Optimizing Video Conferencing Web Application built on WebRTC

Web RTC can provide real time capabilities for multimedia applications like voice, video and data between peers by utilizing the open standards. With the onset of covid, video conferencing has become a need of the day. Optimization of bandwidth, and other features have become the necessity. In the current work, WebRTC protocols are built upon, to improve the connection and success rate, optimize the bitrate and reduce the frame rate. This improvement is carried out without visible or audible loss of clarity in the video sessions. The Session Description Protocol is utilized to accomplish this, and this would not have been possible using WebRTC APIs alone. N-to-N connection among peers is established in an optimized manner, so that the application does not engage an intermediate server to transfer media streams which has resulted in multi-fold improvement in bandwidth performance and also maximized the number of participants, without incurring the cost for an intermediate media server. Conventionally, an intermediate media server is used to stitch streams from various senders into a single stream and then sent to the receivers. Bandwidth utilization is reduced close to 100x with good visibility in the stream. Robust web application is achieved using the TURN (Traversal Using Relays around NAT) server. The proposed work has addressed multiple ways of optimizing for the video conferencing using WebRTC.