Hi @muaz-khan !
Thank for demo about Scalable-Screen-Broadcast. But i want change screen for this demo.
Can you please provider demo about it.

Thanks!

Hi muza-khan,
server.js is based on http connection.
But chrome web-browsers and web-view in android requires https connection.
I tried server.js in node.js. But I'd like to how to setup a http to https connection(redirect?).
Thank you in-advance.

Hi

I just download last google Chrome v48
Do you think that bug of audio forward is now fixed ?

Thanks

Hello Sir i am trying your node server and it runs successfully on "HTTP" but when i change the protocol from http to https then it stops working.Actually i want to run this sample over the internet so i really want https to work So please get back to me Thanks!!

I did install it with npm version 1.4.21:

cd /var/www/
npm install webrtc-scalable-broadcast

When I start

node .\node_modules\webrtc-scalable-broadcast\server.js

I get the error:

module.js:340
    throw err;
          ^
Error: Cannot find module '/var/www/.node_moduleswebrtc-scalable-broadcastserver.js'
    at Function.Module._resolveFilename (module.js:338:15)
    at Function.Module._load (module.js:280:25)
    at Function.Module.runMain (module.js:497:10)
    at startup (node.js:119:16)
    at node.js:906:3

Do I have to install nodes v1.0.1?

Hi guys!
Is there a way to implement the server using Express?

hi,
the screen capture is not effective.the received screen capture delay 7 to 8 seconds than original screen.and the original screen could not refresh itself smoothly.
how to improve it?thanks

Is it possible or going to be possible to use applications such as OBS for streaming? Or if not be able to manage the windows video or audio feeds? Is this a feature that is in the works? If so what kind of work will need to be done to integrate it?

RTCMultiConnection.js:4668 MediaStreamTrack.getSources is deprecated. See https://www.chromestatus.com/feature/4765305641369600 for more details.

This is in need for an update?

I want to use scalable broadcast with text. How can I do it ?

hello

i can do screensharing from windows machine and play those screen shared broadcasting in android is it possible ?

as i did try but getting error not able to play is there anyway ? in mobile chrome browser working and playing fine but in android apk not working even your demo also not work is there way to working ?

I would like someone to explain to me how the server works? I want to rewrite it in PHP because it does not know anything about nodejs ..

hi expert muaz,
i test webrtc-scalable-broadcast,if find some bugs in 20# and 21# post question.i try to resolve it.but i don't know why could not trigger leave event when webrtc-scalable broadcast webpage is closed.i can see numberOfConnectedUsers++ when user enter room,but could not run numberOfConnectedUsers-- when leaving in RTCmutilconnection.js.how could i revised it.could you give me a lead? thanks very much.

hi,nice muaz-khan
i find a bug in share-files.html or maybe in rtcmulticonnection.js.
i make a connect network like this at first:
peer 1<--peer 2<--peer 3
peer2 request file from peer1
peer3 request file from peer2
then i close peer 3
at this time,i transfer png file from peer1 to peer2.but the peer2 could not show the png image,only show 100% with the progressbar.

also,i do many similar experiments to test it,it is same on different connect network.
i trace the code,and find the bugs in share-files.html:

...
        var div = helper.div;
        if (file.type.indexOf('image') != -1) {
            div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><img src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;">';
        } else if (file.type.indexOf('video/') != -1) {
            div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><video src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;" controls></video>';
        } else if (file.type.indexOf('audio/') != -1) {
            div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><audio src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;" controls></audio>';
        } else {
            div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><iframe src="' + file.url + '" title="' + file.name + '" style="width: 80%;border: 0;height: inherit;margin-top:1em;"></iframe>';
        }

        if(connection.broadcastingConnection && connection.broadcastingConnection.numberOfConnectedUsers > 0) {
             connection.broadcastingConnection.send(file);
        }
...

if i close peer3,the connection.broadcastingConnection.numberOfConnectedUsers has not decreased 1,so in fact,the real connection.broadcastingConnection.numberOfConnectedUsers is 0 but your code here connection.broadcastingConnection.numberOfConnectedUsers is still 1.it is not correct.
so the bugs happen.
i don't know how to rectify it.could you help to remove the bugs?
by the way,i do the experiment on three tab of firefox.i have only one PC.i dont know whether the bugs will happen when i do the same experiment on three PC.
could you give me some ideas?thanks

hi,expert,
i test demos of webrtc-scalable-broadcast on my server,i only can see the joined message or stream message from node server console.but i cant see the video/screen/audio in joined terminal browser.what is the setting of webrtc-scalable-broadcast for broadcast my video/screen/audio?could you give me a suggestion?thanks

I can see there's no response on questions asked?

