WebRTC V2 Simple Calling API + Mobile

WebRTC SDK Upgraded! ES6, new camera control and 100x less code than v1.

The following demo uses PubNub for signaling to transfer the metadata and establish the peer-to-peer connection. Once the connection is established, the video and voice runs on Xirsys version 3 STUN/TURN servers. Keep in mind, PubNub can provide the signaling for WebRTC, and requires you to combine it with a hosted WebRTC solution. For more detail on what PubNub does, and what PubNub doesn’t do with WebRTC, check out this article: https://support.pubnub.com/support/solutions/articles/14000043715-does-pubnub-provide-webrtc-and-video-chat-

At PubNub we believe simplicity is essential for our SDK usability. We've taken a simplified approach to WebRTC Peer Connections by creating and easy-to-use SDK for developers. The ideas of simplicity should span all platforms and devices too and that's why we also support Android WebRTC mobile calling with compatibility for iOS native Objective-C based WebRTC SDK. This simple developer WebRTC SDK is powered by PubNub Data Stream Network & Xirsys WebRTC Infrastructure.

Supported WebRTC Features

WebRTC SDK offers many rich features and capabilities to enhance the WebRTC experience. Here is a list of the options available.

1. Photo Snap Camera Transmit (STUN-less Firewall Ready)
2. WebRTC Dialing (STUN-less Firewall Ready)
3. WebRTC Call Receiving (STUN-less Firewall Ready)
4. JSON App Messaging (chat/signals/etc.) (STUN-less Firewall Ready)
5. Multi-party Calling
6. Audio only Calls Optional
7. Broadcasting Mode
8. Instant Connect Dialing Optional
9. Security SSL, AES256, ACL Access Control Management
10. Password Protection via Cipher
11. Event History and Call Records
12. Photo History and Recording Static Snapshots of Calls (only stills)
13. Connectivity Detection and Auto-Recovery
14. Pre-configured Video Element for Streaming Video/Audio
15. Pre-configured Local Camera Video Element for Streaming Video/Audio
16. Network Connectivity Hooks (online/offline)
17. SDK Level Debug Output

Testing Locally

You need an HTTPS (TLS) File Server. To start a local secure file server:

python <(curl -L https://goo.gl/LiW3lp)

Then open your browser and point it to your file in the directory you ran the python HTTPS server.

open https://0.0.0.0:4443/your-file-here.html

This is a Simple Python HTTPS Secure Server https://gist.github.com/stephenlb/2e19d98039469b9d0134

We posted an answer on StackOverflow WebRTC HTTPS. This will get you started testing on your laptop.

Supported WebRTC Calling API Mobile Devices and Browser

List of supported WebRTC Calling Clients including Android and Chrome.

1. Chrome
2. Firefox
3. Opera
4. Mobile Chrome - Android
5. Mobile Firefox - Android
6. iOS Native Objective-C Compatible
7. Android Native Java Compatible

The Basic Concepts of WebRTC Calling

Making a WebRTC phone Call

// Dial Number
var session = phone.dial('123-456');

Receiving a WebRTC phone Call

phone.receive(function(session){
    // On Call Receiving
});

Adding Video Live Stream

phone.receive(function(session){
    session.connected(function(session){
        // Append Live Video Feed
        $('#display-div').append(session.video);
    });
});

Adding Xirsys Account Information

function dial(number) {
    $.ajax({      
        type: 'PUT',
        //Replace "yourChannel" with your actual Xirsys channel ID
        url: 'https://ws.xirsys.com/_turn/yourChannel/',      
        headers: {
            //Replace "ident" and "secret" with your actual Xirsys ident and secret key
        'Authorization': 'Basic ' + btoa('ident:secret')
     },        
        success: function(res) {
            // Dial Number
            phone.dial(number, res.v.iceServers);
            // Show Connecting Status
            set_icon('send');
        },        
    });       
}

Simple WebRTC Walkthrough

Next we will start with a copy/paste example of our SDK. This Simple Example Comes in Two WebRTC Calling Sections.

