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

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

Last synced: 4 months 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 (over 6 years ago)
• Last Synced: 2024-01-30T05:21:04.151Z (5 months ago)
• Language: JavaScript
• Size: 32.2 KB
• Stars: 14
• Watchers: 5
• Forks: 6
• Open Issues: 0
• Metadata Files:
  • Readme: README.md
  • License: LICENSE

Lists

• awesome-peer-to-peer - rtc-stream
• awesome-peer-to-peer - rtc-stream
• awesome-peer-to-peer - rtc-stream

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