I've tried researching this on my own, but fell short.
Let's say I use getUserMedia() and webRTC to make video calls. Is it possible to hook between the getUserMedia() and webRTC and modify the buffer before it is sent?
Simplest example: Sender base64 encodes the buffer before sending it, receiver decodes it and sending and receiving software is none the wiser that that is happening in the background?
You can make it with WebRTC Encoded Transform.
Here is an official example of End-to-End Encryption.