1. Part One will talk about how you can Make a WebRTC Call.
2. Part Two will teach you about Receiving a WebRTC Call.

Making a WebRTC Calling & Receiving - Part One and Two

Make your first html file named dial.html and copy/paste the following:

<!-- dial.html -->

<!-- Video Output Zone -->
<div id="video-out"> Making a Call </div>

<!-- Libs and Scripts -->
<script src="https://stephenlb.github.io/webrtc-sdk/js/webrtc-v2.js"></script>
<script>(()=>{
    // ~Warning~ You must get your own API Keys for non-demo purposes.
    // ~Warning~ Get your PubNub API Keys: https://www.pubnub.com/get-started/
    // The phone *number* can by any string value
    var phone = PHONE({
        number        : '1234',
        publish_key   : 'pub-c-561a7378-fa06-4c50-a331-5c0056d0163c',
        subscribe_key : 'sub-c-17b7db8a-3915-11e4-9868-02ee2ddab7fe',
        ssl           : true
    });

    // As soon as the phone is ready we can make calls
    phone.ready(function(){

        // Dial a Number and get the Call Session
        // For simplicity the phone number is the same for both caller/receiver.
        // you should use different phone numbers for each user.
        var session = phone.dial('1234');

    });

    // When Call Comes In or is to be Connected
    phone.receive(function(session){

        // Display Your Friend's Live Video
        session.connected(function(session){
            phone.$('video-out').appendChild(session.video);
        });

    });

})();</script>

Live WebRTC Call Dialer

If you combine both the WebRTC Dialer and the WebRTC Receiver you will get a complete working phone. We have a live running WebRTC Demo which uses our WebRTC SDK. This demo includes a soft-touch UI for an easy calling experience.

try the live WebRTC Dialing: WebRTC Simple Calling API + Mobile

You can click the link above to try our live WebRTC Demo which is powered by our easy to use SDK.

What Can you build with a WebRTC Simple Calling API?

There are a plethera of important and useful applications which may be built using the PubNub WebRTC Calling SDK. I have cataloged a list of ideas for WebRTC Use Cases:

1. Amazon Mayday Help Button
2. Salesforce SOS Help Button
3. WebRTC Skype Replica
4. Traditional Phone Replacement
5. Chatroulette
6. VoIP Replacement
7. Customer Support Calls
8. In-bound Sales Calls
9. Easy Remote Meetings
10. Call Assistant or Specialists
11. Big Screen Public Announcemnt
12. Live Presentations

So many different options and even more use cases that have yet to be discovered.

What is a WebRTC Session?

A WebRTC Session is a reference to a call instance between two connected parties. The idea is the session is active as long as the two parties are connected. Once one party member disconnects or leaves, the session will be terminated. You have access to several helper methods for session.connected() and session.ended() events.

API Documentation for WebRTC Calling SDK

The WebRTC Simple SDK API Reference will provide to you all the options available to you as the developer.

WebRTC Phone Initialization

PHONE({ ... })

Initialize the phone object which can send/receive multiple WebRTC call sessions without limit. Be careful as your bandwidth is the true limiter.

var phone = PHONE({
    number        : '1234567890',
    publish_key   : 'pub-c-561a7378-fa06-4c50-a331-5c0056d0163c',
    subscribe_key : 'sub-c-17b7db8a-3915-11e4-9868-02ee2ddab7fe',
    media         : { audio : true, video : true },
    ssl           : true
})

WebRTC Phone Number

Your phone number is your true ring-in point of truth. You can set your phone number at init time from the

var phone = PHONE({ number : '1234567890' });

WebRTC Local Camera Video Element

We provide you easy access to your local camera with a pre-initialized video element that is easy to access. Simply append the element to your DOM and the feed will stream automatically.

