Enterprise Voice-over-IP Traffic Monitoring

The contribution of this work is an extensible and flexible framework designed and implemented in order to satisfy the disparate requirements introduced by service oriented network monitoring needs

Systemization of Pluggable Transports for Censorship Resistance

An increasing number of countries implement Internet censorship at different scales and for a variety of reasons. In particular, the link between the censored client and entry point to the uncensored network is a frequent target of censorship due to the ease with which a nation-state censor can control it. A number of censorship resistance systems have been developed thus far to help circumvent blocking on this link, which we refer to as link circumvention systems (LCs). The variety and profusion of attack vectors available to a censor has led to an arms race, leading to a dramatic speed of evolution of LCs. Despite their inherent complexity and the breadth of work in this area, there is no systematic way to evaluate link circumvention systems and compare them against each other. In this paper, we (i) sketch an attack model to comprehensively explore a censor's capabilities, (ii) present an abstract model of a LC, a system that helps a censored client communicate with a server over the Internet while resisting censorship, (iii) describe an evaluation stack that underscores a layered approach to evaluate LCs, and (iv) systemize and evaluate existing censorship resistance systems that provide link circumvention. We highlight open challenges in the evaluation and development of LCs and discuss possible mitigations.Comment: Content from this paper was published in Proceedings on Privacy Enhancing Technologies (PoPETS), Volume 2016, Issue 4 (July 2016) as "SoK: Making Sense of Censorship Resistance Systems" by Sheharbano Khattak, Tariq Elahi, Laurent Simon, Colleen M. Swanson, Steven J. Murdoch and Ian Goldberg (DOI 10.1515/popets-2016-0028

A comparative study of in-band and out-of-band VOIP protocols in layer 3 and layer 2.5 environments

For more than a century the classic circuit-switched telephony in the form of PSTN (Public Service Telephone Network) has dominated the world of phone communications (Varshney et al., 2002). The alternative solution of VoIP (Voice over Internet Protocol) or Internet telephony has increased dramatically its share over the years though. Originally started among computer enthusiasts, nowadays it has become a huge research area in both the academic community as well as the industry (Karapantazis and Pavlidou, 2009). Therefore, many VoIP technologies have emerged in order to offer telephony services. However, the performance of these VoIP technologies is a key issue for the sound quality that the end-users receive. When making reference to sound quality PSTN still stands as the benchmark.Against this background, the aim of this project is to evaluate different VoIP signalling protocols in terms of their key performance metrics and the impact of security and packet transport mechanisms on them. In order to reach this aim in-band and out-of-band VoIP signalling protocols are reviewed along with the existing security techniques which protect phone calls and network protocols that relay voice over packet-switched systems. In addition, the various methods and tools that are used in order to carry out performance measurements are examined together with the open source Asterisk VoIP platform. The findings of the literature review are then used in order to design and implement a novel experimental framework which is employed for the evaluation of the in-band and out-of-band VoIP signalling protocols in respect to their key performance networks. The major issue of this framework though is the lack of fine-grained clock synchronisation which is required in order to achieve ultra precise measurements. However, valid results are still extracted. These results show that in-band signalling protocols are highly optimised for VoIP telephony and outperform out-of-band signalling protocols in certain key areas. Furthermore, the use of VoIP specific security mechanisms introduces just a minor overhead whereas the use of Layer 2.5 protocols against the Layer 3 routing protocols does not improve the performance of the VoIP signalling protocols

Performance evaluation of information and communications technology infrastructure for smart distribution network applications

This thesis was submitted for the degree of Master of Philosophy and awarded by Brunel University.Current electrical networks require secure, scalable and cost-effective Information and Communications Technology (ICT) solutions to facilitate the novel functionalities required by Smart Grids. Countries around the globe are investigating alternative energy sources to mitigate the current energy crisis and environmental issues experienced by many countries due to global warming, rapid growth of population, inefficient energy management, dwindling fossil fuel resources, etc. Therefore, alternative or renewable energy sources, such as wind, solar, hydro, combined heat and power, etc., are required to mitigate such a crisis and such sources will also need to be integrated in to the power grid in a distributed manner. Such distributed energy sources are mainly connected to the distribution networks and introduce huge challenges to the distribution network operator (DNO). Many of these challenges cannot be dealt with effectively using existing network operation mechanisms therefore the research and development of novel ICT solutions to support smart distribution network operation is required. This research investigated suitable ICT solutions to enable the Smart Grid to tackle these challenges and proposes ICT infrastructure models that can be used for simulation studies in order to investigate cost-effective, scalable and secure solutions for the DNOs. Initially, a Quality of Service (QoS) monitoring test-bed was proposed to evaluate the performance of bandwidth intensive applications, such as smart meter data transmission. Simulation studies for different communication technologies, cellular and Power Line Communication (PLC), were also carried out and the simulation models were verified using experimental test results. Finally, the modelling and analysis of smart metering infrastructure was carried out using simulation and extensive studies were performed to evaluate the data transmission rate performance for different configurations of smart meters and concentrators

Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

Buffering principles for mobile multimedia over IP

Masteroppgave i informasjons- og kommunikasjonsteknologi 2001 - Høgskolen i Agder, GrimstadThis thesis suggests a test-bed for reviewing queue algorithms suitable for IP based mobile multimedia services. Subjective performance quality obtained in the test-bed was analysed using a network performance monitor tool. In order to evaluate possible services for use in the test-bed, a comprehensive survey was conducted. The survey revealed that a variety of categorization methods for electronic services exist. However, none of them were targeted at mobile multimedia services. Based on the different methods a framework of six service classes were proposed. Together, these classes provide a useful and easy-to-navigate overview of both existing and future mobile services. Most services on the Internet use TCP for end-to-end message transfer. It is reasonable to expect services in IP based mobile systems to use this protocol as well. TCP offers reliable connection management and is known as an extremely trustworthy protocol when used on wired links. Running TCP over wireless links is another story, though. Selected documents and reports that discuss TCP in wireless environments are evaluated, and proposed solutions are commented. Two queuing algorithms were implemented in a router. Streaming video from a server on the Internet was routed to a host inside the selected test-bed. Run through Microsoft Media Player, the perceived quality of picture and sound was described. This description was then matched to a captured data flow from the same streaming session. The comparison did not reveal strict relations between subjective experience and objective measurements. Possible explanations for this are discussed at the end of the thesis document

Measuring And Improving Internet Video Quality Of Experience

Streaming multimedia content over the IP-network is poised to be the dominant Internet traffic for the coming decade, predicted to account for more than 91% of all consumer traffic in the coming years. Streaming multimedia content ranges from Internet television (IPTV), video on demand (VoD), peer-to-peer streaming, and 3D television over IP to name a few. Widespread acceptance, growth, and subscriber retention are contingent upon network providers assuring superior Quality of Experience (QoE) on top of todays Internet. This work presents the first empirical understanding of Internet’s video-QoE capabilities, and tools and protocols to efficiently infer and improve them. To infer video-QoE at arbitrary nodes in the Internet, we design and implement MintMOS: a lightweight, real-time, noreference framework for capturing perceptual quality. We demonstrate that MintMOS’s projections closely match with subjective surveys in accessing perceptual quality. We use MintMOS to characterize Internet video-QoE both at the link level and end-to-end path level. As an input to our study, we use extensive measurements from a large number of Internet paths obtained from various measurement overlays deployed using PlanetLab. Link level degradations of intra– and inter–ISP Internet links are studied to create an empirical understanding of their shortcomings and ways to overcome them. Our studies show that intra–ISP links are often poorly engineered compared to peering links, and that iii degradations are induced due to transient network load imbalance within an ISP. Initial results also indicate that overlay networks could be a promising way to avoid such ISPs in times of degradations. A large number of end-to-end Internet paths are probed and we measure delay, jitter, and loss rates. The measurement data is analyzed offline to identify ways to enable a source to select alternate paths in an overlay network to improve video-QoE, without the need for background monitoring or apriori knowledge of path characteristics. We establish that for any unstructured overlay of N nodes, it is sufficient to reroute key frames using a random subset of k nodes in the overlay, where k is bounded by O(lnN). We analyze various properties of such random subsets to derive simple, scalable, and an efficient path selection strategy that results in a k-fold increase in path options for any source-destination pair; options that consistently outperform Internet path selection. Finally, we design a prototype called source initiated frame restoration (SIFR) that employs random subsets to derive alternate paths and demonstrate its effectiveness in improving Internet video-QoE

