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    webrtc-sdk

    2.2.7 • Public • Published

    WebRTC V2 Simple Calling API + Mobile

    Known Vulnerabilities npm

    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 public Google 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.

    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://localhost:4443/tutorials/

    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);
        });
    });

    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

    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     */ });

    Set Camera Resolution

    You can change the resolution of your camera's media capture. This allows you to set lower resolutions for slower p2p connections. You can also set HD 4K resolutions if you have the camera to do so.

    // Phone
    const phone = PHONE({
        number        : number
    ,   media         : { video: { width:1280, height:720 } } // <---- set res
    ,   publish_key   : pubkey
    ,   subscribe_key : subkey 
    });

    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
        media         : { audio : true, video : true },
        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.camera.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.camera.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 Session Call Rejection and Accept Permissions

    phone.send

    phone.send allows you to send programmatic messages between two phones without a video/audio stream. You may wish to setup a Call Accept/Reject phase to allow to users to accept or reject calls.

    Before the Sending the Video/Audio Stream, send a signal to ask for call permission:

    let user_number = "1235445"; // my friends number
     
    function call_request(number) {
        phone.send( { "accept" : "Would you like to accept this call?" }, user_number );
    }
    function call_accepted() {
        // start voice/video session
        phone.dial(user_number);
    }
    function call_rejected() {
        // show call rejected screen
    }

    This allows you to create a simple contract between two parties before the video and audio stream begins.

    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).

    Install

    npm i webrtc-sdk

    DownloadsWeekly Downloads

    20

    Version

    2.2.7

    License

    MIT

    Unpacked Size

    1.3 MB

    Total Files

    32

    Last publish

    Collaborators

    • stephenlb