webrtcdevelopment

webrtc

Web realtime communication SDK

This is a ready to deploy webrtc SDK and SaaS for a customized and flexible communication and collaboration solution .

Architecture

---

The Solution primarily contains nodejs frameworks for hosting the project and webbsockets over socket.io to perform offer - answer handshake and share SDP (Session description protocol ).

Technologies used

1. WebRTC Web based real time communication framework. read more on webrtc

2. Node ( v10.0.0) Asynchronous event driven JavaScript runtime

3. socket.io ( v0.9) Communication and signalling

Note : while its possible to use any prtotocol like SIP , XMPP , AJAX , JSON etc for this purpose , modifying thsi libabry will require a lot of rework . It would be advisble to start from apprtc directly in that case .

1. Grunt It is a task Runner and its used to automate running of command in gruntfile

grunt -verbose

SDK

---

Project is divided into 4 parts

1. Core RTC Conn Lib
2. Wrappers for the Core Lib containing feature sets and widgets like screensharing , recording , pointer share , machine learning , face detection etc
3. Demo Applicatins like two party full-features , multi-party full features etc which implement and use the SDK by invoking the constructirs , emitters and listeners .
4. SIgnaller over socket.io for SDP excahnge on offer answer model

Building the SDK

Download the dev dependencies by setting the NODE_ENV to dev . This will install all grunt and gulp dependencies used for building the SDK

NODE_ENV=development npm install

To build the RtcConn , outputs RTCMultiConn

grunt rtcconn

To build the webrtcdev lib . It encapsulates the rtcconn core along with external libs for building various custom features . Outputs webrtcdevelopment.js , webrtcdevelopment_header.js , webrtcdevelopment.css , webrtcdevelopment_header.css and webrtcdevelopmentserver.js

gulp production

Steps

---

To run this project following steps need to be followed in that order :

1. Get the project from github

git clone https://github.com/altanai/webrtc.git webrtc

2. install nvm ( node version manager )

curl -o- https://raw.githubusercontent.com/creationix/nvm/v0.31.2/install.sh | bash
. ~/.nvm/nvm.sh
nvm install v12.0.0
nvm use v12.0.0

3. install npm and update the dependencies It will read the package.json and update the dependencies in node_modules folder on project location

sudo apt-get install npm
npm install 

4. Change ENV variables and Test

To change the ports for running the https server and rest server, goto env.json

{       
    "hostname"      : "host",        
	"enviornment"   : "local",        
	"host"        	: "localhost",
	"jsdebug"      :  true,          
	"httpsPort"    :  8086,
	"restPort"     :  8087
}

To run the tests

npm test

5. Start up the Server

To start the Server in dev mode and stop the server as soon as Ctrl+ C is hit or the terminal widnow is closed .

node webrtcserver.js

read more about node

To start the Server using npm start ( using package.json) , behaves same as earlier run using node. We use supervisor to restart the server incase of exceptions or new code .

npm start

6. JS and CSS Libs

Make a webpage and give holders for video and button elements that SDK will use .

Inside the head tag of html build/webrtcdevelopment_header.css build/webrtcdevelopment_header.js

After the body tag of html build/webrtcdevelopment.css build/webrtcdevelopment.js

or use the minified scripts build/webrtcdevelopment_min.css build/webrtcdevelopment_min.js

7. Configure

Create the webrtc dom object with local and remote objects

local object :

    var local={

        video           :   "myAloneVideo",            // name of the local video element
        videoClass      :   "",                        // class of the localvideo
        videoContainer  :   "singleVideoContainer",    // outer container of the video element
        userDisplay :       false,                     // do you want to display user details
        userMetaDisplay :   false,
        userdetails:{                                   // users details include name , email , color
            username    : username,
            usercolor   : "#DDECEF",
            useremail   : useremail,
            role        : "participant"                 // role of user in the session , can be participant , admin , inspector
        }
    }

remote object :

    var remote={
        videoarr        : ["myConferenceVideo", "otherConferenceVideo"], // conatiners for the video after session is made 
                                                                // first one is usually the local video holder followed by remote video holders
        videoClass      : "",
        maxAllowed      : "6",
        videoContainer  : "confVideoContainer",
        userDisplay     : false,
        userMetaDisplay : false,
        dynamicVideos   : false 
    }

