the congestion avoidance phase. <br> <br> TCP throughput also depends on whether the cross-traffic reduces its send rate upon congestion. We define this cross-traffic property as {em congestion responsiveness}. Since the majority of Internet traffic uses TCP, which reduces its send rate upon congestion, an aggregate of many TCP flows is believed to be congestion responsive. Here, we show that the congestion responsiveness of aggregate traffic also depends on the flow arrival process. If the flow arrival process follows an open-loop model, then even if the traffic consists exclusively of TCP transfers, the aggregate traffic can still be unresponsive to congestion. TCP flows that arrive in the network in a closed-loop manner are always congestion responsive, on the other hand. We also propose a scheme to estimate the fraction of traffic that follows the closed-loop model in a given link, and give practical guidelines to increase that fraction with simple application-layer modifications.

The following thesis focuses on two areas of chemistry, pi-pi interactions and radical cation dimers. Approximations to the exact solution to the Schrodinger equation are investigated for these types of chemical systems with a variety of theoretical methods. The first chapter provides an introduction to the various quatum mechanical methods used in this research. The second chapter focuses specifically on pi-pi interaction. In this chapter, high quality quantum mechanical methods are used to examine how substituents tune pi-pi interactions between monosubstituted benzene dimers in parallel-displaced geometries. In addition, the role of dispersion and coulombic interactions in these systems is investigated to determine the nature of the substituent effect. In the third chapter radical cation dimers are investigated. Benchmark results with full configuration interaction (FCI) and equation-of-motion coupled-cluster for ionized systems (EOM-IP-CCSD) are presented for prototypical charge transfer species. Conclusions regarding chapters 2 and 3 are presented in the final chapter. This work may form the basis for improved approaches to rational drug design, organic optical materials, and molecular electronics.

Computer networks used in Telecommunication, particularly the Internet, have been used to carry computer data only, but now they carry voice and/or video also. Because each type of this traffic has specific flow characteristics, each type has to be handled with a certain level of guaranteed quality. So based on that, IETF introduced a Quality of Service tool, called Differentiated Service. It offers different levels of service to different classes of traffic. Even then, the problem of congestion arises due to sharing of a finite bandwith. In the case of real time multi media traffic, congestion due to inadequate bandwith contributes heavily to the quality, whereas in non-real time traffic the effect of congestion is to make data transfer take a longer time. In contrast, real time data become become obsolete if they do not arrive on time. Therefore what is needed is some way of ensuring that during periods of congestion, real time traffic is not affected at all, or is at least given a higher priority than non-real time. / The motivation for the research that has been carried out was therefore to develop a rule based traffic management scheme for DiffServ networks with a view to introducing QoS (Quality of Service). This required definition of rules for congestion management/control based on the type and nature of IP traffic encountered, and then constructing and storing these rules to enable future access for application and enforcement. We first developed the required rule base and then developed the software based mobile agents using the Java (RMI) application package, for accessing these rules for application and enforcement. Consequently, these mobile agents act as smart traffic managers at nodes/routers in the computer based communication network and manage congestion. The rule base as well as the mobile agent software developed in Java (RMI), were validated using computer simulation. The contents of the research carried out have been presented in the thesis. / Thesis (PhD)--University of South Australia, 2006.

Thesis (PhDTelecommunications)--University of South Australia, 2002.

In recent times, the imminent lack of public IPv4 addresses has attracted the attention of both the research community and industry. The cellular industry has decided to combat this problem by using IPv6 for all new terminals. However, the success of 3G network deployment will depend on the services offered to end users. Currently, almost all services reside in the IPv4 address space, making them inaccessible to users in IPv6 networks. Thus, an intermediate translation mechanism is required. Previous studies on network address translation methods have shown that Realm Base Kluge Address Heuristic-IP, REBEKAH-IP supports all types of services that can be offered to IPv6 hosts from the public IPv4 based Internet, and provides excellent scalability. However, the method suffers from an ambiguity problem which may lead to call blocking. This thesis presents an improvement to REBEKAH-IP scheme in which the side effect is removed, creating a robust and fully scalable system. The improvement can be divided into two major tasks including a full investigation on the scalability of addressing and improvements to the REBEKAH-IP scheme that allow it to support important features such as ICMP and IP mobility. To address the first task a method called REBEKAH-IP with Port Extension (RPX) is introduced. RPX is extended from the original REBEKAH-IP scheme to incorporate centralised management of both IP address and port numbers. This method overcomes the ambiguity problem, and improves scalability. We propose a priority queue algorithm to further increase scalability. Finally, we present extensive simulation results on the practical scalability of RPX with different traffic compositions, to provide a guideline of the expected scalability in large-scale networks. The second task concerns enabling IP based communication. Firstly, we propose an ICMP translation mechanism which allows the RPX server to support important end-toend control functions. Secondly, we extend the RPX scheme with a mobility support scheme based on Mobile IP. In addition, we have augmented Mobile IP with a new tunneling mechanism called IP-in-FQDN tunneling. The mechanism allows for unique mapping despite the sharing of IP addresses while maintaining the scalability of RPX. We examine the viability of our design through our experimental implementation.