Ask HN: Good resource to learn about audio streaming?

I have a remote browser app that needs to stream audio and after wrestling with this for many years I finally got the latency down for the audio to around 1 second. The main way to that I reduced like 5 seconds of lag was basically renicing the MP3 encoding process. but no matter what priority or what kind of different processes in the pipeline reniced I couldn't get the leg beyond 2 seconds. I discovered that most of the lag was due to actually encoding using lame on the server into MP3 and that if I used wave format even though it's a lot more bandwidth that brought the latency down to under 1 second.

The problem then is the wave format is a lot of bandwidth so when your bandwidth drops, the audio can become delayed or gappy. I tried downsampling it to 20KHz but that introduced comparable lag to MP3 encoding.

My idea now is that somehow I can air quotes break the audio stream into little pieces and stream it over webRTC to the browser but I don't understand anything about how audio is streamed in a way that is able to respond to changing bandwidth conditions.

My plan is to use pulseaudio RTP source to send it over the network and I want to pick it up with webrtc in the browser but maybe that's not possible, or wouldn't give me the low latency streaming that I want.

Anyway I know I simply do not understand enough about this and I couldn't really find any good resources where I can learn about it so I'm just asking here if anyone can point me in the right direction or give me some advice about this that would be awesome.