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WebRTC Cheatsheet

🌐 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:

🚀 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:

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umtcylmz profile image
umtcylmz

Very useful information regarding WebRTC. Thanks!