Incoming and outgoing media configuration ( self explanatory ) :

    var incoming={
        audio :  true,
        video :  true,
        data  :  true,
        screen:  true
    };

    var outgoing={
        audio :  true,
        video :  true,
        data  :  true,
        screen:  true

    };

    webrtcdomobj= new WebRTCdom(
        local, remote, incoming, outgoing
    );

7. Adding Widgets

set widgets (explained in section below)

    var widgets={     }

Set widgets and their properties

8. Creating session

Get session id automatically

sessionid = webrtcdevobj.makesessionid("reload");

or get session name from user

sessionid = webrtcdevobj.makesessionid("noreload");

9. Create a session json object with turn credentials and the session created from above step

set preference for the incoming and outgoing media connection. By default all are set to true .

    var incoming={
        audio:  true,
        video:  true,
        data:   true,
        screen: true
    };

    var outgoing={
        audio:  true,
        video:  true,
        data:   true,
        screen: true
    };

10. finally initiate the webrtcdev constructor

webrtcdevsessionobj = webrtcdevobj.setsession(local, remote, incoming, outgoing, session, getWidgets());       

11. Start the session

 webrtcdevobj.startCall(webrtcdevsessionobj)

Widgets

Currently available widgets are * Chat * Fileshare * Timer * Draw * Screen Record * Screen Share * Video Record * Snapshot * Minimising/ maximising Video * Mute (audio and/or video) * Draw on Canvas and Sync * Text Editor and Sync * Reconnect

Description of Widgets with SDK invocation

1. Chat

User RTCDataConnection api from webRTC to send peer to peer nessages within a session. If camera is present the SDK captures a screenshot with user's cemars feed at the isnatnt of typing the message and send along with the message.

When the chat widget is active , if the dom specified by the container id is present then webSDK uses as it is, else it creates one default box

{
    active: true,
    container: {
        id: "chatContainer"                 // dom id of the chat conatiner 
    },
    inputBox:{
        text_id:"chatInputText",            // dom id of the chta's input box
        sendbutton_id:"chatSendButton",     // dom id for the chat's send button
        minbutton_id:"minimizeChatButton"   // dom id for minimizing the Chat conaginer 
    },
    chatBox:{
        id: "chatBoard"                     // dom id for the chat board where all messages are dispalyed 
    },
    button:{                                // on and off button states for the chat widget button
        class_on:"btn btn-warning glyphicon glyphicon-font topPanelButton",  
        html_on:"Chat",
        class_off:"btn btn-success glyphicon glyphicon-font topPanelButton",
        html_off:"Chat"
    }
}

Upcoming : Adding emoticons to Chat

2. File-share

Uses the RTCDataConnection API from WebRTC to exchange files peer to peer. Progress bar is displayed for the chunks of file transferrred out of total number of chunks. Many different kindes of file transfer have been tested such as media files ( mp3 , mp4 ) , text or pdf files , microsoft pr libra office dicuments , images ( jpg , png etc ) etc .

File share widgets creates uses 2 containers - File Share and File List . If the dom ids of the container are not present on the page , the SDK crestes default conainers and appends them to page

The list of files with buttons to view , hide or remove them from file viewers are in file Viewer container . Displaying or playing the text or media files happens in file share container , which also has button to maximize , minimize the viewer window or in case of images to rotate them.

For divided file share container

{
    active : true,
    fileShareContainer : "fileSharingRow",                  // File sharing container
    fileshare:{                                             // components of file sharing container
         rotateicon:"assets/images/refresh-icon.png",       // rotate icon
         minicon:"assets/images/arrow-icon-down.png",       // min icon 
         maxicon:"assets/images/arrow-icon.png",            // max icon
         closeicon:"assets/images/cross-icon.png"           // close icon
    },
    fileListContainer : "fileListingRow",                   // File List container container 
    filelist:{                                              // components of file list conainer 
         downloadicon:"",                                   // icon donwload 
         trashicon:"",                                      // icon trash
         saveicon:"",                                       // icon save
         showicon:"",                                       // icon show
         hideicon:"",                                       // icon hide
    },
    button:{
        id: "fileshareBtn",                                 // dom for widget button to call file share
        class_on: "col-xs-8 text-center fileshareclass",
        html:"File"
    },
    props:{
        fileShare:"divided",                                // Can be divided for two particiapnts , chatpreview  , single for many participants  , hidden 
        fileList:"divided"                                  // same as aboev Can be divided , single   , hidden
    }
}

