https://github.com/node-webrtc/node-webrtc-examples
MediaStream and RTCDataChannel examples using node-webrtc
https://github.com/node-webrtc/node-webrtc-examples
Last synced: about 1 year ago
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MediaStream and RTCDataChannel examples using node-webrtc
- Host: GitHub
- URL: https://github.com/node-webrtc/node-webrtc-examples
- Owner: node-webrtc
- Created: 2019-03-26T04:49:05.000Z (about 7 years ago)
- Default Branch: master
- Last Pushed: 2022-09-09T22:17:23.000Z (almost 4 years ago)
- Last Synced: 2025-04-01T09:36:08.566Z (about 1 year ago)
- Language: JavaScript
- Size: 39.1 KB
- Stars: 524
- Watchers: 24
- Forks: 166
- Open Issues: 25
-
Metadata Files:
- Readme: README.md
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README
node-webrtc-examples
====================
This project presents a few example applications using node-webrtc.
- [audio-video-loopback](examples/audio-video-loopback): relays incoming audio
and video using RTCRtpTransceivers.
- [ping-pong](examples/ping-pong): simple RTCDataChannel ping/pong example.
- [pitch-detector](examples/pitch-detector): pitch detector implemented using
RTCAudioSink and RTCDataChannel.
- [sine-wave](examples/sine-wave): generates a sine wave using RTCAudioSource;
frequency control implemented using RTCDataChannel.
- [sine-wave-stereo](examples/sine-wave-stereo): generates a stereo sine wave
using RTCAudioSource; panning control implemented using RTCDataChannel.
- [video-compositing](examples/video-compositing): uses RTCVideoSink,
[node-canvas](https://github.com/Automattic/node-canvas), and RTCVideoSource
to draw spinning text on top of an incoming video.
- [record-audio-video-stream](examples/record-audio-video-stream) using [FFmpeg](https://www.ffmpeg.org) and RTCVideoSink.
- Broadcast example with one [broadcaster](examples/broadcaster) forwarding to many [viewers](examples/viewer).
Usage
-----
Install the project's dependencies and run the tests before starting the
application server:
```
npm install
npm test
npm start
```
Then, navigate to [http://localhost:3000](http://localhost:3000).
Architecture
------------
Each example application under [examples/](examples) has a Client and Server
component. RTCPeerConnection negotiation is supported via a REST API (described
below), and is abstracted away from each example application. Code for
RTCPeerConnection negotiation lives under [lib/](lib).
### RTCPeerConnection Negotiation
RTCPeerConnections are negotiated via REST API. The Server always offers (with
host candidates) and the Client always answers. In order to negotiate a new
RTCPeerConnection, the Client first POSTs to `/connections`. The Server responds
with an RTCPeerConnection ID and SDP offer. Finally, the Client POSTs an SDP
answer to the RTCPeerConnection's URL.
```
Client Server
| |
| POST /connections |
| |
|---------------------------------------------------->|
| |
| 200 OK |
| { "id": "$ID", "localDescription": "$SDP_OFFER" } |
| |
|<----------------------------------------------------|
| |
| POST /connections/$ID/remote-description |
| $SDP_ANSWER |
| |
|---------------------------------------------------->|
| |
| 200 OK |
| |
|<----------------------------------------------------|
```
### RTCPeerConnection Teardown
RTCPeerConnections can be proactively torn down by sending a DELETE to the
RTCPeerConnection's URL; otherwise, ICE disconnection or failure, if unresolved
within the `timeToReconnect` window, will also trigger teardown. The default
`timeToReconnect` value is 10 s.
```
Client Server
| |
| DELETE /connections/$ID |
| |
|---------------------------------------------------->|
| |
| 200 OK |
| |
|<----------------------------------------------------|
```