-
-
Save guest271314/c38042935db4e0131c1e0b68ca59f4ac to your computer and use it in GitHub Desktop.
| https://stackoverflow.com/questions/61022341/why-is-video-not-playing-on-recording-of-remote-stream | |
| This works without a problem. If I replace the local stream with a remote stream, same code, only received over webRTC, I see the first frame and nothing more... No errors... jsut one Frame. |
f you create an RTCPeerConnection on one tab
The connection part is not a problem. The amount of people are.
If user sends video to user via WebRTC why does the video need to be recorded? For the TODO
Just an other feature so that people can record their conference calls. (not related to this problem)
The MediaStream from client to server can be streamed to other peers. You already have the server operable.
yes that is how I do it atm. Above 20 people the cpu load is just to mutch everything else is working fine.
If the 2 second delay at Firefox is the only issue, Firefox users can be notified of that, correct?
Drawing on an canvas works without delay. https://webrtchacks.github.io/samples/src/content/wasm/libvpx/ maybe I'll do that on firefox, I also have problems with my current solution if I want to start two streams... for some reason mediaSource can only be there once?! (but have to look a bit deeper into this). And there semms to be a problem if someone joins midstream, I need to find a way around that (or restart the recording?)
for some reason mediaSource can only be there once?!
Not certain what is being described?
And there semms to be a problem if someone joins midstream, I need to find a way around that (or restart the recording?)
Joined how? Video and audio? You can use OfflineAudioContext or ChannelMerger and ChannelSplitter to result in multiple audio streams that are all recorded https://stackoverflow.com/questions/40570114/is-it-possible-to-mix-multiple-audio-files-on-top-of-each-other-preferably-with.
If a user exists and another user enters the communication, one approach could be to have one MediaStream that is recorded and use RTCRtpSender.replaceTrack() to replace audio and video tracks in "midstream".
WebRTC, the standard way (Client->Server->Client), is all working fine (Video and Audio). Only problem is the server load on video on many people.
So to solve this I record the video on the server with mediaRecorder,send the chunks back to the clients via websockets, (this way I avoid webRTC video transcoding), and render it via MediaSource. That is working but with two little problems atm:
- If someone joins the session while the video is already running, he is not able to render the stream (because he is not starting on a keyframe or so)
- If I start the video (everything is woking), stop it, and start again I'll get somthing like "MediaStream is closed" on the client even if I did a "new MediaStream" on every new stream start. (but will change that to the canvas solution anyway I think).
If you create an
RTCPeerConnectionon one tab, create offer, then create anRTCPeerConnectionon another tab, store offer, pass offer to other tab, create answer, it should be possible to establish a connection, e.g., https://github.com/ldecicco/webrtc-demos; https://stackoverflow.com/questions/54980799/webrtc-datachannel-with-manual-signaling-example-please; https://stackoverflow.com/questions/20607002/webrtc-how-stun-server-feedbacks-the-sdp-and-ice-candidates; https://webrtchacks.com/the-minimum-viable-sdp/.If user sends video to user via WebRTC why does the video need to be recorded? For the TODO
The
MediaStreamfrom client to server can be streamed to other peers. You already have the server operable.VP8 and VP9 are, in general, lossy encoding algorithms. H264 should always give same value for recordings for same input video https://plnkr.co/edit/021iNc?preview. Am not sure about improving performance. You would need to run those tests, measure result.
If the 2 second delay at Firefox is the only issue, Firefox users can be notified of that, correct?