WebRTC is a technology for performing real-time video and audio communication in a browser, which can implement an audio/video call function between browsers or between a browser and a conventional communications terminal. For example, a video conference can be performed by using two different browsers supporting the WebRTC function. WebRTC technical specifications are specified by the Internet Engineering Task Force (IETF) and the World Wide Web consortium (W3C) together. However, an existing WebRTC technology does not support a WebRTC call transferring function, that is, a user cannot transfer a WebRTC call being performed on one terminal (for example, a personal computer or a mobile phone) to another terminal (for example, a mobile phone or a personal computer) in a seamless manner (that is, the call does not drop).
Some browsers may implement seamless transferring of content, such as a web page, a picture, a novel, a video, a text, and a telephone number, between two terminals of a same user. For example, a “cloud fly” function of the Sky Browser can implement instant “fly” of content, such as a web page, a picture, a novel, a video, a text, and a telephone number, between a personal computer (PC) and a mobile phone. By taking transferring a web page as an example, specifically, a same Sky cloud account is logged in at the PC end and the mobile phone end simultaneously, web page information requiring transferring is first sent to the cloud account, and is then pushed to the other terminal, and the other terminal, after receiving the web page information sent by the cloud account, automatically opens the web page, thereby implementing the seamless transferring.
The seamless transferring function of an existing browser cannot support transferring of a WebRTC call being performed between two terminals of the same user, and therefore, the transferring of a WebRTC call between terminals cannot be implemented currently.