Modelling and Simulation of Voice over Internet Protocol (VOIP)

Abstract

Real time voice transmission is now widely used over the Internet and has become a very significant application. Voice quality is still however an open problem due to the loss of voice packets and the variation of end-to-end delay packet transmission. These two factors are a natural result of the simple ‘best-effort service’ provided by the current network. Indeed, the nowadays Internet provides with it a simple packet delivery service without any guarantee on bandwidth, delay or drop probability. The focus in this paper is the simulation of two types of models; a M/M/1 queue and the M/G/1 queue, both using an input of ë, size of buffer, number of buffers, and the codec type. The output that was examined is the Quality of service parameters such as the End to End Delay, Packet Loss and Jitter. It was found that in order to control system behavior it’s important to make sure that good tuning is used, as based on this paper’s results; it can reduce the network congestion.