$('#display-div').append(phone.video);

WebRTC Phone SSL Mode

You can enable SSL signalling mode for the local client device by setting the ssl : true parameter at init.

var phone = PHONE({
    ssl : true
    ...
})

WebRTC Cipher AES 256 Crypto Mode

You can enable AES256 Encryption (essentially password mode) on your phone for additional security. You're friends have to know your password to call you. AES256 option allows you to password protect your phone and only give access to people you know.
You have to give your friend your password before they can call you.

var phone = PHONE({
    cipher_key : 'SUPER-SECRET-PASSWORD-HERE'
    ...
})

WebRTC Phone Audio Only Mode

You can disable video codec and stream only Audio by setting the following param in your init code. Set video : false in the media section.

var phone = PHONE({
    media : { audio : true, video : false }
    ...
})

WebRTC Phone Mobile Calling on Android

WebRTC calling on Android is web-ready compatible and works out-of-the-box today without modifications or additional code. Also WebRTC Calling is supported for Android and iOS Native too.

WebRTC Photo Sharing Bonus STUN-less Ready

You can easily snap a photo from the video stream and send it to your friends in an instant. You can think of this as an Instagram WebRTC style. Also Photo Sharing works through Corprate Enterprise Firewalls.

WebRTC Camera Photo Sharing Broadcast

phone.snap()

Broadcast your camera photo to all connected sessions. Also get the IMG data as base64 supported format for local display if desired.

phone.ready(function(){
    // Auto Send Camera's Photo to all connected Sessions.
    var photo = phone.snap();
    $('#photo-div').append(photo.image);
});

WebRTC Session Camera Photo Share

session.snap()

Send your camera's latest frame as raw IMG to a specific call session.

phone.ready(function(){
    var session = phone.dial('4321');
    var photo   = session.snap();
    $('#photo-div').append(photo.image);
});

Prevent Camera from Starting Automatically

By default the WebRTC SDK starts user's camera. You can optionally prevent this by setting the autocam flag to false. Here is an example of disabling the camera on initialization.

<!-- dial.html -->
<div id="number"></div>

<button id="startcam">Start Camera</button>
<button id="startcall">Start Call</button><input id="dial">

<!-- Video Output Zone -->
<div id="video-out"></div>

<!-- Libs and Scripts -->
<script src="https://stephenlb.github.io/webrtc-sdk/js/webrtc-v2.js"></script>
<script>(=>(){

    // ~Warning~ You must get your own API Keys for non-demo purposes.
    // ~Warning~ Get your PubNub API Keys: https://www.pubnub.com/get-started/
    // The phone *number* can by any string value
    var number  = Math.ceil(Math.random()*10000);
    var ready   = false;
    var session = null;
    var phone   = PHONE({
        number        : number
    ,   autocam       : false
    ,   publish_key   : 'pub-c-561a7378-fa06-4c50-a331-5c0056d0163c'
    ,   subscribe_key : 'sub-c-17b7db8a-3915-11e4-9868-02ee2ddab7fe'
    ,   ssl           : true
    });

    // Show Number
    phone('number').innerHTML = 'Number: ' + number;

    // Start Camera
    phone.bind( 'mousedown,touchstart', phone.$('startcam'), function() {
        phone.startcamera();
        return false;
    } );

    // Start Call
    phone.bind( 'mousedown,touchstart', phone.$('startcall'), function() {
        console.log('calling');
        session = phone.dial(phone.$('dial').value);
        return false;
    } );

    // As soon as the phone is ready we can make calls
    phone.ready(function(){

        // Dial a Number and get the Call Session
        // For simplicity the phone number is the same for both caller/receiver.
        // you should use different phone numbers for each user.
        ready = true;

    });

    // When Call Comes In or is to be Connected
    phone.receive(function(session){

        // Display Your Friend's Live Video
        session.connected(function(session){
            phone.$('video-out').appendChild(session.video);
        });

    });

})();</script>

WebRTC JSON Messaging Bonus STUN-less Ready

Adding extra realtime capabilities between connected parties is essential for creating collaborative and chat features. You can do that with PubNub's WebRTC SDK easily without firewall restrictions from corporate policies.

