I am currently trying to challange myself and write a little meeting tool in order to learn stuff. I am currently stuck at the audio part. My audio stack is working, each user sends an OPUS stream (20ms packets) to the server. I am not thinking about how I will handle the audio. The outcome shall be: All users receive all audio, but not the own audio (so he does not hear himself). I want the meeting to support as many concurrent users as possible.

I have the following ideas, but none feels quite right:

1. send all audio streams to all users, which would mean bigger traffic, mixing would be done on the client side
2. mix the audio on a per-user-basis, which means if I have n users I would need to include n encodings in each frame.
3. mix the whole audio together, for alle users, send it to all users, each user will receive a second opus package containing the "own" opus audio packet which was sent to the server (or it will be numbered and stored on the client side so it does not need retransmission). I dont know if after decoding I can remove the "own" audio from the stream without getting some unclean audio.

How is this normally done? Are there options I am missing? I have profiled all steps involved, the most expensive part is encoding (one 20ms encoding takes about 600ns), decoding and mixing need near to no-time at all (5-10ns for each step).

Currently I would prefer option 3, but I do not find informations if the audio will be clean or if it will result in washy audio or some cracking.

The whole thing is written in C++, but I did not include the tag since I dont need code examples, just informations on this topic. I tried googling a lot, read a lot of documentation of opus, but did not find anything related to this.