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The Internet is changing from being only an efficient platform for data delivery to become also a platform for audio/video applications. The stability of the traditional Internet is due to the TCP congestion control algorithm. However, the TCP congestion control is not optimal for VoIP applications because of its retransmission mechanism and additive increase multiplicative decrease sliding window control. As a consequence, VoIP applications often employ proprietary and hidden congestion control algorithms executed over the UDP protocol. This paper focuses on Skype audio, which is the most known and used VoIP application, in order to derive a mathematical model of its congestion control algorithm and evaluate properties such as efficiency in network utilization while avoiding congestion, which is particularly detrimental for VoIP traffic.
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