Thesis (M. App. Sc.)--Carleton University, 2003. / Includes bibliographical references (p. 114-122). Also available in electronic format on the Internet.

Thesis (M.App.Sc.) - Carleton University, 2006. / Includes bibliographical references (p. 114-119). Also available in electronic format on the Internet.

Dissertation (MCIS - Computer and Information Sciences) -- AUT University, 2008. / Includes bibliographical references. Also held in print (ix, 65 leaves : ill. ; 30 cm.) in City Campus Theses Collection (T 621.385 WU)

Thesis (M.Sc.) - Carleton University, 2007. / Includes bibliographical references (p. 90-94). Also available in electronic format on the Internet.

Thesis (Ph.D.) - Carleton University, 2007. / Includes bibliographical references (p. 193-203). Also available in electronic format on the Internet.

When adding Confidentiality, Integrity and Availability (CIA) to a multi-user VoIP (Voice over IP) system, performance and quality are at risk. The aim of this study is twofold. Firstly, it describes current methods suitable to secure voice streams within a VoIP system and make them available in an Asterisk-based VoIP environment. (Asterisk is a well established, open-source, TDM/VoIP PBX.) Secondly, this study evaluates the performance cost incurred after implementing each security method within the Asterisk-based system, using a special testbed suite, named DRAPA, which was developed expressly for this study. The three security methods implemented and studied were IPSec (Internet Protocol Security), SRTP (Secure Real-time Transport Protocol), and SIAX2 (Secure Inter-Asterisk eXchange 2 protocol). From the experiments, it was found that bandwidth and CPU usage were significantly affected by the addition of CIA. In ranking the three security methods in terms of these two resources, it was found that SRTP incurs the least bandwidth overhead, followed by SIAX2 and then IPSec. Where CPU utilisation is concerned, it was found that SIAX2 incurs the least overhead, followed by IPSec, and then SRTP.

Magister Scientiae - MSc / SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints. / South Africa

Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.

Orientador: Leonardo de Souza Mendes / Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação / Made available in DSpace on 2018-08-17T15:12:52Z (GMT). No. of bitstreams: 1 Barison_Dherik_M.pdf: 8956445 bytes, checksum: 9549370d08573863a0f2bab25d076945 (MD5) Previous issue date: 2010 / Resumo: A proposta desta dissertação é avaliar a qualidade de chamadas VoIP cifradas com diferentes algoritmos de criptografia através do OpenVPN, com o objetivo de identificar as diferenças de resultados entre os algoritmos de criptografia e também entre as chamadas cifradas e as não cifradas. Esta avaliação ocorrerá utilizando o MOS (Mean Opinion Score), um método que permite indicar a satisfação do usuário quanto a qualidade da comunicação. As chamadas VoIP cifradas irão ocorrer em diferentes cenários de rede que apresentam diversos problemas, tais como perda de pacotes, pacotes fora de seqüência, atraso, largura de banda de rede, etc. Estes cenários foram baseados em algumas situações reais de uso e serão emulados através da ferramenta Traffic Control do Linux, capaz de manipular os pacotes enviados por qualquer uma das interfaces de rede. Os cenários também terão diferentes larguras de banda de rede, para avaliar a influência das mesmas em algumas situações. / Abstract: The purpose of this work is to evaluate the quality of encrypted VoIP calls with different cipher algorithms through OpenVPN software, in order to identify differences in results between encryption algorithms and also differences between non-encrypted and encrypted calls. This evaluation will do by the MOS (Mean Opinion Score), a method to indicate user satisfaction of communication quality. The encrypted VoIP calls will occur in different network scenarios that present different problems, like packet loss, out-of-order packets, delay, network bandwidths, etc. These scenarios were based on some real situations of use and will be emulated with the Traffic Control tool from Linux, able of handling the packages sent by any available network interface. The scenarios will also have different network bandwidths to assess its importance in some situations. / Mestrado / Telecomunicações e Telemática / Mestre em Engenharia Elétrica

Voice-enabled applications, applications that interact with a user via an audio channel, are used extensively today. Their use is growing as speech related technologies improve, as speech is one of the most natural methods of interaction. They can provide customer support as IVRs, can be used as an assistive technology, or can become an aural interface to the Internet. Given that the telephone is used extensively throughout the globe, the number of potential users of voice-enabled applications is very high. VoiceXML is a popular, open, high-level, standard means of creating voice-enabled applications which was designed to bring the benefits of web based development to services. While VoiceXML is an ideal language for creating these applications, VoiceXML gateways, the hardware and software responsible for interpreting VoiceXML applications and interfacing with the PSTN, are still expensive and so there is a need for a low-cost gateway. Asterisk, and open-source, TDM/VoIP telephony platform, can be used as a low-cost PSTN interface. This thesis investigates adding a VoiceXML service to Asterisk, creating a low-cost VoiceXML prototype gateway which is able to render voice-enabled applications. Following the Component-Based Software Engineering (CBSE) paradigm, the VoiceXML gateway is divided into a set of components which are sourced from the open-source community, and integrated to create the gateway. The browser requires a VoiceXML interpreter (OpenVXI), a Text-To-Speech engine (Festival) and a speech recognition engine (Sphinx 4). The integration of the components results in a low-cost, open-source VoiceXML gateway. System tests show that the integration of the components was successful, and that the system can handle concurrent calls. A fully compliant version of the gateway can be used in the real world to render voice-enabled applications at a low cost. / KMBT_363 / Adobe Acrobat 9.55 Paper Capture Plug-in