Message Broadcasting to All Sessions

phone.send(...)

Send a JSON message to all connected sessions.

phone.send({ text : 'HI!' });

Receive a JSON message from Any Session

phone.message(function(message){ ... })

Get all messages sent from any session.

phone.message(function( session, message ) {
    show_chat( session.number, message.text );
} );

Send a JSON Message to One Session

session.send(...)

You can send a direct JSON message to one session only.

session.send({ text : 'Hi there!' });

Receive a JSON message from One Session

session.message(function(){ ... })

You can set callbacks to capture JSON messages from a specific call session.

session.message(function( session, message ) {
    show_chat( session.number, message.text );
} );

WebRTC Phone Ready

phone.ready(function(){ ... })

Making calls is easy but you can only do it when the phone is ready to issue the signals properly and the local interfaces have been configured such as audio/video media.

phone.ready(function(){
    // Ready to make Calls
    var session = phone.dial("my-friend's-number");
});

WebRTC Phone Receiving Calls

phone.receive(function(session){ ... })

It's really ease to setup your phone to receive calls using the phone.receive() method. This method expects a callback function and will pass the WebRTC Session object as the only parameter.

phone.receive(function(session){
    session.connected(function(session){ /* call connected */ });
    session.ended(function(session){     /* call ended     */ });
});

Get Your Phone Number

var num = phone.number()

Sometimes you need to access the phone number that was set during initialization time. You can do that by calling phone.number() method which returns the setup number.

var num = phone.number();

Get Caller Phone Number

var num = session.number

To access current caller number, check the session object number property session.number.

var num = session.number;

Get Call Start Time

var start = session.started

The Session object stores the call start time which you can use to display call timer on the screen.

var start = session.started;

WebRTC Phone Call History via PubNub

phone.history(...)

You can get the call history of a phone number by issuing a PubNub History call on the phne number.

phone.history({
    number  : '1234',
    history : function(call_history) {
        console.log(call_history);
    }
});

WebRTC Phone Dialing Numbers

phone.dial(number)

You can easily make WebRTC calls by executing the dial() method. The number can be any string value such as "415-555-5555".

var session = phone.dial(number);

session.connected(function(session){ /* call connected */ });
session.ended(function(session){     /* call ended     */ });

WebRTC Phone Multi-party Dialing

phone.dial(number)

The PubNub WebRTC Phone Dialer and Receiver supports unlimited party in/out calling.

var sessions = [];

sessions.push(phone.dial('friend-one'));
sessions.push(phone.dial('friend-two'));
sessions.push(phone.dial('friend-three'));
sessions.push(phone.dial('friend-four'));
sessions.push(phone.dial('friend-five'));

sessions.forEach(function(friend){
    friend.connected(function(session){ /* call connected */ });
    friend.ended(function(session){     /* call ended     */ });
});

WebRTC Video and Audio Broadcasting Mode

phone.receive(function(session){ ... })

You can receive unlimited inbound calls and become a broadcast beacon stream. You are limited by your bandwidth upload capacity.

Broadcaster with Audience Members

You'll start by opening the stream for the broadcaster so audience members can join in. Start broadcasting as the broadcaster:

Broadcaster

// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
// Initialize the Broadcaster's Device
// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
var broadcaster = PHONE({
    number        : "BROADCASTER",  // If you want more than one broadcaster, use unique ids
    publish_key   : 'pub-c-561a7378-fa06-4c50-a331-5c0056d0163c',
    subscribe_key : 'sub-c-17b7db8a-3915-11e4-9868-02ee2ddab7fe',
    ssl           : true
});

// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
// Wait for New Viewers to Join
// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
broadcaster.receive(function(new_viewer){
    new_viewer.connected(function(viewer){ /* ... */ }); // new viewer joined
    new_viewer.ended(function(viewer){ /* ... */ });  // viewer left
    //new_viewer.hangup();  // if you want to block the viewer
});

