As I described in the standardization post, the model used in WebRTC for real-time, browser-based applications does not envision that the browser will contain all the functions needed to function as a telephone or video conferencing unit. Instead, is specifies the browser will contain the functions that are needed to run a Web application which would work in conjunction with back-end servers to implement telephony functions as required. According to this, WebRTC is meant to implement the media plane but to leave the signalling plane up to the application. Different applications may prefer to use different protocols, such as SIP or something custom to the particular application. In this approach, the key information that needs to be exchanged is the multimedia session description, which specifies the configuration necessary to establish the media plane. In other words, WebRTC does not specify a particular signalling model other than the generic need to exchange SDP media descriptions in the offer/answer fashion. However, the browser is totally decoupled from the actual mechanism by which these offers and answers are communicated to the remote side.