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Real time audio transmission over the internet has become increasingly popular in many application areas, in particular in the commercial and entertainment sectors. However, the most challenging problem of end-to-end delay over an IP network needs to be addressed and is a topic of current ongoing research. In this paper various issues related to streaming real time audio in an IP network including the protocols used, problems faced in transmission and reception have been reviewed. A new scheme to evaluate the effect of buffer size during reception of an audio transmission in order to achieve an optimized playback is developed and tested. This method can be used to optimize the reception quality with respect to time and provide a way to reduce the noise effects at the receiver. It helps to stress out the factors affecting audio playback performance for the consumer.