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https://github.com/pchalupa/ws-webrtc
WebRTC
https://github.com/pchalupa/ws-webrtc
webrtc workshop
Last synced: about 2 months ago
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WebRTC
- Host: GitHub
- URL: https://github.com/pchalupa/ws-webrtc
- Owner: pchalupa
- Created: 2022-09-02T14:42:13.000Z (over 2 years ago)
- Default Branch: main
- Last Pushed: 2023-07-19T21:55:35.000Z (over 1 year ago)
- Last Synced: 2024-10-28T13:45:54.495Z (4 months ago)
- Topics: webrtc, workshop
- Language: TypeScript
- Homepage:
- Size: 423 KB
- Stars: 1
- Watchers: 1
- Forks: 0
- Open Issues: 3
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Metadata Files:
- Readme: README.md
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README
# WebRTC – Cheat Sheet
Exchange real-time audio, video and data with one or more peers entirely in the browser.
> Imagine a world where your phone, TV, and computer could communicate on a common platform. Imagine it was easy to add video chat and peer-to-peer data sharing to your web app. That's the vision of WebRTC.
## Abberations
| Name | Description |
| ---- | ------------------------------------------------------------------------------------------------------------------------- |
| STUN | "Session Traversal Utilities for NAT" is server which is used to get your device IP address |
| TURN | "Traversal using Relay NAT" is server which is used as fallback when peer-to-peer connection fails |
| SDP | "Session Description Protocol" Contain info about the session connection such as the codec, media type, audio, video, etc |
| ICE | "Interactive Connectivity Establishment" network path between peers |
## RTCPeerConnection
The RTCPeerConnection interface represents a WebRTC connection between the local computer and a remote peer. It provides methods to connect to a remote peer, maintain and monitor the connection, and close the connection once it's no longer needed.
### Events
| Name | Description |
| ----------------------------------------------------------------------------------------------------------------------- | ---------------------------------------------------- |
| [connectionstatechange](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionstatechange_event) | Fires when connection state changes |
| [negotiationneeded](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/negotiationneeded_event) | Fires when negotiating of the connection is required |
| [icecandidate](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/icecandidate_event) | Fires when ICE candidate is added to local peer |
| [datachannel](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/datachannel_event) | Fires when data channel is added to the connection |
| [track](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/track_event) | Fires when new track is added to the connection |
### Properties
| Name | Description |
| ----------------------------------------------------------------------------------------------------- | -------------------------------------------------- |
| [connectionState](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState) | Indicates the current state of the peer connection |
## RTCDataChannel
The RTCDataChannel interface represents a network channel that can be used for bidirectional peer-to-peer transfers of arbitrary data. Every data channel is associated with an RTCPeerConnection, and each peer connection can have up to a theoretical maximum of 65,534 data channels (the actual limit may vary from browser to browser).
### Events
| name | description |
| ---------------------------------------------------------------------------------------- | -------------------------------------- |
| [message](https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/message_event) | Fires when a message has been received |