or for single file share container for all peers

    let filesharewidget = {
        active: true,
        fileShareContainer: "fileSharingRow",
        fileshare: {
            rotateicon: "assets/images/refresh-icon.png",
            minicon: "",
            maxicon: "",
            closeicon: "assets/images/cross-icon.png"
        },
        fileListContainer: "fileListingRow",
        filelist: {
            minicon: "",
            maxicon: "",
            downloadicon: "",
            trashicon: "",
            saveicon: "",
            showicon: "",
            hideicon: "",
            stopuploadicon: ""
        },
        button: {
            id: "fileshareBtn",
            class_on: "file-share",
            html: "File"
        },
        props: {
            fileShare: "single",   //Can be divided , chat preview  , single   , hidden
            fileList: "single"     //Can be divided , single , hidden
        },
        sendOldFiles: false        // If new participant join conf , or listener join , then should he receive old files or not
    }

3. Timer

Creates or assigns a timer for teh ongoing sesssion . Also displays the geolocation and timezone of the peers if perssion if provided . Timer can start upwards or downwards. Can be used for billing and policy control .

{
    active: true,
    type: "forward",                                        // Forwards timer starts from 0:0:00 goes thereafter, backward timer ticks backword from prespecified time limit
    counter:{                   
        hours: "countdownHours",                            // dom id for hours 
        minutes:"countdownMinutes",                         // dom id for mins
        seconds :"countdownSeconds"                         // dom if for seconds
    },
    upperlimit: {                                           // upperlimit of time for the session 
        hour:0 ,                                            
        min: 3 , 
        sec: 60 
    },
    span:{                                                  // dom ids for local and remote time labels
        currentTime_id:"currentTimeArea",
        currentTimeZone_id:"currentTimeZoneArea",
        remoteTime_id :"remoteTimeArea",
        remoteTimeZone_id:"remoteTimeZoneArea",
        class_on:""
    },
    container:{
        id:'collapseThree',
        minbutton_id:'timerBtn'
    },
    button :{
        id: 'timerBtn'                                      // dom for widget timer button to call
    }
}

4. Screen Record

Records everything present on the tab selected along with audio and displays recording as mp4 file. Use an extension and pre-declared safe-site to facilitate captuing the tab.

{
    active : true,
    videoRecordContainer: true,        // container for storing or displaying recorded video
    button:{                           // button to control screen control widget and its on / off states
        id: "scrRecordBtn",
        class_on:"btn btn-lg screenRecordBtnClass On",
        html_on:'',
        class_off:"btn btn-lg screenRecordBtnClass Off",
        html_off: ''
    }
}

5. Screen-share

One of the most powerful features of the SDK is to capture the current screen and share it with peer over RTC Peer connection channel. Simmilar to csreen record , uses an extension and pre-declared site ownership to capture the screen and share as peer to peer stream . Button for screen share has 3 states -

• install button for inline installation of extension from page ,
• share screen button and
• view button for incoming screen by peer .

{
    active : true,
    screenshareContainer: "screenShareRow",      // container to display screen being shared
    button:{
        installButton:{                          // widget button to start inline installation of extension
            id:"scrInstallButton",
            class_on:"screeninstall-btn on",
            html_on:"Stop Install",
            class_off:"screeninstall-btn off",
            html_off:"ScreenShare"
        },
        shareButton:{                                       // widget button to start sharing screen , deactivated once already active or when peer is sharig 
            id:"scrShareButton",
            class_on:"btn btn-lg on",
            html_on:'<img title="Stop Screen Share"  src=assets/images/icon_2.png />',
            class_off:"btn btn-lg off",
            html_off:'<img title="Start Screen Share" src=assets/images/icon_2.png />',
            class_busy:"btn btn-lg busy",
            html_busy:'<img title="Peer is Sharing Screen" src=assets/images/icon_2.png />'
        },
        viewButton:{                                        // button to view the icnoming screen share 
            id:"scrViewButton",
            class_on:"screeninstall-btn on",
            html_on:"Stop Viewing",
            class_off:"screeninstall-btn off",
            html_off:"View Screen"
        }
    }
}

6. Video Record

Records video-stream. Created for each peer video .

