In general, rural areas in Africa are seen as unprofitable by operators and hence these areas do not benefit from typical wired Internet access. Due to the cost involved, only a small number of people in Africa benefit from the Internet. On the other hand, the Global System for Mobile communications (GSM) has thoroughly penetrated Africa and in many cases unutilized capacity exists in rural areas. Since the cost of GSM services is still extremely high when compared to the average income of African communities, many organizations are now considering alternative business models for the provision of Internet access to disadvantaged populations. To try to bridge the digital divide, the aim of this contribution is to show that basic but affordable Internet Protocol (IP) connectivity can be provided to rural communities by using spare capacity on GSM networks to carry Wireless Local Area Network (WLAN) traffic. Since the main problem with WLAN in Africa is not the last mile, but rather finding a way to connect the wireless access point to an existing backbone network, a solution to integrate WLAN with GSM is proposed.

Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Current best-effort packet-switched Internet is not a perfect environment for real-time applications such as transmitting voice-over the network (Voice Over Internet Protocol or VOIP). Due to the unlimited concurrent access to the Internet by users, the packet loss problem cannot be avoided. Therefore, the VOIP based applications encompass problems such as "voice quality degradation caused by lost packets". The effects of lost packets are fundamental issues in real-time voice transmission over the current unreliable Internet. The dropped packets have a negative impact on voice quality and concealing their effects at the receiver does not deal with all of the drop consequences. It has been observed that in a very lossy network, the receiver cannot cope with all the effects of lost packets and thereby the voice will have poor quality. At this point the Active Networks, a relatively new concept in networking, which allows users to execute a program on the packets in active nodes, can help VOIP regenerate the lost packets, and improve the quality of the received voice. Therefore, VOIP needs special voice-packing methods. Based on the measured packet loss rates, many new methods are introduced that can pack voice packets in such a way that the lost packets can be regenerated both within the network and at the receiver. The proposed voice-packing methods could help regenerate lost packets in the active nodes within the network to improve the perceptual quality of the received sound. The packing methods include schemes for packing samples from low and medium compressed sample-based codecs (PCM, ADPCM) and also include schemes for packing samples from high compressed frame-based codecs (G.729). Using these packing schemes, the received voice has good quality even under very high loss rates. Simulating a very lossy network using NS-2 and testing the regenerated voice quality by an audience showed that significant voice quality improvement is achievable by employing these packing schemes. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1147315 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2004

As part of a larger effort to improve telemetry link availability, the U.S. Army Yuma Proving Ground (YPG) is currently in the process of upgrading its telemetry receivers. As YPG begins integrating new receivers into existing range infrastructure, the question of where to place these receivers to provide maximum benefit must be considered. Should the receivers be placed at each of the remote antennas or should they be centrally located at the primary telemetry site? Although many Telemetry-over-Internet Protocol (TMoIP) and radio frequency (RF) over fiber solutions exist to transport these data, there are numerous concerns including network and Cybersecurity limitations to consider when implementing either of these approaches. This paper will document the trade study conducted at YPG to explore the benefits of each approach.

The principle of parsimonious modelling of Internet traffic states that a minimal number of descriptors should be used for its characterization. Until early 1990s, the conventional Markovian models for voice traffic had been considered suitable and parsimonious for data traffic as well. Later with the discovery of strong correlations and increased burstiness in Internet traffic, various self-similar count models have been proposed. But, in fact, such models are strictly mono-fractal and applicable at coarse time scales, whereas Internet traffic modelling is about modelling traffic at fine and coarse time scales; modelling traffic which can be mono and multi-fractal; modelling traffic at interarrival time and count levels; modelling traffic at access and core tiers; and modelling all the three structural components of Internet traffic, that is, packets, flows and sessions. The philosophy of this thesis can be described as: “the renewal of renewal theory in Internet traffic modelling”. Renewal theory has a great potential in modelling statistical characteristics of Internet traffic belonging to individual users, access and core networks. In this thesis, we develop an Internet traffic modelling framework based on fractal renewal processes, that is, renewal processes with underlying distribution of interarrival times being heavy-tailed. The proposed renewal framework covers packets, flows and sessions as structural components of Internet traffic and is applicable for modelling the traffic at fine and coarse time scales. The properties of superposition of renewal processes can be used to model traffic in higher tiers of the Internet hierarchy. As the framework is based on renewal processes, therefore, Internet traffic can be modelled at both interarrival times and count levels.

