The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments. / KMBT_363 / Adobe Acrobat 9.54 Paper Capture Plug-in

Since the deregulation of Voice over Internet Protocol (VoIP) in 2005, many South African organizations are now attempting to leverage its cost saving and competitive values. However, it has been recently cited that VoIP is one of the greatest new risks to organizations and this risk is cited to increase Information Security insurance premiums in the near future. Due to the dynamic nature of the VoIP technology, regulatory and legislative concerns such as lawful interception of communications and privacy may also contribute to business risk. In order to leverage value from the VoIP implementation, an organization should implement the technology with knowledge of the potential risk of civil liability. This is further highlighted by the King III Report which indicates that the Directors of an organization should be ultimately responsible for Corporate Governance and, therefore, IT Governance and Information Security Governance. The report goes further to say that any newly implemented technology, such as VoIP, should comply with all South African legislation and regulations. This responsibility encourages the practice of both due care and due diligence. However, recent trends exercised by Information Security professionals, responsible for drafting Information Security policies and related procedures, often neglect the regulatory requirements and choose to only implement international best practices with no consideration of the risk of civil liability. Although these best practice frameworks may inadvertently comply with existing local legislation, a chance of an oversight is possible. Oversights may not only result in criminal sanctions, but also civil action due to losses or damages suffered. With regard to implementing VoIP, good Corporate Governance could potentially be ensured through the use of both identified regulations and relevant international best practices. This dissertation aims to aid organizations in avoiding or at least mitigating the risk of civil liability to better leverage VoIP’s value, through good Corporate Governance practices. This should aid in the exercise of due care and due diligence when implementing VoIP as a means of conducting business communication.

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.

Thesis (M. Tech.) - Central University of Technology, Free State, 2008

Thesis (MScEng)--University of Stellenbosch, 2011. / ENGLISH ABSTRACT: The perceived quality of Voice over Internet Protocol (IP) (VoIP) communication relies on the network which is used to transport voice packets between the end points. Variable network characteristics such as bandwidth, delay and loss are critical for real-time voice traffic and are not always guaranteed by networks. It is important for network service providers to determine the Quality of Service (QoS) it provides to its customers. The solution proposed here is to predict the perceived quality of a VoIP call, in real-time by using network statistics. The main objective of this thesis is to develop a network analysis tool, which gathers meaningful statistics from network traffic. These statistics will then be used for predicting the perceived quality of a VoIP call. This study includes the investigation and deployment of two main components. Firstly, to determine call quality, it is necessary to extract the voice streams from captured network traffic. The extracted sound files can then be analysed by various VoIP quality models to determine the perceived quality of a VoIP call. The second component is the analysis of network characteristics. Loss, delay and jitter are all known to influence perceived call quality. These characteristics are, therefore, determined from the captured network traffic and compared with the call quality. Using the statistics obtained by the repeated comparison of the call quality and network characteristics, a network specific algorithm is generated. This Non-Intrusive Quality Prediction Algorithm (NIQPA) uses basic characteristics such as time of day, delay, loss and jitter to predict the quality of a real-time VoIP call quickly in a non-intrusive way. The realised algorithm for each network will differ, because every network is different. Prediction results can then be used to adapt either the network (more bandwidth, packet prioritising) or the voice stream (error correction, change VoIP codecs) to assure QoS. / AFRIKAANSE OPSOMMING: Die kwaliteit van spraak oor die internet (VoIP) kommunikasie is afhanklik van die netwerk wat gebruik word om spraakpakkies te vervoer tussen die eindpunte. Netwerk eienskappe soos bandwydte, vertraging en verlies is krities vir intydse spraakverkeer en kan nie altyd gewaarborg word deur netwerkverskaffers nie. Dit is belangrik vir die netwerk diensverskaffers om die vereiste gehalte van diens (QoS) te verskaf aan hul kliënte. Die oplossing wat hier voorgestel word is om die kwaliteit van ’n VoIP oproep intyds te voorspel, deur middel van die netwerkstatistieke. Die belangrikste doel van hierdie projek is om ’n netwerk analise-instrument te ontwikkel. Die instrument versamel betekenisvolle statistiek deur van netwerkverkeer gebruik te maak. Hierdie statistiek sal dan gebruik word om te voorspel wat die gehalte van ’n VoIP oproep sal wees vir sekere netwerk toestande. Hierdie studie berus op die ondersoek en implementering van twee belangrike komponente. In die eerste plek, moet oproep kwaliteit bepaal word. Spraakstrome word uit die netwerkverkeer onttrek. Die onttrekte klanklêers kan dan geanaliseer word deur verskeie spraak kwaliteitmodelle om die kwaliteitdegradasie van ’n spesifieke VoIP oproep vas te stel. Die tweede komponent is die analise van netwerkeienskappe. Pakkieverlies, pakkievertraging en bibbereffek is bekend vir hul invloed op VoIP kwaliteit en is waargeneem. Hierdie netwerk eienskappe word dus bepaal uit die netwerkverkeer en daarna vergelyk met die gemete gesprekskwaliteit. Statistiek word verkry deur die herhaalde vergelyking van gesprekkwaliteit en netwerk eienskappe. Uit die statistiek kan ’n algoritme (vir die spesifieke network) gegenereer word om spraakkwaliteit te voorspel. Hierdie Nie-Indringende Kwaliteit Voorspellings-algoritme (NIKVA), gebruik basiese kenmerke, soos die tyd van die dag, pakkie vertraging, pakkie verlies en bibbereffek om die kwaliteit van ’n huidige VoIP oproep te voorspel. Hierdie metode is vinnig, in ’n nie-indringende manier. Die gerealiseerde algoritme vir die verskillende netwerke sal verskil, want elke netwerk is anders. Die voorspelling van spraakgehalte kan dan gebruik word om òf die netwerk aan te pas (meer bandwydte, pakkie prioriteit) òf die spraakstroom aan te pas (foutkorreksie, verander VoIP kodering) om die goeie kwaliteit van ’n VoIP oproep te verseker.

