A call to a mobile phone via the SIP server¶

Overview¶

A SIP call to a mobile phone is a special case of calls between a browser and a SIP device, when the SIP server either operates as a GSM/PSTN gateway itself or connects to one during the call.

Supported platforms and browsers¶

	Chrome	Firefox	Safari	Edge
Windows	✅	✅	❌	✅
Mac OS	✅	✅	✅	✅
Android	✅	✅	❌	✅
iOS	✅	✅	✅	✅

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

SIP functions are managed using the WebSDK.

Operation flowchart¶

The browser begins a call
WCS connects to the SIP server
The SIP server connects to the GSM/PSTN gateway
The GSM/PSTN gateway connects to the mobile phone
The browser and the phone exchange audio streams

Quick manual on testing¶

For the test we use:
a SIP account;
the Phone UI web application to make a call;
a mobile phone to answer the call.
Open the Phone UI web application. Click Log in and enter the SIP account credentials:
Enter the mobile phone number and click Voice call. Dialing starts:
The mobile phone displays an incoming call on the screen:
Answer the call on the mobile phone:
The browser also shows that the connection is established:
To terminate the call, click the Hangup button.

Call flow¶

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

Phone.html

Phone.js

Creating a call
Session.createCall(), Call.call() code

var outCall = this.session.createCall({
    callee: callee,
    visibleName: this.sipOptions.login,
    localVideoDisplay: this.localVideo,
    remoteVideoDisplay: this.remoteVideo,
    constraints: constraints
    ...
});

outCall.call();

Sending SIP INVITE to the SIP server
Sending SIP INVITE to the GSM/PSTN gateway
Establishing a connection to the mobile terminal
Receiving a confirmation from the GSM/PSTN gateway
Receiving a confirmation from the SIP server

Receiving from the server an event confirming successful connection
CALL_STATUS.ESTABLISHED code

var outCall = this.session.createCall({
    ...
}).on(CALL_STATUS.ESTABLISHED, function(call){
    me.callStatusListener(call);
    ...
});

Participants of the call exchange audio streams

Terminating the call
Call.hangup() code

Phone.prototype.hangup = function () {
    trace("Phone - hangup " + this.currentCall.id() + " status " + this.currentCall.status());
    this.hideFlashAccess();
    if (this.currentCall.status() == CALL_STATUS.PENDING) {
        this.callStatusListener(this.currentCall);
    } else {
        this.currentCall.hangup();
    }
    this.flashphonerListener.onHangup();
};

Sending SIP BYE to the SIP server
Sending SIP BYE to the GSM/PSTN gateway
Receiving a confirmation from the GSM/PSTN gateway
Receiving a confirmation from the SIP server