{
    active : true,
    videoRecordContainer : true,
    button:{
        class_on:"pull-right btn btn-modify-video2_on videoButtonClass on",
        html_on:"<i class='fa fa-circle' title='Stop recording this Video'></i>",
        class_off:"pull-right btn btn-modify-video2 videoButtonClass off",
        html_off:"<i class='fa fa-circle' title='Record this Video'></i>"
    }
}

7. Snapshot

Takes a snapshot from video stream . Will be created for each inidvidual peer video .

{
    active : true,
    snapshotContainer: true,
    button:{
        class_on: "pull-right btn btn-modify-video2 videoButtonClass",
        html_on:"<i class='fa fa-th-large' title='Take a snapshot'></i>"
    }
} 

8. Minimising/ maximising Video

To enable the user to watch video in full screen mode or to inimize the video to hide it from screen. Will be seprately created for each individual peer video .

{
    active: true,
    max: {
        button: {                                 // button to maximise the video to full screen mode 
            id: 'maxVideoButton',
            class_on:"pull-right btn btn-modify-video2 videoButtonClass On",
            html_on:"<i class='fa fa-laptop' title='full Screen'></i>",
            class_off:"pull-right btn btn-modify-video2 videoButtonClass Off",
            html_off:"<i class=' fa fa-laptop' title='full Screen'></i>"
        }  
    } ,
    min : {
        button: {                                  // button to minimize or hide the video 
            id : 'minVideoButton',
            class_on:"pull-right btn btn-modify-video2 videoButtonClass On",
            html_on:"<i class='fa fa-minus' title='minimize Video'></i>",
            class_off:"pull-right btn btn-modify-video2 videoButtonClass Off",
            html_off:"<i class='fa fa-minus' title='minimize Video'></i>"
        }  
    }                    
}

9. Mute (audio and/or video)

Mutes the audio or video of the peer video . Created for each peer video.

{
    active: false,
    audio: {
        active: false,
        button: {
            class_on: "pull-right videoButtonClass on",
            html_on: "<i class='fa fa-microphone-slash'></i>",
            class_off: "pull-right videoButtonClass off",
            html_off: "<i class='fa fa-microphone'></i>"
        }
    },
    video: {
        active: false,
        button: {
            class_on: "pull-right videoButtonClass on",
            html_on: "<i class='fa fa-video-camera'></i>",
            class_off: "pull-right videoButtonClass off",
            html_off: "<i class='fa fa-video-camera'></i>"
        }
    }
}

10 . Reconnect

Allows a user to recoonect a session without refreshing a page . Will enable him to drop the session and create a new one.

{
    active : false,
    button : {
        id: "reconnectBtn",
        class:"btn btn-success glyphicon glyphicon-refresh topPanelButton",
        html:"Reconnect",
        resyncfiles : false
    }
}

11. Cursor

{
    active: false,
    pointer: {
        class_on: "fa fa-hand-o-up fa-3x"
    },
    button: {
        id: 'shareCursorButton',
        class_on: "pull-right videoButtonClass On",
        html_on: "<i class='fa fa-hand-pointer-o fullscreen'></i>",
        class_off: "pull-right videoButtonClass Off",
        html_off: "<i class='fa fa-hand-pointer-o fullscreen'></i>"
    }
}

12. Inspector

{
    active: true,
    button:{
        id: "ListenInButton",
        textbox : "listenInLink"
    }
}

