What is Web RTC?

Web RTC introduced by World Wide Web Consortium (W3C). That supports browser-to-browser applications for voice calling, video chat, and P2P file sharing.

Web RTC implements three API's as shown below −

• MediaStream − get access to the user's camera and microphone.
•  RTCPeerConnection − get access to audio or video calling facility.
•  RTCDataChannel − get access to peer-to-peer communication.

Web RTC required peer-to-peer communication between browsers. This mechanism required signaling, network information, session control and media information. Web developers can choose a different mechanism to communicate between the browsers such as SIP or XMPP or any two-way communications