JS Audio/Video Streaming: WebRTC APIs 101 explained
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Javascript WebRTC APIs explained to make audio/video real-time communications: you will learn how to implement your video conference application from scratch. We will start from the basics, giving an overview of what WebRTC is and then we will explore the fundamental Javascript APIs to create performant real-time applications, no prior knowledge is required. WebRTC is used by all video conference platforms today and we will go under the hood to analyze it from a Javascript perspective: at the end, you will be able to easily develop your video conference application from scratch.
JS Audio/Video Streaming: WebRTC APIs 101 explained
- 1. JS Audio/Video Streaming WebRTC API 101 explained ale_polidori alepolidori
- 3. webrtc.org
- 5. Our Purpose Whats WebRTC How it works 3 Javascript APIs Video Conference App Resources 鐃緒申
- 7. WebRTC Web Real-Time Communication to the WEB
- 8. open-source protocols, JS Web APIs, web Browsers, mobile app
- 11. How WebRTC Works
- 12. How WebRTC Works 1 2 1-2 offer request to start a call
- 13. How WebRTC Works 1 2 1-2 offer request to start a call 3-4 answer to the offer 3 4
- 14. How WebRTC Works 1 2 1-2 offer request to start a call 3-4 answer to the offer 5 peer-to-peer communications 3 4 5
- 15. How WebRTC Works 1-2 offer request to start a call 3-4 answer to the offer 5 peer-to-peer communications
- 16. 1 2 1-2 Offer request to start a call 3-4 Answer to the offer 5 peer-to-peer communications 3 4 5 Signaling Signaling Server
- 18. 3 APIs getUserMedia RTCPeerConnection RTCDataChannel
- 25. Security
- 29. 3 APIs getUserMedia RTCPeerConnection RTCDataChannel
- 30. 1 2 1-2 Offer request to start a call 3-4 Answer to the offer 5 peer-to-peer communications 3 4 5 Signaling Signaling Server
- 31. Signaling Server 1. new RTCPeerConnection 2. get & share Network Info (ICE candidate) 3. get & share local/remote description (SDP) 1. new RTCPeerConnection 2. get & share Network Info (ICE candidate) 3. get & share local/remote description (SDP)
- 32. Demo Time
- 33. Protocols Recap HTTPS RTP, SRTP: media exchange SDP: media description ICE: peer-to-peer connectivity WebSocket: custom signaling (alternative: SIP)
- 35. Resources CityJS Talk: github.com/alepolidori/cityjs-athens-2023 Samples: https://webrtc.github.io/samples/ DOCs https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API Get support https://groups.google.com/forum/#!forum/discuss-webrtc Standard W3C & IETF https://www.w3.org/TR/webrtc/ Debug chrome://webrtc-internals