🌐 WebRTC Basics:

• WebRTC stands for Web Real-Time Communication.
• It is an open framework for enabling real-time communication in web browsers and mobile applications.
• WebRTC allows peer-to-peer audio, video, and data sharing without the need for additional plugins or software.

📹 Video Communication:

• getUserMedia() API captures video from the user's camera.
• MediaStream represents the video and audio streams.
• RTCPeerConnection establishes a direct peer-to-peer connection for video communication.
• RTCDataChannel enables real-time data sharing alongside video communication.

🔊 Audio Communication:

• getUserMedia() API captures audio from the user's microphone.
• MediaStream represents the audio stream.
• RTCPeerConnection establishes a direct peer-to-peer connection for audio communication.
• RTCDataChannel enables real-time data sharing alongside audio communication.

💡 Useful JavaScript APIs:

• getUserMedia(): Grants access to the user's camera and microphone.
• RTCPeerConnection: Handles peer-to-peer communication.
• RTCDataChannel: Enables real-time data sharing between peers.
• MediaStream: Represents audio and video streams.

🔒 Security and Encryption:

• WebRTC uses Secure Real-Time Transport Protocol (SRTP) for encryption.
• Signaling is required to exchange session information and establish a connection.
• Signaling servers help coordinate the communication process but are not part of the WebRTC standard.

🌐 WebRTC Frameworks and Libraries:

• SimpleWebRTC
• PeerJS
• Socket.io
• Twilio Programmable Video
• Janus Gateway

🚀 Deploying WebRTC:

• Cloud hosting platforms like Firebase, AWS, or Heroku can be used to deploy WebRTC applications.
• Consider the server-side requirements for signaling, TURN, and STUN servers.

📚 Resources:

• WebRTC API documentation: https://webrtc.org/
• WebRTC samples and demos: https://antmedia.io/webrtc-samples/
• WebRTR Live Streaming Software: https://github.com/ant-media