Investigating the Extent and Impact of Time-Scaling in WebRTC Voice Traffic Under Light, Moderate and Heavily Congested Wi-Fi APs

Real-time communication (RTC) applications like VoIP ideally require networks that support the necessary quality of service (QoS) whereas the reality is that network impairments such as latency, jitter and packet loss exist. In order to cope with jitter and delay, some VoIP applications employ time-scale modification or warping in the jitter buffer that adjusts the rate of playout while controlling the pitch to minimize Mouth-to-Ear (M2E) delay whilst preserving speech intelligibility and quality. In this paper, we firstly investigate the extent to which timescaling occurs using WebRTC [1] VoIP clients over Wi-Fi networks with different levels of congestion. We then assess the impact of such time-scaling, both subjectively via expert listening test and objectively using POLQA, on quality experienced by the end user, and review the correlation between