I’ve implemented client-side recording on our Jisti app, but the only audio I can capture is the mic of the user (me) but anyone who joins isn’t captured.
I know that I need to find the audio source and I can then make a stream and capture it, but I can’t find the element that is producing audio for other users.
The element does not produce an audio stream. The audio element created when a new user joins isn’t able to be recorded using
captureStream() because the result is that the audio stops in the call.
Using your own local recording implementation (which appears to be using the
getUserMedia WebRTC API) does not capture other user’s audio.
Does anyone have a solution for this?
I’ve attempted to use
captureStream and pipe this audio track to my recorder (using RecordRTC) but as I said, it basically kills the audio stream from the participant.