13. Debug

To turn debug on

{
  debug: false
} 

14. Help

Activates the help log by start captures console logs , info , messages , warning in a retreivabe array. Can also send the logs tto pre-specified URL as paylaod and/or display the logs in dom as specified

{
  active: true, 
  helpContainer : "help-view-body",
  screenshotContainer: "help-screenshot-body",
  descriptionContainer: "help-description-body"
}

15. Stats

Collects network and webrtc stats. Captures them in logs and displays on dom as specified

{
  active : true, 
  statsConainer : "network-stats-body"
}

16. Draw

{
    active: true,
    drawCanvasContainer: "drawBoardRow",
    button: {
        id: "draw-webrtc",
        class_on: "icon-pencil On",
        html_on: '',
        class_off: "icon-pencil Off",
        html_off: ''
    }
}

Assign individual widgets to a json object called widgets

{
    debug: false,
    reconnect: {
        active: false
    },
    timer: timerwidget,
    chat: chatwidget,
    fileShare: filesharewidget,
    mute: mutewidget,
    videoRecord: videorecordwidget,
    snapshot: snapshotwidget,
    cursor: cusrsorwidget,
    minmax: minmaxwidget,
    drawCanvas: drawwidget,
    screenrecord: screenrecordwidget,
    screenshare: screensharewidget,
    listenin: listeninwidget,
    help: helpwidget,
    statistics: {
        active: false,
        statsConainer: "network-stats-body"
    }
}

NAT traversal

From variety of options you can choose

1. Only free STUN from google

var iceservers_array = [{urls: ["STUN stun.l.google.com:19302"]}];

ref : https://stackoverflow.com/questions/20067739/what-is-stun-stun-l-google-com19302-used-for

2. Xirsys free account for TURN

3. self-hosted COTURN

Goto https://coturn.net/turnserver/ to choose the version you want to download , at the time of writing this 4.5.2 was the latest

wget https://coturn.net/turnserver/v4.5.2/turnserver-4.5.2.tar.gz

goto https://packages.qa.debian.org/c/coturn.html the debian coturn package is documented at https://packages.debian.org/jessie/coturn

Install dependencies

sudo apt-get install libssl-dev
sudo apt-get install libsqlite3 (or sqlite3)
sudo apt-get install libsqlite3-dev (or sqlite3-dev)
sudo apt-get install libevent-dev
sudo apt-get install libpq-dev
sudo apt-get install mysql-client
sudo apt-get install libmysqlclient-dev
sudo apt-get install libhiredis-dev

https://quickref.common-lisp.net/cl-libevent2.html

build

./configure 
make 
sudo make install

After the build, the following binary images will be available:

1. turnserver
2. turnadmin
3. turnutils_uclient
4. turnutils_peer
5. turnutils_stunclient.
6. turnutils_rfc5769check

Adding to signalling server

var iceservers_array = [{urls: ["STUN stun.l.google.com:19302"]},
    {url: 'turn:[e-mail]:3478', credential: 'root'}];

supported RFC

• RFC 5766 - base TURN specs;
• RFC 6062 - TCP relaying TURN extension;
• RFC 6156 - IPv6 extension for TURN;

Event listeners

Implemented event listeners :

1. onLocalConnect

2. onSessionConnect

3. onScreenShareStarted

4. onScreenShareSEnded

5. onNoCameraCard

Keys and certs

To generate a CSR for external Certificate Authority such as Godaddy

openssl req -x509 -newkey rsa:4096 -sha256 -nodes -keyout ssl_certs/server.key -out ssl_certs/server.crt -subj "/CN=webrtc.altanai.com" -days 3650

Demo

open tab on chrome or mozilla browser and add a link to the https server using nodejs script https://127.0.0.1:8086/multiparty_fullfeatures.html

