Web SDK 2.0 - EN
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Example of web phone for video calls

Code of the example

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

/usr/local/FlashphonerWebCallServer/client/examples/demo/sip/phone-video

phone-video.css - file with styles
phone-video.html - page of the web phone
call-fieldset.html - form with fields required for connection
call-controls.html - HTML code for call controls
phone-video.js - script providing functionality for the web phone

This example can be tested using the following address:

https://host:8888/client/examples/demo/sip/phone-video/phone-video.html

Here host is the address of the WCS server.

Analyzing the code

To analyze the code, let's take the version of file phone-video.js with hash ecbadc3, which is available here and can be downloaded with corresponding build  2.0.212.

1. Initialization of the API

Flashphoner.init() code

        Flashphoner.init();

2. Connection to server.

Flashphoner.createSession() code

Object with connection options is passed to the method

  • urlServer - URL for WebSocket connection to WCS server
  • sipOptions - object with parameters for SIP connection
    var url = $('#urlServer').val();
    var registerRequired = $("#sipRegisterRequired").is(':checked');

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

    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

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session){
        setStatus("#regStatus", SESSION_STATUS.ESTABLISHED);
        onConnected(session);
		if (!registerRequired) {
            disableOutgoing(false);
		}
    }).on(SESSION_STATUS.REGISTERED, function(session){
        ...
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        ...
    }).on(SESSION_STATUS.FAILED, function(){
        ...
    }).on(SESSION_STATUS.INCOMING_CALL, function(call){ 
        ...
    });

4. Receiving the event confirming successful registration on SIP server

ConnectionStatusEvent REGISTERED code

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session){
        ...
    }).on(SESSION_STATUS.REGISTERED, function(session){
        setStatus("#regStatus", SESSION_STATUS.REGISTERED);
        onConnected(session);
        if (registerRequired) {
            disableOutgoing(false);
		}
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        ...
    }).on(SESSION_STATUS.FAILED, function(){
        ...
    }).on(SESSION_STATUS.INCOMING_CALL, function(call){ 
        ...
    });

5. Receiving the event on incoming call

ConnectionStatusEvent INCOMING_CALL code

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session){
        ...
    }).on(SESSION_STATUS.REGISTERED, function(session){
        ...
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        ...
    }).on(SESSION_STATUS.FAILED, function(){
        ...
    }).on(SESSION_STATUS.INCOMING_CALL, function(call){ 
        call.on(CALL_STATUS.RING, function(){
            ...
        });
		onIncomingCall(call);
    });

6. Outgoing call.

session.createCall(), call.call() code
The following parameters are passed when call is created

  • callee - callee SIP username
  • visibleName - display name
  • localVideoDisplay - <div> element, in which video from camera will be displayed
  • remoteVideoDisplay - <div> element, in which video from the other party will be displayed
    var outCall = session.createCall({
		callee: $("#callee").val(),
        visibleName: $("#sipLogin").val(),
	    localVideoDisplay: localVideo, 
        remoteVideoDisplay: remoteVideo,
        localVideoDisplay: localVideo,
        constraints: constraints,
        sdpHook: rewriteSdp,
        stripCodecs: "SILK"
        ...
    });
	
	outCall.call();

7. Answering incoming call.

call.answer() code
Object with answer options is passed to the method

  • localVideoDisplay - <div> element, in which video from camera will be displayed
  • remoteVideoDisplay - <div> element, in which video from the other party will be displayed
    $("#answerBtn").off('click').click(function(){
		$(this).prop('disabled', true);
		inCall.answer({
            localVideoDisplay: localVideo,
            remoteVideoDisplay: remoteVideo,
            constraints: constraints,
            sdpHook: rewriteSdp,
            stripCodecs: "SILK"
        });
		showAnswered();
    }).prop('disabled', false);

8. Outgoing call hangup.

call.hangup() code

	$("#callBtn").text("Hangup").off('click').click(function(){
        $(this).prop('disabled', true);
	    outCall.hangup();
    }).prop('disabled', false);

9. Incoming call hangup

call.hangup() code

    $("#hangupBtn").off('click').click(function(){
		$(this).prop('disabled', true);
		$("#answerBtn").prop('disabled', true);
        inCall.hangup();
    }).prop('disabled', false);

10. Call hangup on session disconnection

call.hangup() code

function onConnected(session) {
    $("#connectBtn").text("Disconnect").off('click').click(function(){
        $(this).prop('disabled', true);
		if (currentCall) {
			showOutgoing();
			disableOutgoing(true);
			setStatus("#callStatus", "");
			currentCall.hangup();
		}
        session.disconnect();
    }).prop('disabled', false);
}

11. Mute/unmute

currentCall.muteAudio(), currentCall.muteAudio(), currentCall.muteVideo(), currentCall.unmuteVideo() code

// Mute audio in the call
function mute() {
	if (currentCall) {
	    currentCall.muteAudio();
	}
}

// Unmute audio in the call
function unmute() {
	if (currentCall) {
        currentCall.unmuteAudio();
	}
}

// Mute video in the call
function muteVideo() {
	if (currentCall) {
        currentCall.muteVideo();
	}
}

// Unmute video in the call
function unmuteVideo() {
	if (currentCall) {
        currentCall.unmuteVideo();
    }
}

12. WebRTC statistics collection during the call

call.getStats() code

function loadStats() {
    if (currentCall) {
        // Stats shoukld be collected for active calls only #WCS-3260
        let status = currentCall.status();
        if (status != CALL_STATUS.ESTABLISHED && status != CALL_STATUS.HOLD) {
            return;
        }
        currentCall.getStats(function (stats) {
            if (stats && stats.outboundStream) {
                if (stats.outboundStream.video) {
                    $('#videoStatBytesSent').text(stats.outboundStream.video.bytesSent);
                    $('#videoStatPacketsSent').text(stats.outboundStream.video.packetsSent);
                } else {
                    $('#videoStatBytesSent').text(0);
                    $('#videoStatPacketsSent').text(0);
                }

                if (stats.outboundStream.audio) {
                    $('#audioStatBytesSent').text(stats.outboundStream.audio.bytesSent);
                    $('#audioStatPacketsSent').text(stats.outboundStream.audio.packetsSent);
                } else {
                    $('#audioStatBytesSent').text(0);
                    $('#audioStatPacketsSent').text(0);
                }
            }
        });
    }
}
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