Approved for public release, distribution is unlimited / Converging voice and data networks has the potential to save money and is the main reason Voice over Internet Protocol (VoIP) is quickly becoming mainstream in corporate America. The potential VoIP offers to more efficiently utilize the limited connectivity available to ships at sea makes it an attractive option for the Navy. This thesis investigates the usefulness of VoIP for the communications needs of a unit level ship. This investigation begins with a review of what VoIP is and then examines the ship to shore connectivity for a typical unit level ship. An OMNeT++ model was developed and used to examine the issues that affect implementing VoIP over this type of link and the results are presented. / Lieutenant Commander, United States Navy

The issues involved in Internet telephony (Voice over Internet Protocol (VoIP) device) can be best understood by actually implementing a VoIP device and studying its performance. In this regard, an Internet telephony device, providing full duplex voice communication over internet, and a user interface program have been developed. In the process, a number of implementation issues came into focus, which we have touched upon in this thesis. Transport layer network protocols are discussed in the concept of real time streaming applications and Real Time Protocol (RTP) is modified to use as transport layer protocol in developed VoIP device. Adaptive playout buffering algorithms are studied and compared with each other by trace driven simulation experiments with objective measures. A method to solve clock synchronization problem in streaming internet applications is presented. One way and round trip delay measurement functionalities are added to the VoIP device, so that device can be used to investigate the network characteristics.