Other implementation of the SDK are

webrtc_quickstart - https://github.com/altanai/webrtc_quickstart

webrtc_usecases - https://github.com/altanai/webrtc_usercases

Extra

Following are the additional libraries packed with the project

Gulp Minify and concat the js and css files into minified scripts

Task Runner you can run gulp alone to minify and concat the js and css files into min-scripts

gulp

or can run grunt to concat , minifify , ugligy , push to git and npm all together

grunt production 

forever Keeps running even when window is not active

cd WebCall
forever start webrtcserver.js

Notification / Alerting

//tbd

creating doc

 ./node_modules/.bin/esdoc
  open ./docs/index.html

start with process manager pm2

To start the Server using PM2 ( a process manager for nodejs) , install pm2 globally

npm install pm2 -g

create a conf json

pm2 ecosystem

Add config to json

  apps : [{
    script: 'webrtcserver.js',
    watch: '.'
  }]

start pm2

pm2 start ecosystem.config.js 

with env

pm2 start ecosystem.config.js --env production

---

Working steps

1.create a new session

Navigate on browser https://localhost:8082/#2435937115056035 which creates websocket over socket.io wss://localhost:8083/socket.io/?EIO=3&transport=websocket

2.check for channel presence

first client message

[ "presence", 
  {
    channel: "2435937115056035"
    }
 ]

on the server side

 Presence Check index of  2435937115056035  is  false

websocket response from server ["presence", false]

3.If channel doesnt exist already create

client message to open channel

  [  "open-channel", 
    {
      channel: "2435937115056035", 
      sender: "gxh0oi2jrs", 
      maxAllowed: 6
     }
   ]

server response

 ------------open channel-------------  2435937115056035  by  gxh0oi2jrs
registered new in channels  [ '2435937115056035' ]
information added to channel { '2435937115056035':
   { channel: '2435937115056035',
     timestamp: '12/18/2018, 10:18:01 PM',
     maxAllowed: 6,
     users: [ 'gxh0oi2jrs' ],
     status: 'waiting',
     endtimestamp: 0,
     log:
      [ '12/18/2018, 10:18:01 PM:-channel created . User gxh0oi2jrs waiting ' ] } }
     

websocket response from server

  [  "open-channel-resp", 
   { 
    status: true, 
    channel: "2435937115056035"
    }
  ]

4.Join a session and check for channel presence

navigate another browser client to same session url such as https://localhost:8084/#2435937115056035?name=aa&email=[e-mail]

   check presence ["presence", {channel: "2435937115056035"}]
   
   ["presence", true]
  
   Presence Check index of  2435937115056035  is  true

5.If channel is present join the channel

  ["join-channel", {channel: "2435937115056035", sender: "2ilwvn9qq39",…}]
   
------------join channel-------------  2435937115056035  by  2ilwvn9qq39  isallowed  true

[ "join-channel-resp"
 {
 status: true, 
 channel: "2435937115056035", 
 users: ["gxh0oi2jrs", "2ilwvn9qq39"]
}]

Debugging help

CORS

Issue1 CORS exception prevents loading the connection to socket.io server

Access to XMLHttpRequest at 'https://domain.com:8083/socket.io/?EIO=3&transport=polling&t=NiEMZt0' from origin 'https://domain.com:8084' has been blocked by CORS policy: No 'Access-Control-Allow-Origin' header is present on the requested resource

solution1 Ensure that the resource is added to servers cors origin list . By default it works on same origin only.. Note : Same hostname but diff ports still counts as different origins Test cors

curl -H "Origin: https://domain:8084" --head https://domain.com:8083/socket.io

Issue2 Using wildcard

value of the 'Access-Control-Allow-Origin' header in the response must not be the wildcard '*' when the request's credentials mode is 'include'. The credentials mode of requests initiated by the XMLHttpRequest is controlled by the withCredentials attribute.

solution the requested origin for cross origin requests should be loaded to env varaible and will be refered by socket.io and rest api server and signaller

const allowedOrigins = ['/.*localhost.*/'
];
process.env.allowedOrigins = allowedOrigins;

Issue3 CORS with credentails

value of the 'Access-Control-Allow-Credentials' header in the response is '' which must be 'true' when the request's credentials mode is 'include'. The credentials mode of requests initiated by the XMLHttpRequest is controlled by the withCredentials attribute.

solution add credentials access when using cross origin or make credentails false

getusermedia Exceptions

Cases when user deosnt have ir isnt able to acces his audio/video devices due of any of reasons such as

