Overview

• SIP over WebSocket (use real SIP in your web apps)
• Audio/video calls (WebRTC), instant messaging and presence
• Lightweight! (~140KB)
• Easy to use and powerful user API
• Works with OverSIP, Kamailio and Asterisk servers (more info)
• Written by the authors of draft-ietf-sipcore-sip-websocket and OverSIP

Getting Started

The following simple JavaScript code creates a JsSIP User Agent instance and makes a SIP call:

// Create our JsSIP instance and run it:

var configuration = {
  'outbound_proxy_set': 'ws://sip-ws.example.com',
  'uri':                'sip:alice@example.com',
  'password':           'superpassword'
};

var coolPhone = new JsSIP.UA(configuration);

coolPhone.start();


// Make an audio/video call:

var useAudio = true;
var useVideo = true;

// id attribute of existing HTML5 <video> elements in which local and remote video will be shown
var views = {
  'localView':  "my-cam",
  'remoteView': "peer-cam"
};

var eventHandlers = {
  'connecting': function(e){ // Your code here },
  'progress':   function(e){ // Your code here },
  'failed':     function(e){ // Your code here },
  'started':    function(e){ // Your code here },
  'ended':      function(e){ // Your code here }
};

coolPhone.call('sip:bob@example.com', useAudio, useVideo, eventHandlers, views);

Want to see more? Check the full Getting Started section in the project website and our nice demos.

Website and Documentation

• jssip.net

Download

• jssip.net/download

Authors

Main Author

• José Luis Millán (jmillan@aliax.net | github | twitter)

Contributors

• Iñaki Baz Castillo (ibc@aliax.net | github | twitter)
• Saúl Ibarra Corretgé (saghul@gmail.com | github | twitter)

License

JsSIP is released under the MIT license.