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https://github.com/vmolsa/rtc-stream

Webrtc P2P through regular NodeJS stream
https://github.com/vmolsa/rtc-stream

Last synced: about 1 month ago
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Webrtc P2P through regular NodeJS stream

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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