Approved for public release, distribution is unlimited / In this thesis, Voice over Internet Protocol (VoIP) technology will be explored and a recommendation of the operational benefit of VoIP will be provided. A network model will be used to demonstrate improvement of voice End-to-End delay by implementing quality of service (QoS) controls. An overview of VoIP requirements will be covered and recommended standards will be reviewed. A clear definition of a Battle Group will be presented and an overview of current analog RF voice technology will be explained. A comparison of RF voice technology and VoIP will modeled using OPNET Modeler 9.0. / Lieutenant, United States Navy

Thesis (M. Tech. Engineering: Electrical--Vaal University of Technology. / This dissertation discusses the design and implementation of a voice over internet protocol system for the multi-user enterprise. It is limited to small to medium enterprises of which the Vaal University of Technology is an example. Voice communications over existing Internet protocol networks are governed by standards, and to develop such a system it is necessary to have a thorough understanding of these standards. Two such standards namely the International Telecommunications Unions H.323 and the Internet Engineering Task Force's SIP were evaluated and compared to each other in terms of their complexity, extensibility and scalability as well as the services they offer. Based on these criteria it was decided to implement a SIP system. A SIP network consists of application software that act as clients and servers, as well as hardware components such as a proxy and redirect and registrar or location servers that allow users of this network to call each other on the data network. Gateways enable users of the network to call regular public switched telephone network numbers. A test network was set up in the laboratory that contained all the hardware and software components. This was done to understand the installation and configuration options of the different software components and to determine the suitability and interoperability of the software components. This network was then migrated to the network of the Vaal University of Technology which allowed selected users to test and use it. Bandwidth use is a major point of contention, and calculations and measurements showed that the codec being used during the voice call is the determining factor. This SIP system is being used on a daily basis and the users report excellent audio quality between soft phones and soft phones, soft phones and normal telephones and even cellular phones.

Recursive Inter-Network Architecture (RINA) networks have a shorter protocol stack than the current architecture (the Internet) and rely instead upon separation of mech- anism from policy and recursive deployment to achieve large scale networks. Due to this smaller protocol stack, fewer networking mechanisms, security or otherwise, should be needed to secure RINA networks. This thesis examines the security proto- cols included in the Internet Protocol Suite that are commonly deployed on existing networks and shows that because of the design principles of the current architecture, these protocols are forced to include many redundant non-security mechanisms and that as a consequence, RINA networks can deliver the same security services with substantially less complexity.

Tese de mestrado integrado. Engenharia Electrotécnica e de Computadores. Universidade do Porto. Faculdade de Engenharia. 2011

Estágio realizado na Sun Microsystems Ireland Inc / Tese de mestrado integrado. Engenharia Informática e Computação. Faculdade de Engenharia. Universidade do Porto. 2009

Video games are an everyday experience for adolescents and have changed how adolescents interact with one another. Prior research has focused on positive and negative aspects of video game play in general, without distinguishing Voice Over Internet Protocol (VOIPing) as the mode of play. Grounded in entertainment theory, motivational theory, and psychological distress theory, this cross-sectional, correlational study examined the relationship between VOIPing and quality of life (Pediatric Quality of Life Inventory), Yee's motivation to play video games, and resilience (Child and Youth Resilience Measure). A series of linear regression and multivariate canonical correlation models analyzed self-report responses of 103 adolescents aged 13 to18. Results indicated that VOIPing was not statistically related to quality of life or resilience. However, VOIPing correlated positively with motivation to play video games, particularly with the subscales of socialization and relationships. Canonical analysis of motivation for gaming and quality of life indicated that adolescents with high scores on customization and escapism motivation for gaming subscales tended to also have high scores on each of the emotional, social, and school quality of life subscales. Canonical analysis of motivation for gaming and resilience indicated that adolescents with low scores on the escapism motivation for gaming subscale tended to also have high scores on the individual, relationships, and community resilience subscales. The positive aspects of VOIPing, particularly with increased motivation to play video games, can be effectively used in coaching adolescents in social skills and relationship building.