Abstract:
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.
