- 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
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:[email protected]',
'password': 'superpassword'
};
var coolPhone = new JsSIP.UA(configuration);
coolPhone.start();
// Make an audio/video call:
var useAudio = true;
var useVideo = true;
var views = {
'selfView': document.getElementById('my-video'),
'remoteView': document.getElementById('peer-video')
};
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:[email protected]', useAudio, useVideo, eventHandlers, views);Want to see more? Check the full Getting Started section in the project website.
Check our Tryit JsSIP online demo:
- José Luis Millán ([email protected] | github | twitter)
- Iñaki Baz Castillo ([email protected] | github | twitter)
- Saúl Ibarra Corretgé ([email protected] | github | twitter)
JsSIP is released under the MIT license.