webrtcdevelopment
Advanced tools
webrtc based communication and collaboration client. Contains lot of experiments modules
Web realtime communication SDK
This is a ready to deploy webrtc SDK and SaaS for a customized and flexible communication and collaboration solution .
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 ).
WebRTC Web based real time communication framework. read more on webrtc
Node ( v10.0.0) Asynchronous event driven JavaScript runtime
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 .
grunt -verbose
Project is divided into 4 parts
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
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)
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
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
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
}
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
}
}
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: ''
}
}
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 -
{
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"
}
}
}
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>"
}
}
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>"
}
}
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>"
}
}
}
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>"
}
}
}
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
}
}
{
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>"
}
}
{
active: true,
button:{
id: "ListenInButton",
textbox : "listenInLink"
}
}
To turn debug on
{
debug: false
}
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"
}
Collects network and webrtc stats. Captures them in logs and displays on dom as specified
{
active : true,
statsConainer : "network-stats-body"
}
{
active: true,
drawCanvasContainer: "drawBoardRow",
button: {
id: "draw-webrtc",
class_on: "icon-pencil On",
html_on: '',
class_off: "icon-pencil Off",
html_off: ''
}
}
{
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"
}
}
From variety of options you can choose
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
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:
Adding to signalling server
var iceservers_array = [{urls: ["STUN stun.l.google.com:19302"]},
{url: 'turn:user@media.brightchats.com:3478', credential: 'root'}];
supported RFC
Implemented event listeners :
onLocalConnect
onSessionConnect
onScreenShareStarted
onScreenShareSEnded
onNoCameraCard
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
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
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
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
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=abc@gmail.com
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"]
}]
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
Cases when user deosnt have ir isnt able to acces his audio/video devices due of any of reasons such as
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
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
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
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
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:user@media.xxx.com: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'}];
update registry to "registry": "https://registry.npmjs.org " shelved
Create an issue https://github.com/altanai/webrtc/issues https://github.com/altanai/webrtc/issues
MIT
Todo: remove topIconHolder_ul
FAQs
webrtc based communication and collaboration client. Contains lot of experiments modules
We found that webrtcdevelopment demonstrated a not healthy version release cadence and project activity because the last version was released a year ago. It has 1 open source maintainer collaborating on the project.
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