Research Of Real-Time Adaptive Transmission Technology Based On WebRTC

With the fast development of Internet technology, real-time video adaptive transmission applications based on WebRTC has become a hot research area for many developers and users because WebRTC is easy to use and maintain. Nowadays, the Internet based on TCP/IP currently provides a best-effort service model, local network congestion often occurs, which will lead to the degradation of the QoS of the businesses.Unstable internet environments may often cause network congestions,and lead to a bad user experience during real-time video communications.This paper focus on researching real-time communication technology based on WebRTC. We use open source code of WebRTC to build a simulation platform of video adaptive transmission system,optimize the algorithm of the adaptive transmission module, and finally use the simulation platform to verify the improvement of the performance of the optimization.Firstly, this paper research deeply on the basic technology of WebRTC system, analyze the overall framework and communication model of WebRTC. Additionally, we investigate the core module of WebRTC, such as the application program interface, signaling process and date channel etc. Use the relative modules to build a P2P video telephone Demo and network emulators to set the network environment,and finally establish a simulation platform of WebRTC.Secondly, research deeply on the algorithm of the video adaptive transmission technology of WebRTC system, especially on the bitrate control module of WebRTC. Use the simulation platform to research the performance of the bitrate control module.The research shows that,because of the unstable estimation of the bitrate, the video received may has a decline on the video quality. In order to optimize this situation, Try to use the bandwidth change detection algorithm to monitor bandwidth changes, and lock the upper limit of the original bitrate according to the bandwidth changes.Finally, implement the optimized algorithm into WebRTC and use the simulation platform to verify the performance of the optimized algorithm. The test shows that, the optimized bitrate estimation module can output more stable bitrate in the network environment which bandwidth is stable, then get higher quality video and improve the watching experience of the video.