Performance study of voice over frame relay : a thesis presented in partial fulfilment of the requirements for the degree of Master of Engineering in Information Engineering, Massey University, Albany, New Zealand

Frame Relay (FR) represents an important paradigm shift in modern telecommunication. This technology is beginning to evolve from data only application to broad spectrum of multimedia users and potential to provide end users with cost effective transport of voice traffic for intra office communication. In this project the recent development in voice communication over Frame relay is investigated. Simulations were carried out using OPNET, a powerful simulation software. Following the simulation model, a practical design of the LAN-to-LAN connectivity experiment was also done in the Net Lab. From the results of the simulation, Performance measures such as delay, jitter, and throughput are reported. It is evident from the results that real-time voice or video across a frame relay network can provide acceptable performance

The Evolving Technology-Augmented CourtroomBefore, During, and After the Pandemic

Even before the COVID-19 Pandemic, technology was changing the nature of America’s courtrooms. Access to case management and e-filing data and documents coupled with electronic display of information and evidence at trial, remote appearances, electronic court records, and assistive technology for those with disabilities defined the technology-augmented trial courtroom. With the advent of the Pandemic and the need for social distancing, numerous courts moved to remote appearances, virtual hearings, and even virtual trials. This Article reviews the nature of technology-augmented courtrooms and discusses virtual hearings and trials at length, reviewing legality, technology, human factors, and public acceptance, and concludes that virtual hearings will continue after the Pandemic

Integration of Internet and Telecommunications- An Architecture for Hybrid Services

In this article, we propose an architecture for hybrid services, i.e., services that span many network technologies, such as the Public Switched Telephone Network (PSTN), cellular networks and networks based on the Internet Protocol (IP). These services will play an important role in the future because they leverage on the existing infrastructures, rather than requiring new and sophisticated mechanisms to be deployed. We explore a few issues related to hybrid services and propose a platform, as well as a set of components, to facilitate their creation and deployment. The existing infrastructure is only required to generate specific events when requests for hybrid services are detected. We present the design of a service layer, based on Java, that handles the treatment of these special requests. Our service layer is provided with a set of generic components realized according to the JavaBeans model. We illustrate the strength of our architecture by discussing two hybrid-service examples: a calendar service and a call forwarding service

Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information

Fixed Mobile Convergence a IMS Approach

The paper is aimed at studying and analyzing the network performance parameters of SIP protocol. SIP is content based protocol, in which various message are required to be transacted so that a session could be created, terminated or modified. Therefore, the objective is to analyze various SIP activities and the delay incurred in session start-up under various network conditions. Proper functioning of IMS platform is dependent on optimum performance of several protocols specified in the standard. Nearly all of the protocols used in IMS are standardized by the IETF. Some of the major protocols are SIP, SDP- signaling protocol, DIAMETERimprovised version of RADIUS protocol,COPS- Common Open Policy Service, H.248- descendant of MEGACo, RTP/RTCPReal Time Protocol/Real Time Control Protocol, etc. Out of all these, Session Initiation Protocol (SIP) is the prominent protocol used to create, terminate and modify the sessions initiated by the user. In order to improve the performance parameters, this is the area where most of the research work is centralized. Hence, to study various aspects of SIP protocol with respect to the network performance is of great interest