Telephony plays a fundamental role in our society. It enables remote parties to interact and express themselves over great distances. The telephone as a means of communicating has become part of every day life. Organisations and industry are now looking at Voice over IP (VoIP) technologies. They want to take advantage of new and previously unavailable voice services. Various interested parties are seeking to leverage the emerging VoIP technology for more flexible and efficient communication between staff, clients and partners. <o>VoIP is a recent innovation enabled by Next Generation Network (NGN). It provides and enables means of communication over a digital network, specifically the Internet. VoIP is gaining wide spread adoption and will ultimately replace traditional telephony. The result of this trend is a ubiquitous, global and digital communication infrastructure. VoIP, however, still faces many challenges. It is not yet as reliable and dependable as the current Public Switched Telephone Network (PSTN). The employed communication protocols are immature with many security flaws and weaknesses. Session Initiation Protocol (SIP), a popular VoIP protocol does not sufficiently protect a users privacy. A user’s information is neither encrypted nor secured when calling a remote party. There is a lack of control over the information included in the SIP messages. Our specific concern is that private and sensitive information is exchanged over the public internet. This dissertation concerns itself with the communication path chosen by SIP when establishing a session with a remote party. In SIP, VoIP calls are established over unknown and untrusted intermediaries to reach the desired party. We analyse the SIP headers to determine the information leakage at each chosen intermediary. Our concerns for possible breach of privacy when using SIP were confirmed by the findings. A user’s privacy can be compromised through the extraction of explicit private details reflected in SIP headers. It is further possible to profile the user and determine communication habits from implicit time, location and device information. Our research proposes enhancements to SIP. Each intermediary must digitally sign over the SIP headers ensuring the communication path was not be altered. These signatures are added sequentially creating a chain of certified intermediaries. Our enhancements to SIP do not seek to encrypt the headers, but to use these intermediary signatures to reduce the risk of information leakage. We created a model of our proposed enhancements for attaching signatures at each intermediary. The model also provides a means of identifying unknown or malicious intermediaries prior to establishing a SIP session. Finally, the model was specified in Z notation. The Z specification language was well suited to accurately and precisely represent our model. This formal notation was adopted to specify the types, states and model behaviour. The specification was validated using the Z type-checker ZTC. Copyright / Dissertation (MSc)--University of Pretoria, 2010. / Computer Science / unrestricted

In this graduation thesis wants introduce possibilities of voice transportation in IP networks like Internet. There are described behaviors of IP network, internet connectivity possibilities, voice protocols included Skype and economical outputs. There are mapped positives and negatives of technology including implementation issues. All economic comparisons are made to Telefonica O2 Czech Republic Inc., which is still telecommunication company with majority in the Czech Republic. There are compared samples of VoIP providers with preferences like company live time, lowest cost and guaranteed quality. Description starts with model situation and continues to real case studies. Case studies describe economic calculation, implementation and target analysis. There are three case studies: small company with fast migration in one time, medium company witch accelerated migration in two steps and large company with conservative and control migration in three steps.

