Publication details

The purpose of this research is to investigate how Skype adapts its voice sender bitrate in reaction to network congestions and what Skype’s performances are in terms of voice Quality of Experience (QoE) or Mean Opinion Score (MOS) under different network conditions. The voice quality of Skype was evaluated by setting up a VoIP testbed that allowed making voice calls between two Skype clients, each installed on a separate PC. The network conditions were altered as desired and sample speech files from the ITU-T P.50 database were used to play through Skype during a Skype-to-Skype voice call. The resulting audio was recorded along with the call traffic being captured. The audio quality of the recorded speech samples was then evaluated by using the PESQ algorithm and the network parameters were calculated from the captured call traffic using Awk scripts. The results obtained showed that the voice quality of Skype deteriorates with increasing amounts of packet loss in the network. Further, the throughput values that were obtained indicated that Skype tends to reduce its send bitrate when it encounters packet loss in the network which could result in increased delay. The values obtained for the interarrival times indicated that the packet interarrival time increases with increasing packet loss and this could lead to increase in the mouth-to-ear delay. Finally, the jitter values obtained were found to be well within the acceptable limit for VoIP and thus it could be concluded that jitter does not have much effect on the audio quality of Skype.

Pradhan M, Sun L