Video conferencing applications have significantly changed the way in which people communicate over the Internet. Web real-time communication (WebRTC), drafted by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF), has added new functionality to web browsers, allowing audio/video calls between browsers without the need to install any video telephony applications. The Google Congestion Control (GCC) algorithm has been proposed as WebRTC’s receiver congestion control mechanism, but its performance is limited due to using a fixed incoming rate decrease factor, known as an alpha (α). In this thesis, we have proposed a dynamic alpha model to reduce the receiving bandwidth estimate during overuse, as indicated by the overuse detector. Experiments using our specific testbed show that our proposed model achieves a higher incoming rate and a lower Round-Trip Time while slightly increasing the packet loss rate in some cases compared to fixed alpha model. Our mathematical model proves that it is necessary to use an adaptive alpha α as the receiver side controller. The experimental results show improvement in the term of incoming rate, Round-Trip Time, and packet fraction loss rate in some cases. Our model increases the amount of incoming rate and decreases Round-Trip Time and fraction loss.