simplewebrtc

问题 I am getting this error on console while refreshing the page.. Everythng else works fine the Chats and everthing.. just the streaming part is not working NotSupportedError: MediaStreamError at module.exports (http://192.168.1.10:9966/index.js:3081:17) at Object.1.getusermedia (http://192.168.1.10:9966/index.js:4:1) at o (http://192.168.1.10:9966/index.js:1:265) at r (http://192.168.1.10:9966/index.js:1:431) at http://192.168.1.10:9966/index.js:1:460 while creating a video chat app This is my

来源： https://stackoverflow.com/questions/60740054/webrtc-multi-peer-connection-3-clients-and-above

来源： https://stackoverflow.com/questions/60740054/webrtc-multi-peer-connection-3-clients-and-above

问题 I am creating an Ionic 2 app with firebase and I need a way to listen to database changes (specifically on child_added) when the app is closed (I.e. in foreground,background and killed) Basically, I want to use WebRTC to make calls within the app like whatsapp and I am following this post - https://websitebeaver.com/insanely-simple-webrtc-video-chat-using-firebase-with-codepen-demo However, the only thing that puzzles me is how it will work when the app is closed. Can anyone please help me

问题 I am creating an Ionic 2 app with firebase and I need a way to listen to database changes (specifically on child_added) when the app is closed (I.e. in foreground,background and killed) Basically, I want to use WebRTC to make calls within the app like whatsapp and I am following this post - https://websitebeaver.com/insanely-simple-webrtc-video-chat-using-firebase-with-codepen-demo However, the only thing that puzzles me is how it will work when the app is closed. Can anyone please help me

问题 I want to implement a SFU in my WebRTC videoconference app. Jitsi Videobridge seems like a good option, but can I use it with any webrtc API? For instance, I'm using SimpleWebRTC, is it possible to implement a SFU in this case with Jitsi Videobridge? If so, how? If not, what are the alternatives? 回答1: From this thread it looks like Jitsi Videobridge support is not implemented in SimpleWebRTC, which is peer-to-peer only: the jitsi videobridge. But it's not integrated with simplewebrtc (which

问题 About a month ago, a Stackoverflow partner helped me with a big question, like changing the camera and the microphone during a conference. That question was answered in the following link: Using WebRTC how to choose mic and camera? After changing the camera and microphone, the previous media flow remains active. So the other people in the conference can not receive the new flow I have in some way. I would like to know how to renegotiate this new flow, if necessary. The library that I use for

问题 I'm following this tutorial, to make a simple example of WebRTC. But the remote video does not appear in any browser and Chrome does not show the error: Uncaught (in promise) DOMException: Error processing ICE candidate I made a log not setRemoteDescription method: peerConn.setRemoteDescription(new RTCSessionDescription(signal.sdp), function(){ alert('success') }, function(e){ console.log(e); alert(e)}); And then I get the following error: OperationError: Failed to set remote offer sdp:

问题 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- https://firebase.google.com/support/guides/firebase-web#import_your_project_to_the_firebase_console but I don't want to upload my project on cloud. 回答1: I got the solution. Go to this url https://console.firebase.google.com/ and login as gmail account. Click on "Add Project" You will get page like this See

问题 Does SimpleWebRTC has this feature to get data(video/audio) without giving permission to browser to use my camera/microphone? // create our webrtc connection var webrtc = new SimpleWebRTC({ // the id/element dom element that will hold "our" video localVideoEl: 'localVideo', // the id/element dom element that will hold remote videos remoteVideosEl: '', // immediately ask for camera access **autoRequestMedia: true,** debug: true, detectSpeakingEvents: true, autoAdjustMic: false, media: { video: