Use: var constraints = { video: { width: { ideal: 4096 }, height: { ideal: 2160 } } }; This will make the browser use the maximum resolution available, up to 4K. Works in Chrome 63, Edge 41 and Firefox 58. Citing MDN regarding the use of ideal: An ideal value, when used, has gravity, … Read more
There’s two sides to WebRTC. JavaScript APIs (getUserMedia) that allow an app to access camera and microphone hardware. You can use this access to simply display the stream locally (perhaps applying effects), or send the stream over the network. You could send the data to your server, or you could use… PeerConnection, an API that … Read more
You cannot upload the live stream itself while it is running. This is because it is a LIVE stream. So, this leaves you with a handful options. Record the audio stream using one of the many recorders out there RecordRTC works fairly well. Wait until the stream is completed and then upload the file. Send … Read more
The easiest way (although a bit “hacky”) to do this is to copy the source of RTCVideoCapturer to a local class withing your project and just use that local class. You can now edit the class to suit your needs. Even when not copying the file but actually building your own, it’s probably better to … Read more
Yes it is possible, but you have a few problems. 1) Red5 is terribly under-documented and buggy. 2) rtmpe/rtmps 3) performance/scalability 4) 1 through 3 are things you can overcome, but not without getting to know media distribution and all of it’s associated issues very intimately. By the time you’re done, you will be one … 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
Here is the sequence of operations to be used for Webrtc_AGC: Create AGC: WebRtcAgc_Create Initialize AGC: WebRtcAgc_Init Set Config: WebRtcAgc_set_config Initialize capture_level = 0 For kAgcModeAdaptiveDigital, invoke VirtualMic: WebRtcAgc_VirtualMic Process Buffer with capture_level: WebRtcAgc_Process Get the out capture level returned from WebRtcAgc_Process and set it to capture_level Repeat 5 to 7 for the audio buffers … Read more
Yes it is possible as the server can be one of the peers in that peer-to-peer session. If you respect the protocols and send the video in SRTP packets using VP8, the browser will play it. To help you build these components on other applications or servers, you can check this page and this project … Read more
WebRTC gives SDP Offer to the client JS app to send (however the JS app wants) to the other device, which uses that to generate an SDP Answer. The trick is that the SDP includes ICE candidates (effectively “try to talk to me at this IP address and this port”). ICE works to punch open … Read more
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