JsSip Demo

jssip.md

JSSIP with Catapult API

⚠️ This has been updated at JsFiddle*

Prerequisites

​

• Register for a Catapult (Bandwidth Application Platform) account here
• Register a SIP domain
• Create an endpoint/user
• If you want to make calls to the PSTN (normal phones) you will need a server to handler events from Catapult
• Make phone calls ​ For a more in depth guide, view this article ​

Quick Start

Full docs are here ​ CDN hosted library: minified not-minified ​ JSFIDDLE Demo Client ​

Outbound Call

var callOptions = {
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};
​
var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'password': 'password', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443'
});
bwPhone.start();
​
bwPhone.on("registered", function(){
    bwPhone.call("222-333-4444", callOptions);    
});
​
bwPhone.on("newRTCSession", function(data){
    var session = data.session; // outgoing call session here
    var dtmfSender;
    session.on("confirmed",function(){
        //the call has connected, and audio is playing
        var localStream = session.connection.getLocalStreams()[0];
        dtmfSender = session.connection.createDTMFSender(localStream.getAudioTracks()[0])
    });
    session.on("ended",function(){
        //the call has ended
    });
    session.on("failed",function(){
        // unable to establish the call
    });
    session.on('addstream', function(e){
        // set remote audio stream (to listen to remote audio)
        // remoteAudio is <audio> element on page
        remoteAudio.src = window.URL.createObjectURL(e.stream);
        remoteAudio.play();
    });
    
    //play a DTMF tone (session has method `sendDTMF` too but it doesn't work with Catapult server right)
    dtmfSender.insertDTMF("1");
    dtmfSender.insertDTMF("#");
​
    //mute call
    session.mute({audio: true});
​
    //unmute call
    session.unmute({audio: true});
​
    //to hangup the call
    session.terminate();
​
    
});
​

Inbound Call

var callOptions = {
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};
​
var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'password': 'password', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443'
});
bwPhone.start();
​
bwPhone.on("newRTCSession", function(data){
    var session = data.session; 
    
    if (session.direction === "incoming") {
        // incoming call here
        session.on("accepted",function(){
            // the call has answered
        });
        session.on("confirmed",function(){
            // this handler will be called for incoming calls too
        });
        session.on("ended",function(){
            // the call has ended
        });
        session.on("failed",function(){
            // unable to establish the call
        });
        session.on('addstream', function(e){
            // set remote audio stream (to listen to remote audio)
            // remoteAudio is <audio> element on page
            remoteAudio.src = window.URL.createObjectURL(e.stream);
            remoteAudio.play();
        });
        
        // Answer call
        session.answer(callOptions);
        
        // Reject call (or hang up it)
        session.terminate();
    }
});

Passwordless connection (via auth token)

​
var authToken = "1234567890"; // you can get this token by POST http request to /v1/users/<userId>/domains/<domainId>/endpoints/<endpointId>/tokens
​
var authHeader = "X-Callsign-Token: " + authToken;
​
var callOptions = {
  extraHeaders: [authHeader], // set auth token here (it will be passed on making calls and answering incoming call) 
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};
​
var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443',
});
bwPhone.registrator().setExtraHeaders([authHeader]); // set auth header on register
​
bwPhone.on('registered', function(){
     // ready to make calls and receive incoming calls
     // making a call
     bwPhone.call("222-333-4444", callOptions);
});
​
bwPhone.on("newRTCSession", function(data){
    var session = data.session; 
    
    if (session.direction === "incoming") {
        // answer incoming call
        session.answer(callOptions);
    }
});
​
bwPhone.start();
​

​ ​

Supported Browsers

• Firefox
• Chrome
• Opera ​ You can check if browser supports WebRTC by code ​

    if (!JsSIP.rtcninja.hasWebRTC()) {
        alert("WebRTC is not supported. Please use another browser.");
    }

Thank you

• Thank you, @GaborTorma for updating and answering questions.

Generally it is codec, NAT or route problem.

I want to made audio call
the attached log after successfully call made and pickup