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Table of Contents

Audio SIP call in one click example

This example allows to place outgoing audio call with one button click using account specified in server config file

/usr/local/FlashphonerWebCallServer/conf/apps/click-to-call/accounts.xml

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Code of the example

The path to the source code of the example on WCS server is:

/usr/local/FlashphonerWebCallServer/client/examples/demo/sip/click-to-call

click-to-call.css - stylesheet for the example page
click-to-call.html - page of the example
click-to-call.js - script providing functionality for the example

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Here host is the address of the WCS server.

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Analyzing the code

To analyze the code, let's take the version of file click-to-call.js with hash 26fc5afa246baea80867fa011305da811edab1f002c8028, which is available here and  and can be downloaded with corresponding build 2.0.5178.5.1894.

1. Initialization of the API. line 11API is initialized after loading the page. For Flash support, the path to SWF file is passed to the init() method.

Flashphoner.init() code

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    try {
        Flashphoner.init();
    } catch(e) {
        $("#notifyFlash").text("Your browser doesn't support WebRTC technology needed for this example");
        return;
    }

2. Connection to server.

createSession() code

Object with connection options is passed to the method when session is created

  • urlServer - URL for WebSocket connection to WCS server
  • sipOptions - SIP connection parameters from a hidden form fields
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Flashphoner.init({flashMediaProviderSwfLocation: '../../../../media-provider.swf'});

2. Connection to server. line 41

...

        var url = $('#urlServer').val();
        var sipOptions = {
            login: $('#sipLogin').val(),
            authenticationName: $('#sipAuthenticationName').val(),
            password: $('#sipPassword').val(),
            domain: $('#sipDomain').val(),
            outboundProxy: $('#sipOutboundProxy').val(),
            port: $('#sipPort').val(),
            registerRequired: true
        };
    
        var connectionOptions = {
            urlServer: url,
            sipOptions: sipOptions
        };

        //create session
        console.log("Create new session with url " + url);
        Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session){
            ...
        });

3. Receiving the event confirming successful connection

ConnectionStatusEvent ESTABLISHED code

On this event, outgoing call is created

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        Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.REGISTEREDESTABLISHED, function(session){
		    setStatus("Session", SESSION_STATUS.REGISTEREDESTABLISHED);
    			//session connected, place call
		    call(session);
	    }).on(SESSION_STATUS.DISCONNECTED, function(){
    setStatus("Session", SESSION_STATUS.DISCONNECTED);
        ...
	    onHangup();
}).on(SESSION_STATUS.FAILED, function(){
      setStatus("Session", SESSION_STATUS.FAILED);
      onHangup();
...
		});

Session is created with method createSession(). Callback function, which will be called to place outgoing call in case of successfully established connection (status SESSION_STATUS.REGISTERED), is added.

Object with connection options is passed to the method when session is created

  • urlServer - URL for WebSocket connection to WCS server
  • appKey - internal server-side application 'clickToCallApp'

Parameters for SIP connection are taken from config accounts.xml.

3. Outgoing call. line 62

New call is created with method session.createCall().4. Outgoing call.

session.createCall(), call() code

The following parameters are passed when call is created

  • callee - callee SIP username
  • visibleName - display name
  • localVideoDisplay - <div> element for local display (will be used for Flash Player settings dialog in case of Flash media provider)
  • remoteVideoDisplay - <div> element for remote audio
  • constraints - constraints for the call (in this case 'video' is set to 'false' to call with audio only - line 56)
  • receiveAudio - set to true to receive audio

  • receiveVideo - set to false to receive audio only

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    var constraints = {
        audio: true,
        video: false
};
  • receiveAudio - set to 'true' to receive audio
  • receiveVideo - set to 'false' to receive audio only
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    };
	
	//prepare outgoing call 
    var outCall = session.createCall({
    		callee: $("#callee").val(),
    		visibleName: "Click To Call",
    		localVideoDisplay: localDisplay,
    		remoteVideoDisplay: remoteDisplay,
    		constraints: constraints,
    		receiveAudio: true,
        receiveVideo: false
}).on(CALL_STATUS.RING, function(){
       setStatus("Call", CALL_STATUS.RING);
}).on(CALL_STATUS.ESTABLISHED, function(){
    setStatus("Call", CALL_STATUS.ESTABLISHED);
}).on(CALL_STATUS.FINISH,  ...
	});
	
	outCall.call();

5. Call hangup

call.hangup() code

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    $("#callBtn").text("Hangup").removeClass("btn-success").addClass("btn-danger").off('click').click(function(){
     setStatus("Call", CALL_STATUS.FINISH   $(this).prop('disabled', true);
	    onHangupoutCall.hangup();
    });

When call is created, callback functions for events CALL_STATUS.RING, CALL_STATUS.ESTABLISHED, CALL_STATUS.FINISH are added to make appropriate changes in controls of the interface.

Outgoing call is placed with method call(). line 79

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outCall.call();

4. Call hangup. line 83

Method hangup() is used to hang up call.

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outCall.hangup(.prop('disabled', false);