La croissance rapide du trafic aérien et les besoins en nouveaux services notamment pour les passagers imposent l'introduction de nouveaux moyens de communication pour les avions avec une bande passante globale fortement accrue. Les satellites sont appelés à jouer un rôle important dans ce contexte, non seulement en complément des systèmes terrestres pour les services « cockpit » (services ATM, Air Traffic Management) mais aussi pour les services « cabine » (In-Flight Entertainment). L'objectif de la thèse est d'étudier l'architecture d'un système satellite supportant l'ensemble de ces services, en se focalisant sur l'architecture du terminal embarqué à bord des aéronefs. L'architecture retenue repose sur des liaisons DVB-S2/DVB-RCS normalisées par l'ETSI. Cette option permet d'utiliser efficacement l'importante bande passante disponible en bande Ka pour les services mobiles aéronautiques (allocation primaire) ou en bande Ku (allocation secondaire). Ces normes ont été conçues pour les applications multimédia (Broadband Satellite Multimedia). Le défi est alors d'utiliser de telles liaisons satellite pour des services aux caractéristiques et besoins fortement hétérogènes. Par ailleurs, l'utilisation de la bande Ka n'est pas concevable sans l'activation de techniques de lutte contre les affaiblissements (FMT – Fade Mitigation Techniques). L'utilisation d'une marge statique conduit à une perte importante de capacité. Les techniques FMT reposent sur une évaluation dynamique du bilan de liaison et permettent une modification de la forme d'onde. Le système utilise ainsi la forme d'onde la plus efficace spectralement pour chaque terminal et maximise la capacité globale du système. Par contre, chaque terminal observe une modification de la ressource allouée au fil du temps. L'objectif de la thèse est de concevoir une architecture au niveau terminal qui permette d'exploiter les liaisons DVB-S2/RCS afin de fournir les services passagers (Internet et téléphonie mobile de type GSM/UMTS) et un canal haute fiabilité pour les services aéronautiques. Deux approches ont été retenues. La première repose sur une application du modèle ETSI BSM (Broadband Satellite Multimedia) en couches séparant strictement les couches dépendantes satellite et les couches indépendantes satellite. Les simulations de cette architecture montrent que les liaisons ne peuvent être utilisées de façon efficace sans une interaction entre couches afin de tenir compte de l'évolution de la capacité disponible. La seconde approche consiste en la concentration de la gestion de la ressource et la gestion de la qualité de service dans la même couche protocolaire. L'idée de départ est d'utiliser la méthode d'encapsulation générique Generic Stream Encapsulation (GSE). GSE a été conçu pour la projection des paquets de couches supérieures à l'intérieur des trames DVB-S2. GSE tient compte de la taille variable des trames DVB-S2 et introduit une capacité de multiplexage entre différents flux (identification de fragments). Sur cette base, une gestion de l'accès est introduite pour gérer la liaison DVB-RCS au format MF-TDMA. Nous introduisons ainsi une utilisation conjointe de GSE, d'une politique de service différentiée et de flux de signalisation inter-couches (« cross-layer »). Les performances des deux approches sont étudiées à l'aide d'un modèle de simulation développé à l'aide du logiciel OPNET Modeler (simulations à événements discrets). Les résultats obtenus démontrent le meilleur comportement de la seconde architecture avec une meilleure utilisation de la ressource et des performances de transmission satisfaisant les objectifs. / The rapid growth of air traffic needs a new communication infrastructure with increased bandwidth, high speed services and applications to satisfy expected air-ground communication requirements. Satellite communication systems play a significant role in this context, not only as a complement to terrestrial systems for Air Traffic Management (ATM) by offering global coverage but also as a promising solution to enrich In-Flight Entertainment (IFE) for passengers. DVB-S2/RCS technology is an attractive proposition to provide the cost-effective broadband services for both ATM and IFE, mainly because a large radio bandwidth is primarily allocated to aeronautical mobile communications in Ka-band, where the open standards DVB are implemented. However, such system design with convergence of heterogeneous traffics involves two main challenges. Firstly, using Ka-band means the implementation of Fade Mitigation Techniques (FMT) in order to cope with deep fades caused by atmospheric attenuation. Otherwise, the waste of capacity would be excessively high in a constant link margin design. FMT adapt in real time the link budget to the propagation conditions, this adaptivity has an impact not only on physical layer but also on upper layers. An efficient ressource management strategy with dynamic bandwidth allocation is required in this case, especially in DVB-RCS return link where FMT are not natives. Secondly, the proposed system must be able to multiplex the trafic flows with highly different characteristics and Quality of Service (QoS) expectations into a single link, the corresponding capacity management and QoS support seem with higher complexity. In this paper, we present an adaptive system design using a single DVB-S2/RCS based satellite link to provide Internet access and mobile telephony (GSM/UMTS) for passengers and a high-reliability channel for ATM. Concerning the airborne terminal architecture, two approaches are investigated. The first one is in compliance with ETSI Broadband Satellite Multimedia (BSM) architecture and based on a layering paradigm. The conducted simulation experiments highlight the need of dynamic interactions and adaptations among the layers to achieve an overall performance optimization. We propose then an enhanced approach with the concentration of both resource allocation and QoS management at the same interface – adaptation layer. The idea comes from the success of the recent Generic Stream Encapsulation (GSE) protocol, which carries the network protocol packets over DVB-S2 forward link in a simple, flexible and efficient way, especially when used with Adaptive Coding and Modulation (ACM). Furthermore, GSE can be easily extended to use in our design for DVB-RCS return link thanks to a proper design of MF-TDMA structure in which the suitable FMT (ACM and Dynamic Rate Adaptation) are context-aware configured. With the combined use of GSE, service policy and the interactions between adaptation and access layers, incoming heterogeneous traffics can be dynamically scheduled, segmented and encapsulated at the same adaptation layer. Performance evaluation of two proposed approaches is derived by a network-level simulation model developed using OPNET. Results prove the enhanced approach outperforms the first one leading to better resource utilization and satisfactory performance.