Viewer

// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
// Initialize the Viewer's Device
// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
var viewer = PHONE({
    number        : "VIEWER-"+new Date,
    publish_key   : 'pub-c-561a7378-fa06-4c50-a331-5c0056d0163c',
    subscribe_key : 'sub-c-17b7db8a-3915-11e4-9868-02ee2ddab7fe',
    ssl           : true
});

// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
// Join a Broadcast as a Viewer
// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
viewer.ready(function(){
    var broadcaster = phone.dial("BROADCASTER");
});

// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
// Show Broadcast's Video Stream
// =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
viewer.receive(function(broadcaster){
    broadcaster.connected(function(broadcaster){
        document.body.appendChild(broadcaster.video);
    });
    broadcaster.ended(function(broadcaster){ /* broadcast ended */ });
});

WebRTC Phone Hangup

phone.hangup()

There are two ways to hangup WebRTC calls. You can use the phone-level method phone.hangup() which will hangup all calls at once. Or you can use the session-level method session.hangup() which will only hangup that call session.

// hangup all calls
phone.hangup();

// hangup single session
session.hangup();

WebRTC Phone Network Events

PHONE.disconnect(function(){ ... })

You need to keep track of the connectivity state of your local network connection. You can do that using these helper methods.

PHONE.connect(function(){    console.log('network LIVE.') })
PHONE.disconnect(function(){ console.log('network GONE.') })
PHONE.reconnect(function(){  console.log('network BACK!') })

WebRTC Phone Unable to Initialize

phone.unable(function(details){ ... })

Some devices or in certain situations the phone may not initialize. We give you a simple callback for when the phone startup fails.

phone.unable(function(details){
    console.log("Phone is unable to initialize.");
    console.log("Try reloading, or give up.");
});

WebRTC Stop Camera and Mic

phone.stop()

You may want to Stop the Camera/Mic recording. By default the camera and mic are turned on as soon as possible. This allows for faster calling connection speeds.

    var streamref = phone.stop();

WebRTC Phone Debugging

phone.debug(function(details){ ... })

You might want to see under the covers of WebRTC Calling by enabling debugging mode on the WebRTC SDK.

phone.debug(function(details){
     console.log(details);
});

WebRTC Phone Auto Hangup and Goodbye on Unload

The WebRTC Calling SDK will attempt an automatic goodbye upon graceful disconnection attempts. This allows the 2nd party on the other end of the phone line to receive a call ended signal. This happens automatically.

The WebRTC Session Object

A WebRTC Session represents the connection between two parties with access to the session.video element as well as the place to register event callbacks as needed such as session.connected and also the ended callback for when the call disconnects or hangs up.

session.ended(function(session){})

A session object is generated automatically for you upon dialing

var session = phone.dial('...')

and also inside registered event callbacks.

phone.receive(function(session){})

WebRTC Session Number

session.number

Returns the 2nd party's (caller/callee) Phone Number associated with the Call Session.

var session = phone.dial('12345');
console.log(session.number == '12345');

WebRTC Session Connected Callback

session.connected(function(session){})

Sets the callback for when the session is connected and the video stream is ready to display.

session.connected(function(session){
    var body = phone.$$('body')[0];
    body.appendChild(session.video);
});

WebRTC Session Ended Callback

session.ended(function(session){})

The session has ended by one of the parties. Any secondary session will continue to stream.

session.ended(function(session){
    sounds.play('sound/goodbye');
    console.log("Bye!");
});

WebRTC Session Hangup

session.hangup()

End the session right now. The ended callback will fire for both connected parties.

