Web Call Server supports audio and video calls from a browser to SIP devices, PBX servers, SIP-GSM gates, VoIP conferences and other devices supporting the SIP protocol. Therefore, a web application can work in a browser as a software phone with the support for the SIP protocol, receive and initiate voice and video calls.

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 with the REST API.

Operation flowchart

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

2: SIP server as a server to transfer calls only

1. The browsers initiates a call using the /call/startup REST query.
2. WCS connects to the SIP server.
3. The SIP server connects to the SIP device receiving the call.
4. The browser and the SIP device exchange audio and video streams.

Call flow

Below is the call flow when using the Phone example to create a call.

phone.html

phone.js

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. Establishing a connection to the callee

4. Receiving a confirmation from the SIP device

5. Receiving a confirmation from the SIP server

6. Receiving from the server an event confirming successful connection.

CallStatusEvent ESTABLISHED code

    var outCall = session.createCall({
		callee: $("#callee").val(),
        visibleName: $("#sipLogin").val(),
		localVideoDisplay: localDisplay,
		remoteVideoDisplay: remoteDisplay,
		constraints: constraints,
		receiveAudio: true,
        receiveVideo: false
	}).on(CALL_STATUS.RING, function(){
        ...
    }).on(CALL_STATUS.ESTABLISHED, function(){
		setStatus("#callStatus", CALL_STATUS.ESTABLISHED);
        $("#holdBtn").prop('disabled',false);
        onAnswerOutgoing();
    }).on(CALL_STATUS.HOLD, function() {
        ...
    }).on(CALL_STATUS.FINISH, function(){
        ...
    }).on(CALL_STATUS.FAILED, function(){
        ...
    });
	
	outCall.call();

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

8. 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);
}

9. Sending the command to the SIP server

10. Sending the command to the SIP device

11. Receiving a confirmation from the SIP device

12. Receiving a confirmation from the SIP server

Making an outgoing call from a browser to a SIP device

1. For the test we use:

• two SIP accounts;
• the Phone Video web application to make a call;
• a software phone to answer the call.

2. Open the Phone Video web application. Enter the data of the SIP account making the call from a browser:

3. Run the software phone, enter the data of the SIP account receiving the call:

4. Click the Connect button in the browser. Then enter the identifier of the SIP account that receives the call and click the Call button:

5. Answer the call in the softphone by clicking the answer a video call button:

In a separate video, the video broadcast from the browser is shown:

6. The browser also displays the video:

7. To terminate the call, click the Hangup button in the browser or in the softphone.

Receiving an incoming call from a SIP device in a browser

1. For the test we use:

• two SIP accounts;
• a software phone to make the call.
• the Phone Video web application to answer the call.

2. Open the Phone Video web application. Enter the data of the SIP account receiving the call in a browser:

3. Run the software phone, enter the data of the SIP account making the call:

4. Click the Connect button in the browser, a connection to the server is established. In the softphone enter the identifier of the SIP account that receives the call and click the Call button:

5. Answer the call in the browser by clicking the Answer button:

6. The browser displays the video:

7. The video broadcast from a browser also displays in a separate window of the softphone:

8. To terminate the call, click the Hangup button in the browser or the end call button in the softphone.

Known issues

1. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect

Symptoms: SIP call stucks in PENDING state.

Solution: according to the standard, 'SIP Login' and 'SIP Authentification name' should not contain any of unescaped spaces and special symbols and should not be enclosed in angle brackets '<>'.

For example, this is not allowed by the standard

sipLogin='Ralf C12441@host.com'
sipAuthenticationName='Ralf C'
sipPassword='demo'
sipVisibleName='null'

and this is allowed

sipLogin='Ralf_C12441'
sipAuthenticationName='Ralf_C'
sipPassword='demo'
sipVisibleName='Ralf C'