Performance evaluation of WebRTC-based online consultation platform

Abstract

Information technologies give patients the opportunity to communicate with medical professionals remotely. Telemedicine uses these technologies to provide advanced healthcare and medical services. We present a medical online consultation application based on Web Real-Time Communications (WebRTC) technology enabling chat, audio, and video calls. Communication architecture and protocols of the application are explained in detail. Additionally, the user interface of the application is shown via performed calls. The application is tested and evaluated on different network connections (3G, 4G, local, and DSL) and different browsers and mobile operating systems (Android, Chrome, Firefox, Internet Explorer, iOS, Opera, Safari). During calls, communication quality parameters such as round-trip time (RTT) and packet loss, obtained via the WebRTC application programming interface, are analyzed. 3G, 4G, and local connections show low packet losses (<1%). Packet losses are high (>1%) in Android, Chrome, iOS, Opera, and Safari for DSL connection, but RTT values are low (<100 ms) in all different conditions excluding iOS. In the presented application, RTT and packet loss remain lower than 100 ms and 1%, respectively, in various scenarios, indicating good communication quality. RTT and packet loss are related to total time and hang time parameters, which describe the necessary time to establish and to end a call. It is shown that communication quality of the application can simply be measured by analyzing the total time parameter. This enables predictable information for communication quality for WebRTC-based applications without continuously monitoring RTT and packet loss for the first time.