HTML5 Real Time Communication (RTC)
What is RTC?
There are three API's which are implemented by the RTC:
- RTC was introduced by the World Web Consortium (W3C).
- RTC is abbreviated as Real Time Communication.
- It supports the browser for applications like voice calling, video chat and P2P file sharing.
It gives access to the user's camera and microphone.
It gives access to the audio and video calling facility.
it gives access for peer to peer communication.
- Synchronized streams of media are represented by the MediaStream.
- It contains the stream.getAudioTracks() and the stream.VideoTracks().
- An empty array is returned and it will also check the video stream if there are no audio tracks.
- If there is a webcam connected to the stream.getVideoTracks() method then it will return an array of the MediaStreamTrack which represents the stream from the webcam.
- Screen capturing can be done in the chrome browser with the mediaStreamSource which requires HTTPS.
Session Control, Network & Media Information
- Peer-to-peer communication is required between the browsers by the Web RTC.
- Mechanisms such as signaling, network information, session control and media information.
- Any mechanism can be chosen by the web developer to communicate between browsers such as SIP or XMPP or two way communications.