Here's my use case. I want to be able to listen to a presentation on one connection and provide real-time interpretation into a different language on a second connection. It's crucial that I keep bandwidth to an absolute minimum, because there could be many listeners -- but also because the listeners might be in the same venue as the speaker, sharing a single wi-fi connection. One-way, audio-only broadcast is the obvious way to go. The part I don't understand is whether the scaling will work directly, device-to-device.

Edit: Oops, I see now that it says "Android devices not tested." I would be happy to help test, if you give me detailed instructions.

Hello there can we make this work on IE and safari too by using third party temasys or skylink plugin that enables webrtc for these browsers? if yes then where should i make the required changes
Thanks

hi,expert
which peer will be selected by later joined partner in this Scalable Broadcast sample.is the connectivity network graph like a single line or like graph in your sample
how can i control which peer will be selected for a new joined partner.
if i want the new joined partner connect only PC peer.how to revise the sample code.

another question,if the connectivity network graph like graph in your sample.
,peer 2 is broken,then peer 5,6,7,8 will also be broken.how to resolve the problem?
thanks.

I start server.js and and a console message is printed:
info - socket.io started
However when I access http://localhost:8888 in the browser an error occurs and the server stops.
with the following log:
C:\Users\Fernando\Dropbox\www\aaa\node_modules\webrtc-scalable-broadcast\server.js:12
path.exists(filename, function (exists) {
^
TypeError: undefined is not a function
at Server. (C:\Users\Fernando\Dropbox\www\aaa\node_modules\webrtc-scalable-broadcast\server.js:12:10)
at Manager.handleRequest (C:\Users\Fernando\Dropbox\www\aaa\node_modules\webrtc-scalable-broadcast\node_modules\socket.io\lib\manager.js:588:28)
at Server. (C:\Users\Fernando\Dropbox\www\aaa\node_modules\webrtc-scalable-broadcast\node_modules\socket.io\lib\manager.js:119:10)
at Server.emit (events.js:110:17)
at HTTPParser.parserOnIncoming as onIncoming
at HTTPParser.parserOnHeadersComplete (_http_common.js:111:23)
at Socket.socketOnData (_http_server.js:343:22)
at Socket.emit (events.js:107:17)
at readableAddChunk (_stream_readable.js:163:16)
at Socket.Readable.push (_stream_readable.js:126:10)

Maybe I am missing something?...

I was playing with your Scalable Broadcast back in May and it was kind of working. At least I was able to start a presenter and add viewers and actually see the video. But I stopped looking at it for various reasons. Now I am looking at v3. Or trying. I cannot get anything to work. After install, when I browse to "IP":9001 I get browser error on FireFox and Chrome...saying cannot access. Have I forgotten to install something? I have sent Muaz an email, but I know Muaz is very busy so if someone else can assist?

Thanks,

Ray

I just started to look at WebRTC-Scalable-Broadcast. Initially I was skeptical of the concept but after thinking about it I figured I'd give it a try.

So I had the originator and 4 viewers up on my local LAN. Everything looked good. Lag was minimal.
However, after 10 minutes or so, all video froze. I have no idea why? I was browsing the internet for other reason and left the broadcast up. Where is the log file for this feature?

Oh, one more thing, once this happened, the camera light is still on. After doing another node server.js, nothing happens with the camera when I browse to IP:8888?

PS: I am very impressed with your work Muaz.
Thanks,
Ray

hi,our teacher,muaz
scaleable broadcast is a great job for us.i try my best to write my test code based on https://rtcmulticonnection.herokuapp.com/demos/Scalable-Broadcast.html for screen+audio funciton. i start from this sample,it is just because i want replace video stream with screen stream and i also want to save screen+audio stream to a webm file after stoping the streamming,it is similar with Scalable-Broadcast.html.i change
connection.session = {
audio: true,
video: true,
oneway: true
};
replace video:true with screen:true
and i could not get screen stream by firefox.
i am not familiar with webrtc.could any leader give me a sample for screen+audio.i am appreciated.
thanks

I managed to get this working, now I have an input field where I have to enter a "broadcast-id", what is that? and where do I get an example stream or video?

Hi

In chrome (android ) I can't select the camera of my phone.

Could you please add a button in order to switch front/rear camera?

regards
MaX

The code doesn't work. It stops at line 12 already: https://github.com/muaz-khan/WebRTC-Scalable-Broadcast/blob/master/server.js#L12

1.

On localhost:8888 I get:

Error: ENOENT, open '/var/www//index.html'

2.