A Masters Dissertation submitted in fulfilment of the requirements of the degree of Master of Commerce in the faculty of Commerce, Law and Management August, 2016 / Online activity through the Internet and mobile phones has dramatically increased over the last five years in South Africa (Nyirenda-Jere & Tesfaye , 2015). Lower costs for Internet and mobile phones are the main reasons for more and more people being connected (Price Waterhouse Cooper South Africa, 2012). But discrepancies exist, namely between the people who are connected and those who are not. This discrepancy is referred to as the digital divide and contributing factors towards it include income, education, age and other factors which were discussed in this paper (Nievhaves, Gorbacheva & Plattfaut, 2012). Free Wi-Fi is one of the solutions to bridge the digital divide to a certain extent and it is also a very valuable tool to marketers and business owners. This research study was aimed at understanding the impact of free Wi-Fi on consumers’ purchase intentions and likelihood to return in townships in South Africa. People in townships are an important group to analyse, because of the millions of inhabitants. Infrastructure in terms of the Internet is not as good as the infrastructure standards in suburbs or in the city. The purpose of the study was to find out the impact of free Wi-Fi on the likelihood of customers to return and their likelihood to purchase something at a location where free Wi-Fi is offered. For the purpose of this research a quantitative approach was used to investigate the impact of free Wi-Fi and factors leading to return and intention to conduct purchases. Non-probability sampling was used in the form of convenience sampling. A self-administered questionnaire was developed to investigate behaviour. Four hundred questionnaires were distributed to people living in Soweto. The analysis indicated that the four access variables, namely - material, mental, usage and skills access - have an influence on the intention to use free Wi-Fi which, in turn, has an influence on the likelihood to return or purchase something. Implications for marketers and businesses is: the marketers have to consider the digital divide when marketing to consumers in South Africa and that offering free Wi-Fi at a commercial place has positive implications for both customers and businesses. / MT2017

Approved for public release, distribution is unlimited / Recent advances in information technology communications have brought about increases in bandwidth and processing speeds to encourage the growth of Internet Protocol Telephony (IPT), a method of transmitting voice conversations over data networks. Many organizations are replacing portions of their traditional phone systems to gain the benefits of cost savings and enhanced feature sets through the use of IPT. The Coast Guard has an interest in exploiting this technology, and has taken its first steps by implementing IPT at Headquarters Support Command in Washington D.C. This thesis investigates the successful implementation practices and security policies of commercial, educational, and government organizations in order to create recommendations for IPT security policies and implementation practices relevant to the Coast Guard. It includes the discussion of the public switched telephone network, an overview of IPT, IPT security issues, the safeguards available to counter security threats, the tradeoffs (e.g., voice quality, cost) required to mitigate security risks, and current IPT security policy and implementation guidance. It is supported by the study and analysis of the IPT system at Coast Guard Headquarters. The Coast Guard gains an understanding of the advantages, limitations, and security issues that it will face as it considers further implementation of IPT. / Lieutenant, United States Coast Guard

<p>Providing carrier grade characteristics for Internet Protocol (IP) communication applications is a significant problem for IP application providers in order to offer integrated services that span IP&nbsp / and telecommunication networks. This thesis addresses the provision of life-cycle management, which is only one carrier grade characteristic, for a SoftBridge application, which is an example of IP communication applications. A SoftBridge provides semi-synchronous multi-modal IP-based communication. The work related to IP-Telecommunication integrated services and the SoftBridge is analyzed with respect to life-cycle management in a literature review. It is suggested to use an Application Server in a Next Generation Network (NGN) to provide life-cyclemanagement functionality for IP-Telecommunication applications. In this thesis, the Application Server is represented by a JAIN Service Logic Execution Environment(JSLEE), in which&nbsp / a SoftBridge application can be deployed, activated, deactivated, uninstalled and upgraded online.Two methodologies are applied in this research: exploratory prototyping, which evolves the development of a SoftBridge application, and empirical comparison, which is concerned with the empirical evaluation of a SoftBridge application in terms of carriergrade capabilities. A SoftBridge application called SIMBA&nbsp / provides a Deaf Telephony service similar to aprevious Deaf Telephony SoftBridge, However, SIMBA&rsquo / s SoftBridge design and implementation are unique to this thesis. In order to test the life-cycle&nbsp / management ability of SIMBA, an empirical evaluation is carried out including the experiments oflife-cycle management and call-processing performance. The final experimental results of the evaluation show that a JSLEE is able to provide life-cycle management for SIMBA without causing a significant decrease in performance. In conclusion, the life-cycle management can be provided&nbsp / or a SoftBridge application by using an Application Server such as a JSLEE. Futhermore, the results indicate that&nbsp / approach of using Application Server (JSLEE) integration should be&nbsp / sufficiently general to provide life cycle management, and indeed other carrier grade capabilities, for other IP communication applications. This allows IP communication applications to be&nbsp / &nbsp / &nbsp / integrated into an NGN.</p>