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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 ?, size of buffer, number of buffers, and the codec type.
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