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Hi,
I'm using , RTCMultiConnection 1.4 with socket io for Signaling on chrome 29 or Firefox 23.
The thing is :
- The initiator init a session.
- Peers connect to session.
- When peer leave the ses…
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### Why?
Because current RTCMultiConnection presence detection model is kind of very poor. 98% of connections failed out of signaling and presence detection failures.
### How?
Need to separate API i…
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In the README file you state that the user should provide their own signaling server in order not to use the default lio.app signaling server.
This is understandable, but a typical user (e.g. a rea…
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Someone should clone muaz's WebRTC-Experiment and support it! It would be a shame to waste all of his good code!
Also, support of RTCMulticonnection if you are a Javascript and Node expert would …
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I don't really understand how this communication goes, can you clarify a little bit how this goes?
I'm using your RTCMultiConnection with the signaling on node.js with socket.io, I start by creating a…
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In my application, based on the Scalable video broadcast demo and https://github.com/muaz-khan/RTCMultiConnection/blob/master/demos/change-resolutions.html I applied **mute/unmute** method of the stre…
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Hello.
Party A (Firefox) initiated a call.
Party B answered a call.
Party B leaves a call.
Party A receives error:
"Error: track.stop is not a function
setDefaults/connection.stopMediaStream/
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https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html users ejection ejects user who started new session, not the one who joined. My firebase saves only userid of the room owner, i …
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After testing multiple demo systems like.
https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html
https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html
I not…
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I was looking to add a permanent user to the many to many audio conference demo. That way everyone will connect to the same room automatically when the users join. I think the randomly generated rooms…