@zitterorg/animi-accusantium-similique NPM

Overview

Runs in the browser and Node.js.
SIP over WebSocket (use real SIP in your web apps)
Audio/video calls (WebRTC) and instant messaging
Lightweight!
Easy to use and powerful user API
Works with OverSIP, Kamailio, Asterisk. Mobicents and repro (reSIProcate) servers (more info)
Written by the authors of RFC 7118 "The WebSocket Protocol as a Transport for SIP" and OverSIP

NOTE

Starting from 3.0.0, JsSIP no longer includes the rtcninja module. However, the @zitterorg/animi-accusantium-similique-rtcninja package is based on the 2.0.x branch, which does include rtcninja.

Support

For questions or usage problems please use the @zitterorg/animi-accusantium-similique public Google Group.
For bug reports or feature requests open an Github issue.

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 socket = new JsSIP.WebSocketInterface('wss://sip.myhost.com');
var configuration = {
  sockets  : [ socket ],
  uri      : 'sip:alice@example.com',
  password : 'superpassword'
};

var ua = new JsSIP.UA(configuration);

ua.start();

// Register callbacks to desired call events
var eventHandlers = {
  'progress': function(e) {
    console.log('call is in progress');
  },
  'failed': function(e) {
    console.log('call failed with cause: '+ e.data.cause);
  },
  'ended': function(e) {
    console.log('call ended with cause: '+ e.data.cause);
  },
  'confirmed': function(e) {
    console.log('call confirmed');
  }
};

var options = {
  'eventHandlers'    : eventHandlers,
  'mediaConstraints' : { 'audio': true, 'video': true }
};

var session = ua.call('sip:bob@example.com', options);

Want to see more? Check the full documentation at https://@zitterorg/animi-accusantium-similique.net/documentation/.