$("#hangup").click(function(){
    // End the call
    session.hangup();
});

WebRTC Session Video Element

session.video

The Session Video Element is Accessable and Ready inside the connected only. The Session Video ref is an HTML Video Element <video>.

session.connected(function(session){
    var body  = phone.$$('body')[0];
    var video = session.video;

    body.appendChild(video);
});

WebRTC Session Image Element

session.image

The Session Image Element is Accessable and Ready inside the thumbnail, connected and ended callbacks. The Session Image ref is an HTML Image Element <img>.

session.thumbnail(function(session){
    var body  = phone.$$('body')[0];
    var image = session.image;

    body.appendChild(image);
});

WebRTC Session RTCPeerConnection Reference

session.pc

Reference to WebRTC RTCPeerConnection.

var sesionPeerConnection = session.pc;

WebRTC Adding Custom STUN and TURN Servers

You may desire to add your own custom stun or turn servers by using the servers parameter on initialization. For example http://xirsys.com/ offers paid-stun solution.

var phone = PHONE({
    servers : [
        {"username":"free","url":"turn:127.0.0.1?transport=udp","credential":"free"},
        {"username":"free","url":"turn:127.0.0.1?transport=tcp","credential":"free"}
    ]
    // ...
})

SDK Possible Upgrade Future Patches

- Race - During Ring/Receive Handshake, a Hangup will create Race
- Wire-pulled Disconnect Detect via DataChannels Pings
- 5 Second Timeout to Retry with 30 Second of Retries
- Auto-reconnect re-SDP/ICE Recovery
- Custom Message Events
- Presence
- Call History
- User Lists

Implementation Reference Upgrades

- Pre-Allow Transmit - Before "allow" fire a PubNub message
- Chat on Screen
- Multi-Party Video in GUI
- Full Screen Mode
- Controlling an iFrame

What is Happens Inside the Simple WebRTC SDK

Signaling and Exchanging ICE Candidates via PubNub

The goal is to exchange ICE candidate packets between two peers. ICE candidate packets are structured payloads which contain possible path recommendations between two peers. You can use a lib which will take care of the nitty gritty such as WebRTC Simple Calling API + Mobile however below is the general direction that is taken inside the SDK itself.

Note that the demonstration code below is intintionally incomplete. Note however the PubNub WebRTC Signaling SDK properly covers most Calling Situations.

Signaling Example Code Follows

<script src="https://cdn.pubnub.com/pubnub-3.6.3.min.js"></script>
<script>(function(){
    
    // INIT P2P Packet Exchanger
    var pubnub = PUBNUB({
        publish_key   : 'demo',
        subscribe_key : 'demo',
        ssl           : true
    })
    
    // You need to specify the exchange channel for the peers to
    // exchange ICE Candidates.
    var exchange_channel = "p2p-exchange";
    
    // LISTEN FOR SDP and ICE CANDIDATES
    pubnub.subscribe({
        channel : exchange_channel,
        message : receive_ice_candidates
    })
    
    // SDP and ICE CANDIDATES RECEIVER PROCESSOR FUNCTION
    function receive_ice_candidates(ice_candidate) {
        // Attempt peer connection or upgrade route if better route...
        console.log(ice_candidate);
        // ... RTC Peer Connection upgrade/attempt ...
    }
    
    // SEND SDP and ICE CANDIDATES
    function send_ice_candidate(ice) {
        pubnub.publish({
            channel : exchange_channel,
            message : ice
        })
    }

})();</script>

Generate ICE Candidates Example Code Follows:

<script>(function(){
    // CREATE ICE CANDIDATES
    var pc = new RTCPeerConnection();
    navigator.getUserMedia( {video: true}, function(stream) {
        pc.onaddstream({stream:stream});
        pc.addStream(stream);
        pc.createOffer( function(offer) {
            pc.setLocalDescription(
                new RTCSessionDescription(offer),
                send_ice_candidate,
                error
            );
        }, error );
    } );

})();</script>

WebRTC Troubleshooting

You may need to force clear your cache on your device, close the app completley, then restart it. This is uncommon. You can also enable debugging at the code-level by hooking onto the phone.unable(fn) and phone.debug(fn).