Voice over IP (VoIP) is a key enabling technology for the migration of circuit-switched PSTN architectures to packet-based IP networks. However, this migration is successful only if the present problems in IP networks are addressed before deploying VoIP infrastructure on a large scale. One of the important issues that the present VoIP networks face is the problem of unwanted calls commonly referred to as SPIT (spam over Internet telephony). Mostly, these SPIT calls are from unknown callers who broadcast unwanted calls. There may be unwanted calls from legitimate and known people too. In this case, the unwantedness depends on social proximity of the communicating parties. For detecting these unwanted calls, I propose a framework that analyzes incoming calls for unwanted behavior. The framework includes a VoIP spam detector (VSD) that analyzes incoming VoIP calls for spam behavior using trust and reputation techniques. The framework also includes a nuisance detector (ND) that proactively infers the nuisance (or reluctance of the end user) to receive incoming calls. This inference is based on past mutual behavior between the calling and the called party (i.e., caller and callee), the callee's presence (mood or state of mind) and tolerance in receiving voice calls from the caller, and the social closeness between the caller and the callee. The VSD and ND learn the behavior of callers over time and estimate the possibility of the call to be unwanted based on predetermined thresholds configured by the callee (or the filter administrators). These threshold values have to be automatically updated for integrating dynamic behavioral changes of the communicating parties. For updating these threshold values, I propose an automatic calibration mechanism using receiver operating characteristics curves (ROC). The VSD and ND use this mechanism for dynamically updating thresholds for optimizing their accuracy of detection. In addition to unwanted calls to the callees in a VoIP network, there can be unwanted traffic coming into a VoIP network that attempts to compromise VoIP network devices. Intelligent hackers can create malicious VoIP traffic for disrupting network activities. Hence, there is a need to frequently monitor the risk levels of critical network infrastructure. Towards realizing this objective, I describe a network level risk management mechanism that prioritizes resources in a VoIP network. The prioritization scheme involves an adaptive re-computation model of risk levels using attack graphs and Bayesian inference techniques. All the above techniques collectively account for a domain-level VoIP security solution.

The objective of this thesis was to develop an 8051 microcontroller soft core in the Verilog hardware description language (HDL). Each functional unit of the 8051 microcontroller was developed as a separate module, and tested for functionality using the open-source VHDL Dalton model as benchmark. These modules were then integrated to operate as concurrent processes in the 8051 soft core. The Verilog 8051 soft core was then synthesized in Quartus® II simulation and synthesis environment (Altera Corp., San Jose, CA, www.altera.com) and yielded the expected behavioral response to test programs written in 8051 assembler residing in the v8051 ROM. The design can operate at speeds up to 41 MHz and used only 16% of the FPGA fabric, thus allowing complex systems to be designed on a single chip. Further research and development can be performed on v8051 to enhance performance and functionality.

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Reverse engineering of hardware IP block is a common practice for competitive purposes in the semiconductor industry. What is done with the information gathered is the deciding legal factor. Once this information gets into the hands of an attacker, it can be used to manufacture exact clones of the hardware device. In an attempt to prevent the illegal copies of the IP block from flooding the market, layout-level obfuscation based on switchable dopant is suggested for the hardware design. This approach can be integrated into the design and manufacturing flow using an obfuscation tool (ObfusTool) to obfuscate the functionality of the IP core. The ObfusTool is developed in a way to be flexible and adapt to different standard cell libraries and designs. It enables easy and accurate evaluation of the area, power and delay v/s obfuscation trades-offs across different design approaches for hardware obfuscation. The ObfusTool is linked to an obfuscation standard cell library which is based on a prototype design created with Obfuscells and 4-input NAND gate. The Obfuscell is a standard cell which is created with switchable functionality based on the assigned dopant configurations. The Obfuscell is combined with other logic gates to form a standard cell library, which can replace any number of existing gates in the IP block without altering it's functionality. A total of 160 different gates are realized using permutated combinations starting with 26 unique gate functions. This design library provide a high level of obfuscation in terms of the number of combinations an adversary has to go through increase to 2 2000 approximately based on the design under consideration. The connectivity of the design has been ignored by previous approaches, which we have addressed in this thesis. The connectivity of a design leaks important information related to inputs and outputs of a gate. We extend the basic idea of dopant-based hardware obfuscation by introducing "dummy wires". The addition of dummy wires not only obfuscates the functionality of the design but also it's connectivity. This greatly reduces the information leakage and complexity of the design increases. To an attacker the whole design appears as one big 'blob'.This also curbs the attempts of brute force attacks. The introduced obfuscation comes at a cost of area and power overhead on an average 5x, which varies across different design libraries.

Intellectual Property (IP) is used to speed up the design process and save money. The use of IP and complex CAD tools reduce visibility into the design and what is actually happening during synthesis and implementation. All of the complexity makes it easier for an attacker to insert malicious logic or tamper with the design in ways that are difficult to detect. Not very much work has been done towards the creation of tools to facilitate the safe use of 3rd-party IP. This work presents Physical and Functional Assurance, two approaches that aim to accomplish this task through physically and logically identical IP instantiation respectively. The approaches and their results and performance impact are presented across a suite of 53 experiments. The Physical Assurance approach is successful at instantiating the 3rd-party IP in the user design without modification and it is also successful at catching even minute tampers along the way. The Functional Assurance approach is shown to be feasible, but still requires work to become a fully-fledged tool.