• user has no webcam or microphone
• intentioanlly/accidentally denied access to the webcam
• plugs in the webcam/microphone after getUserMedia() code has initialized
• device is already used by another app on Windows
• user dismisses the privacy dialog

Rejections of the returned promise are made by passing a DOMException error object to the promise's failure handler. The DOMException interface represents an abnormal event

Possible errors are:

openrmc.webrtc.Errors = {
    NOT_SUPPORTED : 'NOT_SUPPORTED',
    CONSTRAINTS_REQUIRED : 'CONSTRAINTS_REQUIRED',
    AUDIO_NOT_AVAILABLE : 'AUDIO_NOT_AVAILABLE',
    VIDEO_NOT_AVAILABLE : 'VIDEO_NOT_AVAILABLE',
    AV_NOT_AVAILABLE : 'AV_NOT_AVAILABLE'
} ;

• AbortError - Although the user and operating system both granted access to the hardware device, and no hardware issues occurred that would cause a NotReadableError, some problem occurred which prevented the device from being used.

• NotAllowedError - One or more of the requested source devices cannot be used at this time. This will happen if the browsing context is insecure (that is, the page was loaded using HTTP rather than HTTPS). It also happens if the user has specified that the current browsing instance is not permitted access to the device, the user has denied access for the current session, or the user has denied all access to user media devices globally. On browsers that support managing media permissions with Feature Policy, this error is returned if Feature Policy is not configured to allow access to the input source(s). Older versions of the specification used SecurityError for this instead; SecurityError has taken on a new meaning.

• NotFoundError - No media tracks of the type specified were found that satisfy the given constraints. NotReadableError Although the user granted permission to use the matching devices, a hardware error occurred at the operating system, browser, or Web page level which prevented access to the device.

• OverconstrainedError - The specified constraints resulted in no candidate devices which met the criteria requested. The error is an object of type OverconstrainedError, and has a constraint property whose string value is the name of a constraint which was impossible to meet, and a message property containing a human-readable string explaining the problem. Because this error can occur even when the user has not yet granted permission to use the underlying device, it can potentially be used as a fingerprinting surface.

• SecurityError - User media support is disabled on the Document on which getUserMedia() was called. The mechanism by which user media support is enabled and disabled is left up to the individual user agent.

• TypeError - The list of constraints specified is empty, or has all constraints set to false. This can also happen if you try to call getUserMedia() in an insecure context, since navigator.mediaDevices is undefined in an insecure context.

ref : https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia

Errors on gulp

sourcemap related USe gulp-babel@8.0.0

arrow functions related use tarns compiler with preset env plugin for changes arrow function to normals ones before minifying

WSS errors

Issue1 net::ERR_CONTENT_LENGTH_MISMATCH 200 (OK)
solution This error is definite mismatch between the data that is advertised in the HTTP Headers and the data transferred over the wire. It could come from the following: Server: If a server has a bug with certain modules that changes the content but don't update the content-length in the header or just doesn't work properly. It was the case for the Node HTTP Proxy at some point (see here) Proxy: Any proxy between you and your server could be modifying the request and not update the content-length header.

Issue2 wss error connecting to webrtcserver like

{"code":3,"message":"Bad request"} 

or

Error: read ECONNRESET
Emitted 'error' event on TLSSocket instance at:
    at emitErrorNT (internal/streams/destroy.js:84:8)
    at processTicksAndRejections (internal/process/task_queues.js:84:21) {
  errno: -104,
  code: 'ECONNRESET',
  syscall: 'read'
}

Solution ECONNRESET error means that peer closed connection https://nodejs.org/api/errors.html . To overcome this example either set try catch and reconnect to prevent sever from crashing or client from disconnectinig or if you are running the http and wss server on the sae port like i was doing . Put them on seprate ports . I started seeing this problem a lot after I upgraded the http protocol version from https to http2 ( using native node module )
for example for http server

const app = http2.createSecureServer(options, (request, response) => {
    request.addListener('end', function () {
        file.serve(request, response);
    }).resume();
});
app.listen(properties.http2Port);

the again declare it separately for wss server

const server = require('http2').createSecureServer(options);
const io = require('socket.io')(server, {
    secure: true,
    serveClient: false,
    pingInterval: 10000,
    pingTimeout: 5000,
    cookie: false
});
io.origins('*:*');
io.on('connect', onConnection);
server.listen(properties.wss2Port);

