Comments (13)
Sure; you can host any experiment on your own web server.
However, only these two requires a node.js
server:
All experiments are HTML/JS based; so just copy/paste and done!
from webrtc-experiment.
The Node.js server will only establish connection or will it handle the streaming of the videos too? I want develop something that doesn't involve any server for video streaming. Just peer to peer video conferencing. I know a server is must to establish a connection. What happens after that? Does that play any role in the actual conferencing?
from webrtc-experiment.
Rmrinmoy, node.js only establish connection
from webrtc-experiment.
The Node.js server will only establish connection or will it handle the streaming of the videos too?
Only connection.
I want develop something that doesn't involve any server for video streaming. Just peer to peer video conferencing. I know a server is must to establish a connection. What happens after that?
In Offer/Answer model; a signaling gateway is mandatory however signaling server is not. Signaling can be done by copy
/paste
.
Does that play any role in the actual conferencing?
Node.js is good client for instant messaging. That's why it can be used for signaling too.
What signaling really is? — Just an exchange of session descriptions required for initial handshake.
You can use Node.js to share custom activities of the participants e.g.
- If someone leaves the room
- If someone mutes the audio/video
- If someone willing to share the screen i.e. renegotiation process
Etc.
from webrtc-experiment.
Thank you very much for your help.
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Hi sir,
you have said that firebase used for signaling purpose , could you please define how to set data in firebase how actually we can use that firebase
from webrtc-experiment.
First of all, link firebase.js
:
<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
Then start using it:
// unique for your site; however, it is optional.
var channel = 'abcdef';
// "chat" is PUBLIC firebase instance
var firebase = new Firebase('https://chat.firebaseio.com/' + channel);
// fired for each new message pushed over firebase servers
socket.on('child_added', function (snap) {
data = snap.val();
alert('this is your ' + data);
});
// push your own message; it never overwrites old data
firebase.push('watever');
// or overwrite existing messages
firebase.set('overwrite');
Firebase documentation available here: https://www.firebase.com/docs/javascript/firebase/index.html
from webrtc-experiment.
Hello sir,
can u explain me
var firebase = new Firebase('https://chat.firebaseio.com/' + channel);
what this above line do actually...?
from webrtc-experiment.
- http://www.rtcmulticonnection.org/FAQ/
- https://www.webrtc-experiment.com/docs/WebRTC-Signaling-Concepts.html
- http://www.rtcmulticonnection.org/docs/sessionid/
- http://www.rtcmulticonnection.org/docs/channel-id/
- http://stackoverflow.com/questions/13504320/socket-io-namespaces-channels-co
It means that data transmitted/exchanged over this channel
will be received only by the user using same channel. It gives us private environment that can be used to exchange sdp/ice privately between users.
Unique-channels concept allows us setup private text-chatting rooms as well.
var firebase1 = new Firebase('https://chat.firebaseio.com/channel-unique-id-1');
var firebase2 = new Firebase('https://chat.firebaseio.com/channel-unique-id-2');
var firebase3 = new Firebase('https://chat.firebaseio.com/channel-unique-id-3');
All these three firebase instances are referencing to unique session on Firebase servers; data transmitted over one channel can't be received on other.
from webrtc-experiment.
Hello Sir,
i copied file "http://www.rtcmulticonnection.org/latest.js" and "firebase.js" and putted in my local pc now when i try to run following example it generates errors
Example ::
Open New Room
<script> var connection = new RTCMultiConnection().connect(); document.getElementById('openNewSessionButton').onclick = function() { connection.open(); }; </script>ERROR:-
edia hints: {
"audio": true,
"video": {
"mandatory": {
"minWidth": 640,
"minHeight": 360,
"maxWidth": 1280,
"maxHeight": 720,
"minAspectRatio": 1.77
},
"optional": []
}
} latest.js:2458
{
"constraintName": "",
"message": "",
"name": "PermissionDeniedError"
} latest.js:3612
connection.onMediaError latest.js:3612
_captureUserMedia.mediaConfig.onerror latest.js:398
Maybe microphone access is denied. latest.js:3612
Maybe webcam access is denied. latest.js:3612
WebSocket connection to 'wss://s-dal5-nss-16.firebaseio.com/.ws?v=5&ns=rtcweb&s=cKTcywdDFPrZoH12zYTvnNLCwrubscUH' failed: WebSocket is closed before the connection is established. firebase.js:48
Sir please help me to overcome with this
from webrtc-experiment.
Either another application is using your webcam (maybe Firefox) or you accidentally clicked "Deny" button. You should go to this page: chrome://settings/contentExceptions#media-stream
and search for your locahost. Both audio and video MUST be Allow
in the list.
from webrtc-experiment.
Sir, we can use your firebase for all connection? I would like to implement in my project but I do not sure if it is possible.
Thank you
from webrtc-experiment.
I am using RTCMulticonnection for audio video broadcasting and created custom account with xirsys and used own credentials for stun and turn connection and firebase.io for signalling. However, Xirsys states that it provides signalling server for webrtc so my concern is that how could we eliminate firebase io and use xirsys signalling.
from webrtc-experiment.
Related Issues (20)
- "WebRTC Video Conferencing Demos" link refer to nonexistant page
- screen sharing doesn't work on chrome HOT 2
- How can we control which port(s) will be used to create peer to peer connection in WebRTC? HOT 1
- ERROR - {"code":0,"message":"Transport unknown"}
- Two LED screens are spliced, and the recording recording files is 0 bytes.
- Scalable Broadcasting Camera Switch
- Stop after 5 seconds
- Test001123
- Anyone using Scalable Broadcasting in Production?
- Only users who are on the same network can communicate on socket.io
- muazkhan.com socket.io signaling server is down
- How to controller video quality ex. I want to save video only 480 or 720
- Facing issue on webrtc video conference using firebase
- Is it support G723 or G729
- Muaz is gone! HOT 1
- Conference Calling on Android
- Cleanup of getDisplayMedia demo HOT 1
- Can I share some Windows?
- Video Recording
- Can a shared surface select its derivative window?
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