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Questions tagged [simplewebrtc]

SimpleWebRTC is an open source javascript library for joining and managing mesh p2p WebRTC conferences. Questions using this tag should be specific to this library.

Questions about the SimpleWebRTC library should use the simplewebrtc tag.

WebRTC is a technology stack for establishing (S)RTP and SCTP streams between peers. Peers are normally browser sessions.

p2p means "peer to peer," indicative of a transport between peers without a server or translation layer in the middle.

SimpleWebRTC normalizes browser API differences for peer connections and user media access. The library manages multiple RTCPeerConnections to provide a clean API for joining mesh (fully connected graph of p2p sessions) "rooms."

See https://simplewebrtc.com/ for more information and demos. See https://github.com/andyet/simplewebrtc for source code and issues.

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SimpleWebRtc - latest-v2.js stoped working

i'm using SimpleWebRtc on my webistes, but about week ago all sites with video peer (video chat) stoped working. I can see that my scripts try to include this file: "https://simplewebrtc.com/latest-v2.js", but it's not active - so video chat is not…
LukDan
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WebRTC GainNode not working on iOS (iPhone)

The Aim Change the gain (microphone volume) on the live stream. In the demo, you will be able to change from a low gain (0.1) to a high gain (1). When it is set to 0.1, you will hardly hear yourself, it's like a whisper mode. The Problem The code…
b4oshany
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simpleWebRTC video chat Failed to construct 'RTCPeerConnection': error

Ok so I am doing tutorials on webRTC and I've been using the following two tutorials to help me. Sitepoint tutorial and Scotch tutorial First thing to note, for the first tutorial even with the source code cloned from github here:…
RudolphRedNose
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Uncaught Error: This Firebase project requires import into Google Cloud Platform to resume service

Getting this Uncaught Error: This Firebase project requires import into Google Cloud Platform to resume service. When I am trying to conference video call through Webrtc. I got this url-…
Nitesh S Chauhan
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WebRTC - Networking issue on Codelab STUN/TURN servers

this is my first question here in stackoverflow, I am very blocked and I need some help here. First of all, thank you very much. I am trying to develop a WebRTC solution in a web app to stream video from different machines. I am developing it using…
dev_ed
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Is it possible to implement a MCU with SimpleWebRTC and signalmaster? How?

I have a peer to peer videoconference app using simpleWebRTC and signalmaster for signaling. When more than 4 users connects the stress it causes on the network and the TURN server is too big, so I was thinking, is it possible to implement a MCU in…
ee11131
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electron + simplewebrtc workable?

Does anyone use electron + simplewebrtc successfully? I used simplewebrtc to build a web app. It works fine in browsers. When I migrated it into electron, something weird happened. At first I used two copies of my electron app to make them connect…
Daniel
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AWS EC2 websocket over https not working

websocket on my ec2 instance working fine with http://live.talkrecruit.com:8888/socket.io/ However i am not able to connect it over ssl https://live.talkrecruit.com:8888/socket.io/ and ssl is installed correctly on this instance which works fine on…
Rizwan
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Xirsys v3 demo setup

I am attempting to get the demos project setup for xirsys v3 on windows 10. I have updated the credentials for the node server with my credentials from the portal but I receive a timeout error when entering the room. It is a websocket connection…
user2798927
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Failed to set session description: OperationError: Failed to set remote answer sdp: Called in wrong state: STATE_INPROGRESS

I used Angular JS : - Getting Room Id/Token from server side to connect web socket Following code used in application - app.controller("videoCallingController", ["$scope", "$location", "$rootScope", "$localStorage", 'AuthenticationService',…
Shivprasad P
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I'm trying to display webrtc format like :https://test.webrtc.org/. but it's not working

I'm a newbie in webrtc. I'm trying to use https://github.com/webrtc/testrtc this code but it's not working. showing some file missing errors. I wanna show like this : https://test.webrtc.org/
Tausib Raut
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WebRTC createOffer error

Using Firefox 53.0.2 (32bit) on Windows 10 64 bit. I am getting error when trying to createOffer on a RTCPeerConnection in which I have not added any local media stream and set var sdpOptions = { 'mandatory': { 'OfferToReceiveAudio': true,…
user1428926
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TURN server connection fails while trying to connect via SimpleWebRTC

I'm trying to make videochat app using SimpleWebRTC, Signalmaster and Coturn. On the webpage everything works fine when I try to connect from one tab to another in same browser or over LAN. I'm encountering problems while connecting between…
Alexander Lisovyi
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SimpleWebRTC Peer Discovery not working properly

I am using SimpleWebRTC library found here: https://simplewebrtc.com I got the signal-master running which has STUN/TURN configured properly. It's capable of detecting other peers so I assume STUN/TURN is functional. My problem is that when a peer…
user3186337
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webrtc per to per video chat but only need one side to send video to another

I read a lot of example about webrtc but I can't understand how to chat video p2p between A and B, but just only need A sends stream video to B using p2p connection, How to do this? I have tried to disable video local in B {video : false} but it…
nobjta_9x_tq
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