WebRTC Cheatsheet

Tim · 2023-05-21T22:37:41Z

🌐 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

#realtimecommunication #videochat #webrtc #peertopeer

🌐 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.