How to process remote audio/video stream on WebRTC server in real-time?

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I'm new to audio/video streaming. I'm using AntMedia Pro for audio/video conferencing. There will be 5-8 hosts who will be speaking and the expected audience size would be 15-20k (need to mention this as it won't be a P2P conferencing, but an MCU architecture).

I want to give a feature where a user can request for "convert voice to female / robot / whatever", which would let the user hear the manipulated voice in the conference.

From what I know is that I want to do a real-time processing on the server to be able to do this. I want to intercept the stream on the server, and do some processing (change the voice) on each of the tracks, and stream it back to the requestor.

The first challenge I'm facing is how to get the stream and/or the individual tracks on the server?

I did some research on how to process remote WebRTC streams, real-time on the server. I came across some keywords like RTMP ingestion, ffmpeg.

Here are a few questions I went through, but didn't find answers that I'm looking for:

  1. Receive webRTC video stream using python opencv in real-time
  2. Extract frames as images from an RTMP stream in real-time
  3. android stream real time video to streaming server

I need help in receiving real-time stream on the server (any technology - preferable Python, Golang) and streaming it back.

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