Should I use RTP or WebRTC for local network audio communication

Question

I have a set of Raspberry Pi Zeros that I would like to use as a home intercom. I initially set them up to send audio to each other using golang with gRPC and bidirectional streaming, which works for short calls, but the lag builds up over time, so I think I need to switch to a real-time protocol like RTP or WebRTC. Since I already know the IP address of each device, and the hardware/supported codecs for each is the same, and they are all on the same network, is there any advantage to using WebRTC over using plain RTP? My understanding is that WebRTC mainly provides some additional security and connection orchestration like ICE and SDP, which I wouldn't necessarily need. I am trying to minimize resource usage since these devices are not as powerful as a phone or desktop. If I do use WebRTC, I can do the SDP signaling with gRPC or some other direct delivery method. Since there are more than 2 devices, I'm also curious about multicast functionality, which seems pure-RTP specific, while WebRTC (which uses RTP), doesn't necessarily support multicasting, and would require (n-1)! p2p connections. I'm very unclear/unsure about this point.

Also, does either support mixing audio channels natively, or would that need to be handled in the custom software?

Answer 1

You could use WebRTC, but you'd need to rig a signalling server, and a STUN / TURN server. These can be super simple and low capacity because everything is on a private network, but you still need 'em. The signalling server handles the necessary SDP interchange. Going full WebRTC might be overengineering this. (But of course learning to get WebRTC working can be useful.)

You've built out a golang infrastructure. Seeing as how you're on a private network, you could change up that program to send multicast UDP packets or RTP packets. Then you can rig your listeners to listen to them.

No matter what you do, you'll need to deal with the lag. A good way to do it in the packet world: don't build a queue of buffers ready to play. Instead, always put each received packet as the next-to-play packet, even if you have to overwrite a previously received packet. (That is, skip ahead.) You may get a pop once in a while, but with reasonably short packets, under 50ms, it shouldn't affect the user experience significantly. And the lag won't build up.

The oldtimey phone system ran on a continent-wide 8K synchronous clock. So lag was not an issue. But it's always a problem when audio analog-to-digital and digital-to-analog clocks aren't synchronized. That's true whenever they are on different devices. The slightest drift builds up over time. (RPis don't have fifty-dollar clock parts in them with guaranteed low drift.)

If all your audio sources run at the same sample rate, you can average them to mix them. That should get you started. (If you're using WebRTC in a browser, it will mix multiple sources for you. )