webrtc

Web realtime communication SDK

!alt webrtc development

!Node.js Package

[![Build Status][BS img]][Build Status]
[![Dependency Status][DS img]][Dependency Status]
[![NPM Status][NS img]][NPM Status]

[Gitter]: https://gitter.im/altanai/webrtc?utm_source=badge&utm_medium=badge&utm_campaign=pr-badge
[Build Status]: https://travis-ci.org/altanai/webrtc
[Dependency Status]: https://david-dm.org/altanai/webrtc
[NPM Status]: https://www.npmjs.com/package/webrtcdevelopment

[GS img]: https://badges.gitter.im/altanai/webrtc.svg
[BS img]: https://api.travis-ci.org/altanai/webrtc.png
[DS img]: https://david-dm.org/altanai/webrtc.svg
[NS img]: https://nodei.co/npm/webrtcdevelopment.png

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 ).
!alt webrtc development architecture

$3

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 .

4. 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

$3

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 )
`shell script
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

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

To run the tests
`shell script
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 .
`shell script
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 .
`shell script
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 :
`javascript
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 :
`javascript
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 ) :
`javascript
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)
`javascript
var widgets={ }
`
Set widgets and their properties


8. Creating session

Get session id automatically
`javascript
sessionid = webrtcdevobj.makesessionid("reload");
`
or get session name from user
`javascript
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 .
`javascript
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
`javascript
webrtcdevsessionobj = webrtcdevobj.setsession(local, remote, incoming, outgoing, session, getWidgets());
`

11. Start the session
`javascript
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

$3

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

`json
{
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

$3

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
`json
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

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 .

`json
{
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
}
}
`

$3

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.

`json
{
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: ''
}
}
`

$3

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 .

`json
{
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:'',
class_off:"btn btn-lg off",
html_off:'',
class_busy:"btn btn-lg busy",
html_busy:''
},
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"
}
}
}
`

$3

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:"",
class_off:"pull-right btn btn-modify-video2 videoButtonClass off",
html_off:""
}
}
`

$3

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:""
}
}
`

$3

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:"",
class_off:"pull-right btn btn-modify-video2 videoButtonClass Off",
html_off:""
}
} ,
min : {
button: { // button to minimize or hide the video
id : 'minVideoButton',
class_on:"pull-right btn btn-modify-video2 videoButtonClass On",
html_on:"",
class_off:"pull-right btn btn-modify-video2 videoButtonClass Off",
html_off:""
}
}
}
`

$3

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: "",
class_off: "pull-right videoButtonClass off",
html_off: ""
}
},
video: {
active: false,
button: {
class_on: "pull-right videoButtonClass on",
html_on: "",
class_off: "pull-right videoButtonClass off",
html_off: ""
}
}
}
`

$3

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
}
}
`

$3

`
{
active: false,
pointer: {
class_on: "fa fa-hand-o-up fa-3x"
},
button: {
id: 'shareCursorButton',
class_on: "pull-right videoButtonClass On",
html_on: "",
class_off: "pull-right videoButtonClass Off",
html_off: ""
}
}
`

$3

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

$3

To turn debug on
`
{
debug: false
}
`

$3

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

$3

Collects network and webrtc stats. Captures them in logs and displays on dom as specified
`json
{
active : true,
statsConainer : "network-stats-body"
}
`

$3

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

$3

`json
{
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

$3

`js
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

$3

$3

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
`shell
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
`shell
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
`shell
./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
`js
var iceservers_array = [{urls: ["STUN stun.l.google.com:19302"]},
{url: 'turn:user@media.brightchats.com: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
`shell script
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
`

$3

To start the Server using PM2 ( a process manager for nodejs) , install pm2 globally
`shell script
npm install pm2 -g
`

create a conf json
`shell script
pm2 ecosystem
`

Add config to json
`json
apps : [{
script: 'webrtcserver.js',
watch: '.'
}]
`
start pm2
`shell
pm2 start ecosystem.config.js
`
with env
`shell
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
`shell script
[ "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
`shell script
[ "open-channel",
{
channel: "2435937115056035",
sender: "gxh0oi2jrs",
maxAllowed: 6
}
]
`

server response
`shell script
------------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
`shell
[ "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=abc@gmail.com
`shell
check presence ["presence", {channel: "2435937115056035"}]

["presence", true]

Presence Check index of 2435937115056035 is true
`

5.If channel is present join the channel

`shell script
["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

$3

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

$3

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:
`json
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

$3

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

$3

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
`json
{"code":3,"message":"Bad request"}
`
or
`shell script
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
`javascript
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
`javascript
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
`javascript
ioServer(httpApp,{
transports: ['websocket'],
secure: true
})
`
But client tries polling connection
https://localhost:8086/socket.io/?userid=iu02bk1b77g&sessionid=httpslocalhost8082clientindexhtm&transport=polling&t=N7ToS63

$3

Issue 6 CERT INVALID ERROR such as
`shell script
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
!CERT_AUTHORITY


Issue 7 GoDaddy SSL ecrts key gives no start line
`shell script
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
`shell script
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
`shell script
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

$3

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
`shell
turnserver -X EXT_IP/INT_IP
`
or in config external-ip=EXT_IP/INT_IP


Issue 3 Assigning address
`shell
errno=99
Cannot bind local socket to addr: Cannot assign requested address
`
solution Check if the ports are open
`shell
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
`json
var iceservers_array = [{urls: 'stun:stun.l.google.com:19302'},
{url: "turn:user@media.xxx.com:3478", credential: 'root'}];
`
Solution change this to
`json
var iceservers_array = [{urls: 'stun:stun.l.google.com:19302'},
{ username: "user",
credential: "root",
url: 'turn:media.xxx.com:3478'}];
``

$3

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

Reporting a Vulnerability

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

$3

MIT

Todo:
remove topIconHolder_ul