Issue 3 WSS errors on socket.io as, error in connection establishment: net::ERR_SSL_PROTOCOL_ERROR
or WebSocket opening handshake was cancelled
solution recheck the session connection to socket.io , especially the ports and whther or not they are already in use

Issue 4 Error during WebSocket handshake: Unexpected response code: 403
solution Related to ECONNRESET

Issue 5 {code: 0, message: "Transport unknown"} code: 0 message: "Transport unknown" or Status Code: 400 Bad Request
solution Either specify same protocol on both client and servers ide or do not specify and transport protocol at all . For isntance this problem arises when server specifies websocket transport but client tries connecting over polling server specifying tarsnport websocket

ioServer(httpApp,{
    transports: ['websocket'],
    secure: true
})

But client tries polling connection https://localhost:8086/socket.io/?userid=iu02bk1b77g&sessionid=httpslocalhost8082clientindexhtm&transport=polling&t=N7ToS63

errors on SSL certs

Issue 6 CERT INVALID ERROR such as

NET::ERR_CERT_AUTHORITY_INVALID

Solution Since the certs are self signed , navigate to the wss port on http and allow permission under teh advanced button in scren below

Issue 7 GoDaddy SSL ecrts key gives no start line

  library: 'PEM routines',
  function: 'get_name',
  reason: 'no start line',
  code: 'ERR_OSSL_PEM_NO_START_LINE'

Solution first check whether the key file has valid certificate

openssl x509 -text -in file.key

Check if it prints an error including the text "unable to load certificate", then your file is not sufficient. See if the format is correct

openssl pkcs8 -in key.txt  -inform pem
Error reading key
140542854250944:error:0909006C:PEM routines:get_name:no start line:../crypto/pem/pem_lib.c:745:Expecting: ENCRYPTED PRIVATE KEY  

If not then re-save the file with charectar encoding UTF-8 and Line ending Unix/Linux

Errors on TURN

Issue 1 Pass issues on starting coturn

CONFIG ERROR: Empty cli-password, and so telnet cli interface is disabled! Please set a non empty cli-password!
0: : WARNING: cannot find certificate file: turn_server_cert.pem (1)
0: : WARNING: cannot start TLS and DTLS listeners because certificate file is not set properly
0: : WARNING: cannot find private key file: turn_server_pkey.pem (1)
0: : WARNING: cannot start TLS and DTLS listeners because private key file is not set properly

solution use no-auth in config or cli

Issue 2

0: : NO EXPLICIT LISTENER ADDRESS(ES) ARE CONFIGURED
0: : ===========Discovering listener addresses: =========
0: : Listener address to use: 127.0.0.1
0: : Listener address to use: 172.31.13.206
0: : Listener address to use: ::1

Solution Happens on ec2 container. Map the exteral initernal specifically in conf ot cli

turnserver -X EXT_IP/INT_IP 

or in config external-ip=EXT_IP/INT_IP

Issue 3 Assigning address

errno=99
Cannot bind local socket to addr: Cannot assign requested address

solution Check if the ports are open

ps -ef | grep 3478

and kil any processes that may be found running ref : https://github.com/coturn/coturn/issues/311

Issue 4 Both username and credential are required when the URL scheme is "turn" or "turns". at new WrappedRTCPeerConnection

var iceservers_array = [{urls: 'stun:stun.l.google.com:19302'},
    {url: "turn:[e-mail]:3478", credential: 'root'}];

Solution change this to

var iceservers_array = [{urls: 'stun:stun.l.google.com:19302'},
    {   username: "user",
        credential: "root",
        url: 'turn:media.xxx.com:3478'}];

Errors on git

update registry to "registry": "https://registry.npmjs.org " shelved

Reporting a Vulnerability

Create an issue https://github.com/altanai/webrtc/issues https://github.com/altanai/webrtc/issues

License

MIT

Todo: remove topIconHolder_ul