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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
Updated on: 28-Jan-2020
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