In general, a WebRTC- enabled application needs to:
Obtain an audio, video or other data stream or gather network information (e.g., IP addresses and ports) and exchange these with other WebRTC clients;
Error reporting, session initiation and closing are established with the help of signalling communication. The clients looking to use the WebRTC applications must exchange information about media, such as resolution and codecs, stream the audio, video or data.
WebRTC implements three APIs:
Media Stream - allows the client (e.g. the web browser) to access the stream, such as the one from a WebCam or microphone;
RTC PeerConnection - enable audio or video data transfer, with support for encryption and bandwidth management;
RTCDataChannel - enables peer-to-peer communication for any generic data.
We can deliver complex WebRTC solutions which include one to many, many to many audio/video broadcasting
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