In Google Meet, if you click “Present Now” then “A Chrome Tab”, it brings up a window that lets you share from another tab. This is different to the standard Screen Capture API as if you simply run you get a window where you can choose between sharing your whole screen, sharing a window, or sharing a tab. I would
Tag: webrtc
WebRTC change microphone or webcam while in call
I am trying to figure out how to change the microphone or webcam while you are in a videochat with someone. I have been now trying for a few days and nothing works. I was following this example, but it seems it is much harder to achieve the change while someone is already connected. The issues I have: If I
Unable to change camera / stream for WebRTC call
Source: https://github.com/anoek/webrtc-group-chat-example/blob/master/client.html I’m trying to modify this Webrtc example to add the ability of changing camera (Cross-browser support). Normal usage works perfectly, after changing camera, failed in renegotiation. 1) Get a list of devices via navigator.mediaDevices.enumerateDevices() 2) Change local_media_stream after getting new stream 3) Trigger renegotiation function (Copied from line 132 of Source code) I believe that my approaches are
Get data URL from video stream?
I have a video (webm) capture script that works fine. It records video then offers it as a download. The pertinent part of the code is this: This works, as I say. However, the console says that passing media streams to URL.createObjectURL is deprecated, and I should use HTMLMediaElement srcObject instead. So I changed it to: …and although everything still
Bug reporter: Alternatives to getDisplayMedia?
I am trying to implement a bug reporter on my website. My goal is that the user will be able to describe the problem audibly and record the browser tab while walking through the problem. The bug report will then just be a video file, which can be emailed to me. It appears that the proposed navigator.mediaDevices.getDisplayMedia is exactly what
Need clarification on Kurento’s API to connect webRTCEndpoint to RTPEndpoint
I am attempting to use Kurento’s bridging of webRTCendpoint to RTPendpoint. The webRTCendpoint client is a Chrome browser. The RTPendpoint client is a SIP server (proxy/B2BUA). Here is the basic code or pseudo-code I have (I am using Kurento-client.js in my app server): I have several questions: Is the overall structure correct? What is the use of webRTCEndpoint.gatherCandidates? The documentation
Combining audio and video tracks into new MediaStream
I need to get create a MediaStream using audio and video from different MediaStreams. In Firefox, I can instantiate a new MediaStream from an Array of tracks: Unfortunately, this doesn’t work in Chrome: ReferenceError: MediaStream is not defined Is there an alternative method in Chrome for combining tracks from separate streams? Answer still vendor-prefixed with webkit:
Port Knocking UDP with Javascript in Browser
So I am trying to send a port knock sequence from javascript. The TCP part is simple enough with websockets. I’ve read that WebRTC is the closest thing the browser offers to sending a UDP packet… but WebRTC is a lot to digest just to attempt to send a UDP “knock”. Is it even possible to “knock” via UDP using
WebRTC issue when using RecordRTC
We use the RecordRTC library to record user audio to our system. But an user got this error: Uncaught sample-rate must be under range 22050 and 96000 And I’m not sure what does it mean, as far as I could find on google it has something to do with his hardware (mic or headphone). Is that correct? There’s nothing much
Detecting that the peer’s browser was closed in a webrtc videochat
I’ve been implementing a webrtc videochat. Everything is working smoothly except for the case when the peer closes the browser. I’ve been trying to handle this event by implementing an onended callback on the remote mediastream. Though, this callback does not seem to ever be called. How can I detect that the peer’s browser has been closed or that the