Study of voice quality in IP networks

Orientador: Helio WaldmanDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de ComputaçãoAbstract: This work describes the study of voice quality in IP networks based on a revision of quality of service (QoS) protocols and mechanisms and aspects of the system impact associated with the presence or absence of them; revision of the diverse evaluation methods of voice quality with emphasis in the automatic methods (objective and repetitive) used to analyze the effects in the voice due to diverse factors presented in packet networks, such as packet loss, delay and jitter, as well as the proper voice coding at low rate; revision of the main protocols of signalling for implementation of voice over IP (VoIP) or IP telephony, considering its strong and weak points with regard to implementation facility, scalability and adequacy for some network applications and quality of service (QoS) and accomplishment of tests in simulated and experimental IP networks. The main objective is the characterization of voice service in IP networks taking into account the effect of the network factors in call set-up (connection) and in voice quality . For the simulation of the IP network the Shunra¿s Cloud software was used where it is possible to deal with, in isolated form, the influence of packet loss, fixed delay, delay variation ( jitter), as well as the composed effect of packet loss and jitter. Solutions to such effects are investigated in experimental tests. Results of system simulations are presented and discussed. Degradations in voice due to such effects are evaluated and a practical method to solve them is considered. The experimental results demonstrate the technical feasibility of using voice over IP (or IP telephony) by service providers, as well as private corporations being able to forward voice and data over the same converged IP networkResumo: Este trabalho descreve o estudo da qualidade de voz em redes IP a partir de uma revisão dos protocolos e mecanismos relativos a qualidade de serviço (QoS) e os aspectos do impacto sistêmico na presença ou ausência destes; revisão dos diversos métodos de avaliação da qualidade da voz com ênfase nos métodos automáticos (objetivos e repetitivos) para auxiliar na análise dos efeitos na voz dos diversos fatores presentes em uma rede de pacotes, tais como perda de pacote, atraso e jitter, bem como a própria codificação da voz em baixas taxas; revisão dos principais protocolos de sinalização utilizados para implementação de voz sobre IP (VoIP) ou telefonia sobre IP, evidenciando-se seus pontos fortes e fracos com relação a facilidade de implementação, extensibilidade e adequabilidade para várias aplicações de rede e qualidade de serviço (QoS) e realização de testes em redes IP simulada e experimental. O principal objetivo é a caracterização do serviço de voz em redes IP levando-se em consideração os efeitos dos fatores de rede e gateway no tempo de estabelecimento de uma chamada (conexão) e na qualidade da voz. Para simulação da rede IP foi utilizado o software Cloud da Shunra onde é possível tratar, de forma isolada, a influência da perda de pacote, do atraso fixo, do atraso variável (jitter), bem como do efeito conjunto da perda de pacote e jitter. Soluções a tais efeitos são investigadas em testes experimentais. Resultados de simulações sistêmicas são apresentados e discutidos. As degradações na voz devidas a tais efeitos são avaliadas e um método prático para solucionar é testado. Os resultados experimentais demonstram a viabilidade técnica da utilização da voz sobre IP (ou telefonia IP) pelos provedores de serviço, bem como pelas corporações privadas, podendo trafegar voz e dados em uma mesma rede convergente IPMestradoTelecomunicações e TelemáticaMestre em Engenharia Elétric

3D multiple description coding for error resilience over wireless networks

This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University.Mobile communications has gained a growing interest from both customers and service providers alike in the last 1-2 decades. Visual information is used in many application domains such as remote health care, video –on demand, broadcasting, video surveillance etc. In order to enhance the visual effects of digital video content, the depth perception needs to be provided with the actual visual content. 3D video has earned a significant interest from the research community in recent years, due to the tremendous impact it leaves on viewers and its enhancement of the user’s quality of experience (QoE). In the near future, 3D video is likely to be used in most video applications, as it offers a greater sense of immersion and perceptual experience. When 3D video is compressed and transmitted over error prone channels, the associated packet loss leads to visual quality degradation. When a picture is lost or corrupted so severely that the concealment result is not acceptable, the receiver typically pauses video playback and waits for the next INTRA picture to resume decoding. Error propagation caused by employing predictive coding may degrade the video quality severely. There are several ways used to mitigate the effects of such transmission errors. One widely used technique in International Video Coding Standards is error resilience. The motivation behind this research work is that, existing schemes for 2D colour video compression such as MPEG, JPEG and H.263 cannot be applied to 3D video content. 3D video signals contain depth as well as colour information and are bandwidth demanding, as they require the transmission of multiple high-bandwidth 3D video streams. On the other hand, the capacity of wireless channels is limited and wireless links are prone to various types of errors caused by noise, interference, fading, handoff, error burst and network congestion. Given the maximum bit rate budget to represent the 3D scene, optimal bit-rate allocation between texture and depth information rendering distortion/losses should be minimised. To mitigate the effect of these errors on the perceptual 3D video quality, error resilience video coding needs to be investigated further to offer better quality of experience (QoE) to end users. This research work aims at enhancing the error resilience capability of compressed 3D video, when transmitted over mobile channels, using Multiple Description Coding (MDC) in order to improve better user’s quality of experience (QoE). Furthermore, this thesis examines the sensitivity of the human visual system (HVS) when employed to view 3D video scenes. The approach used in this study is to use subjective testing in order to rate people’s perception of 3D video under error free and error prone conditions through the use of a carefully designed bespoke questionnaire.Petroleum Technology Development Fund (PTDF