Bitrate Control Method based on Quality and Data Volume for WebRTC Simulcast

In this paper, we propose a video bitrate control method based on the quality and transfer data amount for WebRTC simulcast. Web conferencing services are typically controlled on the basis of quality of service, such as network bandwidth, jitter, and packet loss. However, these methods have two problems. First, they may provide excessive quality to users in network conditions where sufficient throughput is available. Providing excessive quality increases the amount of data transfer and operational costs. Second, WebRTC simulcast generally sets a fixed multiple bitrate, making it inflexible to user environment changes such as device type and small network throughput fluctuations. To solve these problems, we propose a video bitrate control method for WebRTC simulcast that reduces the amount of data transfer while achieving the required quality on the basis of device type, network condition, and quality of each user. The proposed method is implemented and evaluated under several conditions. This method can suppress the excess quality to control the bitrate on the basis of the quality considering the device type. In addition, it was also found that the bandwidth can be effectively utilized, and the quality can be improved by flexible bitrate control even in environments where the network fluctuates.