Should WebRTC Prioritise Intelligibility over Speech Quality?

Network delay remains a challenge for real-time voice communication on the web. Jitter buffer algorithms have been widely deployed in popular platforms such as webRTC to reduce the impact of delay with playout adjustments. A trade off must be made between speech loss and voice degradations as adjustments can either drop segments resulting in a loss of speech intelligibility or change the rate of playout and impact the pitch or natural sound of the speech. Both options can negatively influence a listener's quality of experience (QoE). Optimising this trade-off requires knowledge of how intelligibility and quality are perceived and priorities when a listener syntheses both factors into a fused QoE judgement. This study conducted two subjective experiments to evaluate intelligibility and quality independently along with a short descriptive analysis to address the interplay between the two factors. The study uses a dataset that simulated listener-end speech under extreme but realistic network delay conditions using webRTC's standard jitter buffer and a variation that prioritised minimisation of packet loss. The results show that intelligibility is a key dimension in quality judgement for the scenarios tested. As a result, this study calls for attention when comparing the quality scores as the overlooked non-traditional quality attributes are proven to be actively contributing to the overall QoE. The descriptive analysis also indicates there is inconsistency in the interpretation of `quality' among the assessors. This finding questions the methodology used in standard QoE subjective experiment designs and proposes adopting a more flexible approach to measure subjective QoE.