the answer from @Ichigo is correct, but it is a litte bit bigger. Every ICE contains ‘a node’ of your network, until it has reached the outside. By this you send these ICE’s to the other peer, so they know through what connection points they can reach you. See it as a large building: one … Read more
jeffreyveon’s answer will help reduce the chance that your user will choose deny, since she will only have to choose once. In case she does click deny, you can provide a message that explains why you need the permission and how to update her choice. For example: navigator.getUserMedia ( // constraints { video: true, audio: … Read more
As it was pretty much covered here, what you are trying to do here is not possible with plain, old-fashionned WebRTC (strictly peer-to-peer). Because as it was said earlier, WebRTC connections renegotiate encryption keys to encrypt data, for each session. So your broadcaster (B) will indeed need to upload its stream as many times as … Read more
Yes, an admin can override the prompts with a policy. VideoCaptureAllowedUrls Patterns in this list will be matched against the security origin of the requesting URL. If a match is found, access to audio capture devices will be granted without prompt. NOTE: This policy is currently only supported when running in Kiosk mode. On Windows, … Read more
For RTSP<->WebRTC / RTMP<->WebRTC conversions, you need to run some kind of WebRTC gateway / media server software that works with all these formats/protocols and can transmux between all of them. Try Wowza / Unreal Media Server / Flashphoner. https://en.wikipedia.org/wiki/Comparison_of_streaming_media_systems So in your case you want to publish the screen from browser to media server … Read more
Update: working example near bottom. This depends greatly on which browser you are using at the moment, due to an evolving spec. In the specification and Firefox, peer connections are now fundamentally track-based, and do not depend on local stream associations. You have var sender = pc.addTrack(track, stream), pc.removeTrack(sender), and even sender.replaceTrack(track), the latter involving … Read more
Edit: A nice implementation in github.io taken from comment to another answer( choose “relay” in IceTransports value): Test TURN Server following Benjamin Trent’s advice, I wrote the below code to test TURN server’s connectivity, works on both firefox n chrome: function checkTURNServer(turnConfig, timeout){ return new Promise(function(resolve, reject){ setTimeout(function(){ if(promiseResolved) return; resolve(false); promiseResolved = true; }, … Read more
It’s outdated code. It contains 6 problems that track the evolution of the WebRTC API. TL;DR: It doesn’t work because you’re not checking for errors and you’ve only tested one browser. 1) Old vendor prefixes (remove them): yourConnection = new RTCPeerConnection(configuration); theirConnection = new webkitRTCPeerConnection(configuration); // <– wrong webkit-names won’t work in Firefox or Edge. … Read more
The source of your confusion is likely that Google Chrome’s implementation of getStats() pre-dates the standard and has not been updated yet (the example you link to is Chrome-specific, so I presume you are using Chrome). If you were to try Firefox, you would find that it implements getStats() to the standard (however it does … Read more
The issue is not just the prefixed function name; the stream provided works differently in different browsers. Here, I’ll walk you through it. I assume you’ve already set up a video element in the variable called video. //I don’t usually like to overwrite publicly accessible variables, but that’s just me var getUserMedia = navigator.getUserMedia || … Read more