A mathematical model of the Skype VoIP congestion control algorithm

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. In this paper we focus 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. To the purpose, the controller input/output variables are first identified and then their dynamic relation is described in the form of a hybrid automaton. Main findings are: (1) Skype does not implement a delay based control; (2) the loss ratio is the main input that affects the sending rate; (3) the sending rate matches the available bandwidth with a finite error.