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https://github.com/vmolsa/rtc-stream
Webrtc P2P through regular NodeJS stream
https://github.com/vmolsa/rtc-stream
Last synced: 9 days ago
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Webrtc P2P through regular NodeJS stream
- Host: GitHub
- URL: https://github.com/vmolsa/rtc-stream
- Owner: vmolsa
- License: mit
- Archived: true
- Created: 2015-03-15T20:48:24.000Z (over 9 years ago)
- Default Branch: master
- Last Pushed: 2018-01-26T03:06:06.000Z (almost 7 years ago)
- Last Synced: 2024-10-13T11:07:41.495Z (26 days ago)
- Language: JavaScript
- Size: 32.2 KB
- Stars: 14
- Watchers: 5
- Forks: 6
- Open Issues: 0
-
Metadata Files:
- Readme: README.md
- License: LICENSE
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README
# WebRTC Stream
Webrtc P2P through regular NodeJS stream
# Client Example
``````````
var net = require('net');
var rtcStream = require('rtc-stream');
var socket = net.connect({ host: 'localhost', port: 1337 }, function() {
socket.setNoDelay(true);
var rtc = new rtcStream(socket);
rtc.addStun('stun.l.google.com:19302');
rtc.on('end', function() {
console.log('Peer closed :(');
});
rtc.on('error', function(error) {
console.log('Peer error:', error.toString());
});
rtc.on('open', function() {
console.log('Peer Connected :)');
});
rtc.createChannel('echo', function(channel) {
channel.setEncoding('utf8');
channel.on('data', function(data) {
console.log(data);
channel.end();
});
channel.on('error', function(error) {
console.log('Channel error:', error.toString());
});
channel.on('end', function() {
console.log('Channel closed :(');
rtc.end();
});
channel.write('HELLO SERVER!\n');
});
});
``````````
# Server Example
``````````
var net = require('net');
var rtcStream = require('rtc-stream');
var server = net.createServer(function(socket) {
socket.setNoDelay(true);
var rtc = new rtcStream(socket);
rtc.addStun('stun.l.google.com:19302');
rtc.on('end', function() {
console.log('Peer closed :(');
});
rtc.on('error', function(error) {
console.log('Peer error:', error.toString());
});
rtc.on('open', function() {
console.log('Peer Connected :)');
});
rtc.on('channel', function(channel) {
channel.pipe(process.stdout);
channel.on('error', function(error) {
console.log('Channel error:', error.toString());
});
channel.on('end', function() {
console.log('Channel closed :(');
rtc.end();
});
channel.write('EHLO FROM SERVER!\n');
});
socket.on('end', function() {
console.log('Disconnected...');
});
});
server.listen({
host: 'localhost',
port: 1337,
});
``````````
# Examples
https://github.com/vmolsa/rtc-stream/tree/master/examples
# Prototype
``````````
var rtc = new rtcStream([stream]) // returns Stream object
rtc.write(data, encoding, callback)
rtc.end(data, encoding, callback)
rtc.addStun('url', 'username', 'password')
rtc.addTurn('url', 'username', 'password')
rtc.useAudio(boolean);
- enables / disables for receiving audio stream
- default: disabled
- on media webrtc connection: enabled
rtc.useVideo(boolean);
- enables / disables for receiving video stream
- default: disabled
- on media webrtc connection: enabled
rtc.onChannel('channelName', callback(channel))
rtc.offChannel('channelName');
rtc.createMedia([options], [callback(media)])
- Opens another webrtc stream by using primary webrtc connection for signaling.
- defaults on media is useAudio(true) and useVideo(true)
rest of settings are inherits with primary webrtc stream.
rtc.createChannel(['channelName'], [callback(channel)], [timeout])
- Opens webrtc datachannel
- on success 'channel' / callback is called with channel object.
- on error media / primary webrtc stream 'error' event is called with Error object and
callback is called with null.
rtc.createStream([options], [callback(stream)])
- https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia
- on success 'stream' / callback is called with stream object.
- on error media / primary webrtc stream 'error' event is called with Error object and
callback is called with null.
rtc.addStream(stream)
- add audio / video stream to webrtc.
- both ends must have audio or video enabled using useAudio(true) / useVideo(true)
rtc.removeStream(stream)
- removes stream from webrtc connection
rtc.getLocalStreams()
- returns array of active audio / video local streams.
rtc.getRemoteStreams()
- returns array of active audio / video remote streams.
``````````
# Events
``````````
'error', callback(error)
'close', callback()
'end', callback()
'media', callback(media)
'channel', callback(channel)
``````````
# Datachannel Prototype
http://nodejs.org/api/stream.html
# License
MIT