WebRTC using JSON via XMLHttpRequest and SIP over WebSocket: initial signalling overhead findings

Abstract

Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This position paper discusses the two common methods of doing real-time communication in Web browsers through WebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTCJSON/XHR. This signalling overhead findings are useful in informing the WebRTC working groups in terms of additional overhead introduced by proposed WebRTC methods, the finding could also help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.