Quality-based Video Bitrate Control for WebRTC-based Teleconference Services

Abstract

In this article, we propose a quality-based video bitrate control method for web real-time communication (WebRTC)-based teleconferences. Video bitrate is controlled on the basis of quality of service (QoS) parameters such as delay and packet-loss rate in WebRTC. Therefore, the amount of transferred data may increase because media streams are transmitted at excessive quality levels depending on QoS conditions (e.g., the jitter and packet-loss rate are low). An increase in transferred data leads to higher operational cost (i.e., data transferred cost) and affects profitable growth. In the proposed method, quality desired by a service provider is set as TargetQuality, and the video bitrate of each stream is controlled aiming at TargetQuality, thereby suppressing the amount of transferred data while maintaining sufficient quality. The proposed method is implemented to an actual teleconference system and is evaluated in terms of its effect at reducing the amount of transferred data. The results show that the amount of transferred data can be reduced by more than 40% by setting the value of TargetQuality appropriately.