I am trying to record google meet audio and video using webRtc. I found mediaRecorder Api for same. But i am unable to record meeting, as it is only capturing my webcam and audio. How can i record a whole meeting through code in any language?
One solution is to use Neko to record conferences. It is like jibri but works for all any website!
It is a headless Chromium instance that runs on a server. You launch the neko instance and then join the conference call. You then can capture the web page's audio and video.
You can specify a RTMP output (Twitch, Youtube...) or you can save the files.
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I would like to know if it is possible to join a MS Teams, Google Meet or Zoom Meeting via the URL as a bot and record the video/audio from the call.
Is this possible using WebRTC. Any help is appreciated.
I have been using neko to do this. It is a headless Chromium instance that you can run on a remote server. I join my call using this headless instance.
Neko uses GStreamer so you can send things out via RTMP hasn't been merged to master yet but I bet you could also easily update to save the video frames to disk!
Making it into a bot will be a little harder though. You will have to script interaction with each site. Chrome has nice plugins to script site interaction, possibly that could solve the issue?
I have a product that can analyze video after inputting an rtsp url.
I would like to use a webcam to stream and feed my product the webcam rtsp.
How can I do that?
It will depend on the webcam you are using - most support RTSP but many do not publish the interface to access the stream as they are designed to be used with the webcam's own companion app.
There are some web resources which provide the RTSP urls for common web cams - you may find it hard to find a match as new versions of webcams roll out but it should give you a feel how to try accessing a vendors camera if you have a specific web cam you are testing against. Some examples (at the time of writing):
https://www.getscw.com/decoding/rtsp
https://soleratec.com/get-support/rtsp/
If you can't find the info for the camera you are using, and you have the companion app, you can also use a network sniffer tool like Wireshark (https://www.wireshark.org) and try to search the traffic for 'rtsp://' pattern.
If you just need to test your app and have access to a raspberry pi with a camera module you can also use this to generate an RTSP stream - there are several approaches for this but one I have found reliable is the v4l2rtspserver server:
https://github.com/mpromonet/v4l2rtspserver
There are specific instructions for setting it up on PI (https://github.com/mpromonet/v4l2rtspserver/wiki/Setup-on-Pi) and you can also verify it is working using VLC player on a laptop etc before testing in your specific application.
There are also a small number of test RTSP urls available on the web - the most reliable seem to be the one at this link provided by Wowza (again, link valid at time of writing):
https://www.wowza.com/html/mobile.html
I am trying to make an online examination portal. When students start the exam, their webcam will start automatically and record the stream live and store in the server. Invigilators will either watch the students live or they can watch the saved live streams later.
I researched about this and found WebRTC as a possible solution along with a gateway server like Kurento. But later found out that WebRTC is not supported in Safari, which is a setback! My application should run successfully in web portal in any modern browsers which includes safari and also in android or iphone.
So can anyone suggest a possible solution to my problem? Which technology should I use that can support all browsers and OS?
Also, it would be helpful if you can provide links to good documentation or tutorials.
Note from the future (2020): This answer really isn't accurate anymore.
WebRTC is one problem... capture from the camera with getUserMedia is another. Safari doesn't support either.
There is no video capture API in Safari currently. The only thing you can do is make a native app for iOS.
Worse yet, because of Apple's restrictive policies, alternative browsers, such as Chrome, are crippled on iOS as they aren't allowed to use their own browser engines.
Use standards based technologies like getUserMedia and WebRTC for your primary web-based application. If you decide that the economics of your situation enable it, you can make an iOS app to work alongside until Apple decides to participate in modern browser standards like everyone else.
You can use Mediadevices.getUserMedia (https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia) to capture webcam stream on browser (chrome and firefox).
To play with webcam stream on safari, you would have to use a pollyfill - https://github.com/Temasys/AdapterJS
To record the video/audio stream, you can make use of Media recorder api https://developer.mozilla.org/en-US/docs/Web/API/MediaRecorder (Note : recording stream is still a challenge in Safari as there is no support/pollyfill. However, it works perfectly on Chrome and Firefox latest versions).
Helpful demonstrations :
https://webrtc.github.io/samples/
https://mozdevs.github.io/MediaRecorder-examples/index.html
https://codepen.io/collection/XjkNbN/
https://hacks.mozilla.org/2016/04/record-almost-everything-in-the-browser-with-mediarecorder/
I want to let WebRTC encoded and play h264(NAL) stream(local file).
In the WebRTC tutorial, getUserMedia is use for get local camera connecting to the system, I don`t know if the getUserMedia function support
capture the local stream file like h264 stream.
If it doesn't work that way, may be I should modify WebRTC source code(I'm studying it).
Here is the question, If i change WebRTC code, how can i integration the new code into browser? Made it a plugin?
Firefox supports an extension to the <video> element that you can use to do this.
First, set the source of a video element:
v1.src = "file:///...";
Then you can call the (currently prefixed) mozCaptureStream or mozCaptureStreamUntilEnded function to get a MediaStream.
stream = v1.mozCaptureStream();
The proposed specification.
Note however that you need to ensure that the file is same origin with respect to the page. The same origin rules for file:/// are probably going to cause issues. Otherwise your MediaStream isn't going to be accessible to you. One way to ensure that is not to set the location directly, but to load the file using an <input type="file"> element.
As noted in other answers, Firefox currently only supports the baseline profile of H.264.
First, you are right getusermedia will not work for you. However, there are a couple of options.
Hack a stream together using RTCDataChannel. Breaking up the media stream and delivering each packet and then handling it on the client side.
Take a look at this demo for precorded media streams. I do not believe that H264 is addressed but it could help you on your way(probably for Firefox only)
Use some sort of webrtc breaker/endpoint that is native to stream the file. I know specifically that others(including myself) have streamed H264 to Firefox through the Janus-Gateway
Couple of asides:
Firefox only supports Baseline profiles in streaming h264 for a webrtc peerconnection
Chrome does not support h264 for webrtc at all
Are you trying to have getUserMedia return a h.264 encoded stream?
In which case, today it will only be possible with Firefox today, under some specific environment (cisco 264 plugin installed) and only for the base profile.
Chrome promised in november to add this capacity, but there is no timeline that I know of Expect at least Q2 2015.
Using our (temasys) commercial plugin you will soon be able to do that in IE and Safari.
Those are the only options on client side I can think of. On server side you can use whatever you want to transcode, including janus, kurento, powermedia, licode/lynkia, ....
Note: using other means like Datachannel or WebSocket are ok to transfer files, but would greatly reduce the user experience as you would not have all the added recovery (and security) mechanisms included in SRTP, DTLS, and would also not have specific mistreated media enhancements that are in webRTC like jitter, buffers, netQ, ect ...
I am developing a website in which video recording is the core feature. We would ideally want video recording to continue even if internet connection goes down. If user is recording the video and internet gets disconnected then the video may get locally saved and when internet connection resumes, the video gets uploaded to the server.
I have gone through some websites which provides APIs for recording video via webcam but they do not work even if internet connection goes down for a moment.
Any suggestions will be highly appreciated.
I guess for your case you need to offer an off-line application rather online streaming solution, then later the user can upload the video. PresentationTube at http://presentationtube.com uses this technique. As they said: PresentationTube offers a PowerPoint presentation recorder and video sharing network to help teachers, students and business professionals produce and share presentations in video format.