Monitoring the user perceived quality of SILK-based voice calls

Quality measurements are required to support VoIP traffic in the Internet. The widely used average Mean Opinion Score is, however, not sufficient for this task. In this paper, we provide a detailed analysis of the Skype SILK codec and compare it with the iLBC and GSM codec. The results show that the SILK codec is superior to the other codecs in scenarios with random and bulk packet loss. This increased tolerance of packet loss enables the option of QoE Monitoring under reasonable network conditions. From analysis, we derive an estimation, which can be used to monitor the MOS value of the users in the worst case. Furthermore, we show how sampling can effectively decrease the required measurement effort.