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Thesis (Ph. D.)--University of Texas at Austin, 2003. / Vita. Includes bibliographical references. Available also from UMI Company.

Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2010. / Committee Chair: Altunbasak, Yucel; Committee Co-Chair: Zhou, G. Tong; Committee Member: Al-Regib, Ghassan; Committee Member: Kenney, James Stevenson; Committee Member: Ma, Xiaoli; Committee Member: Pan, Ronghua. Part of the SMARTech Electronic Thesis and Dissertation Collection.

The purpose of this thesis is to explore the following research question: How does action research serve as a research method in the discipline of Communications? Specifically, this study will approach the question by analyzing existing literature on action research and also performing an action research trial of a new marketing approach using digital telephony. The study finds that action research has a combination of four characteristics that make it a discrete method of research. Action research involves collaboration, invokes change, requires a researcher's vested interest, and allows a kind of knowing that can only come from direct involvement in a change. / Department of Telecommunications

Thesis (Ph. D.)--Carleton University, 2003. / Includes bibliographical references (p. 151-158). Also available in electronic format on the Internet.

Thesis (Ph. D.)--University of North Texas, Dec., 2007. / Title from title page display. Includes bibliographical references.

Thesis (M. Eng.)--Carleton University, 2000. / Includes bibliographical references (p. 94-99). Also available in electronic format on the Internet.

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.

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

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.