dtolb/jssip.md

## jssip.md

      
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              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:sip-user@your-domain.bwapp.bwsip.io', 
  '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:sip-user@your-domain.bwapp.bwsip.io', 
  '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:sip-user@your-domain.bwapp.bwsip.io', 
  '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.