A SIP call between two browsers made via WCS is a special case of calls between a browser and a SIP device when the web application in a browser serves as a softphone for both parts of the call.

Overview

Supported platforms and browsers

Supported protocols

• WebRTC
• RTP
• SIP 

Supported codecs

• H.264
• VP8
• G.711
• Speex
• G.729
• Opus 

Supported SIP functions

• DTMF
• Holding a call
• Transferring a call

Management of SIP functions is performed using the REST API.

Operation flowchart

SIP server as a proxy server to transfer calls and RTP media

Image Added 

SIP server as a server to transfer calls only

 Image Added

1. The browser 1 begins a call from the Caller account to the Callee account 
2. WCS connects to the SIP server
3. The SIP server

...

1. transfers the call to the Callee to WCS
2. WCS sends to the browser 2 an event that a call is received
3. Browsers exchange audio and video streams

Without an external SIP server. SIP and RTP media are processed by WCS

 Image Added

1. The browser 1 begins a call from the Caller account to the Callee account
2. WCS establishes a SIP connection between accounts
3. WCS sends to the browser 2 an event that a call is received
4. Browsers exchange audio and video streams

Quick manual on testing

1. For the test we use:

• two SIP accounts;
• the Phone web application to make a call

2. Open the Phone web application. Enter the data of the SIP account and click the Connect button to establish a connection with the server:

Image Added

3. Open the Phone web application in a new browser tab. Enter the data of the second SIP account and click the Connect button:

Image Added

4. Enter the identifier of the SIP account receiving the call and click the Call button:

Image Added

5. Answer the call by clicking the Answer button:

Image Added

The call starts:

Image Added

6. To terminate the call, click the "Hangup" button.

Call flow

Below is the call flow when using the Phone example to create a call. The SIP server is used as a proxy server to transfer commands and media.

phone.html

phone.js

Image Added

1. Sending the /call/startup REST query using JavaScript API:

session.createCall(), call.call() code

    var outCall = session.createCall({
		callee: $("#callee").val(),
        visibleName: $("#sipLogin").val(),
		localVideoDisplay: localDisplay,
		remoteVideoDisplay: remoteDisplay,
		constraints: constraints,
		receiveAudio: true,
        receiveVideo: false
        ...
    });
	
	outCall.call();

2. Establishing a connection to the SIP server

3. The SIP server establishes a connection to WCS

4. Sending to the second browser an event notifying about the incoming call

CallStatusEvent RING code

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){
        ...
    }).on(SESSION_STATUS.INCOMING_CALL, function(call){ 
        call.on(CALL_STATUS.RING, function(){
		    setStatus("#callStatus", CALL_STATUS.RING);
            ...
        });

5. The second browser answers the call

call.answer() code

function onIncomingCall(inCall) {
	currentCall = inCall;
	
	showIncoming(inCall.caller());
	
    $("#answerBtn").off('click').click(function(){
		$(this).prop('disabled', true);
        var constraints = {
            audio: true,
            video: false
        };
		inCall.answer({
            localVideoDisplay: localDisplay,
            remoteVideoDisplay: remoteDisplay,
            receiveVideo: false,
            constraints: constraints
        });
		showAnswered();
    }).prop('disabled', false);
    ...
}

6. Sending a confirmation to the SIP server

7. Receiving a confirmation from the SIP server

8. The first browser receives from the server an event confirming successful connection.

CallStatusEvent ESTABLISHED code

    var outCall = session.createCall({
        ...
    }).on(CALL_STATUS.ESTABLISHED, function(){
		setStatus("#callStatus", CALL_STATUS.ESTABLISHED);
        $("#holdBtn").prop('disabled',false);
        onAnswerOutgoing();
        ...
    });
	
	outCall.call();

9. The caller and the callee exchange audio and video streams

10. Terminating the call

call.hangup() code

function onConnected(session) {
    $("#connectBtn, #connectTokenBtn").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. Sending the command to the SIP server

12. Receiving the command from the SIP server

13. Sending to the second browser an event confirming termination of the call

CallStatusEvent FINISH code

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){
        ...
    }).on(SESSION_STATUS.INCOMING_CALL, function(call){ 
        call.on(CALL_STATUS.RING, function(){
		    ...
        }).on(CALL_STATUS.FINISH, function(){
		    setStatus("#callStatus", CALL_STATUS.FINISH);
			onHangupIncoming();
		    currentCall = null;
            ...
        });

14. Sending a confirmation to the SIP server

15. Receiving a confirmation from the SIP server

SIP calls without an external SIP server

WCS may establish a SIP call and process its traffic withoun an external SIP server (see the scheme above). To do this, the following parameters must be set in flashphoner.properties file

enable_local_videochat=true
sip_add_contact_id=false

1. For test we use:

• Phone web application to make a call

2. Open Phone web application page. Enter the following:

• user name
• password
• SIP Domain: set WCS server IP address (not domain name!)
• SIP Outbound Proxy: set WCS server IP address (not domain name!)
• set SIP Port to 0
• clean Register requred checkbox

Click Connect

Image Added

3. Open Phone web application page in another browser window. Enter the following:

• second user name
• password
• SIP Domain: set WCS server IP address (not domain name!)
• SIP Outbound Proxy: set WCS server IP address (not domain name!)
• set SIP Port to 0
• clean Register requred checkbox

Click Connect

Image Added

4. Enter callee name and click Call

Image Added

5. Accept the call by clicking Answer

Image Added

6. The call is established

Image Added