On the console I get the error:

path.exists is now called `fs.exists`.

hi,muaz expert
i test WebRTC Scalable Broadcast in firefox browser.i select screen option.and when i joined,i get the error:
HTTPs i.e. SSL-based URI is mandatory to use screen capturing.
i think it means sample need https server.
then i revised the server.js.now i can use https.but i meet a new error
like these

navigator.mozGetUserMedia have been replaced by navigator.mediaDevices.getUserMedia RTCMultiConnection.js:3895:12
participant asked for availability RTCMultiConnection.js:2616:29
target user has no stream; it seems one-way streaming or data-only connection. RTCMultiConnection.js:3027:21
target user's SDP has? {
"OfferToReceiveVideo": false,
"OfferToReceiveAudio": false
} RTCMultiConnection.js:3038:1
accepting request from 1ifvblkl9eof20529 RTCMultiConnection.js:3060:13
sdp-constraints {
"OfferToReceiveAudio": false,
"OfferToReceiveVideo": false
} RTCMultiConnection.js:4207:1
optional-argument {
"optional": [
{
"DtlsSrtpKeyAgreement": true
},
{
"googImprovedWifiBwe": true
},
{
"googScreencastMinBitrate": 300
}
],
"mandatory": {}
} RTCMultiConnection.js:4221:1
rtc-configuration {
"iceServers": [
{
"url": "stun:stun.l.google.com:19302"
},
{
"url": "stun:stun.anyfirewall.com:3478"
},
{
"url": "turn:turn.bistri.com:80",
"credential": "homeo",
"username": "homeo"
},
{
"url": "turn:turn.anyfirewall.com:443?transport=tcp",
"credential": "webrtc",
"username": "webrtc"
}
],
"iceTransports": "all"
} RTCMultiConnection.js:4245:17
(getLocalDescription) peer createType is offer RTCMultiConnection.js:3956:17
RTCIceServer.url is deprecated! Use urls instead.
onSdpError: {}

what is wrong?
i think it is because i cant access google.com and other website showed above.but i cant change this.could expert give some advice to change my rtciceserver?
thanks

Can you please describe the solution of given issue by putting an image showing its scalability ?

Hi, I want to know what happen if on of the peers in the chain has a slow connection , it will slowdown all the new peers isnt it?

Is there any plan to implement something like webtorrent ?

Thanks in advance

We built a ubuntu 14.04 server, and can run the demo on http on port 8888
but then get the warning that getUserMedia() needs to be secure.

We purchased an SSL certificate (GoDaddy TLS 1.2) and installed

Then we get website unexpectedly closed the connection.

In server.js I changed this line
var app = require('http').createServer(function (request, response)

to
var app = require('https').createServer(function (request, response)

and we get the error
ERR_SSL_VERSION_OR_CIPHER_MISMATCH

hi,expert
i find a bug in share-files.html or maybe in rtcmulticonnection.js.
i make a connect network like this at first:
peer 1<--peer 2<--peer 3
peer2 request file from peer1
peer3 request file from peer2
then i close peer 3
at this time,i transfer png file from peer1 to peer2.but the peer2 could not show the png image,only show 100% with the progressbar.

also,i do many similar experiments to test it,it is same on different connect network.
i trace the code,and find the bugs in share-files.html:

    ...
            var div = helper.div;
            if (file.type.indexOf('image') != -1) {
                div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><img src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;">';
            } else if (file.type.indexOf('video/') != -1) {
                div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><video src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;" controls></video>';
            } else if (file.type.indexOf('audio/') != -1) {
                div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><audio src="' + file.url + '" title="' + file.name + '" style="max-width: 80%;" controls></audio>';
            } else {
                div.innerHTML = '<a href="' + file.url + '" download="' + file.name + '">Download <strong style="color:red;">' + file.name + '</strong> </a><br /><iframe src="' + file.url + '" title="' + file.name + '" style="width: 80%;border: 0;height: inherit;margin-top:1em;"></iframe>';
            }

-----------------------------------------------------------------------------------------------------
            if(connection.broadcastingConnection && connection.broadcastingConnection.numberOfConnectedUsers > 0) {
                 connection.broadcastingConnection.send(file);
            }
    ...

if i close peer3,the connection.broadcastingConnection.numberOfConnectedUsers has not decreased 1,so in fact,the real connection.broadcastingConnection.numberOfConnectedUsers is 0 but your code here connection.broadcastingConnection.numberOfConnectedUsers is still 1.it is not correct.
so the bugs happen.
i don't know how to rectify it.could you help to remove the bugs?
by the way,i do the experiment on three tab of firefox.i have only one PC.i dont know whether the bugs will happen when i do the same experiment on three PC.
could you give me some ideas?thanks

Hi muaz,

I have deploy on my ubuntu server broadcast, It works perfectly on desktop.
But on mobile i can view broadcast on mobile (on Chrome / IOS or Chrome / Android).

Event, when i use chrome on ipad/galaxy, i can't view broadcast that create from desktop. PS: i've used
WebRTC Scalable Broadcast (https://rtcmulticonnection.herokuapp.com/demos/Scalable-Broadcast.html)

I hope you understand that i said and suggest me a solution to resolve it. I've read some solution from you, such as build hybrid app by phonegap or native app but how about the mobile app?

Thanks !

Hey, is there a way that we can go live on youtube, facebook and instagram at the same time using this code.