[audray]

I may be missing something but this appears to me to be a WebRTC application. It is very easy to build something like this using any modern browser's WebRTC APIs, just disable audio processing in the media constraints and munge the opus SDP to stereo, maybe play with the network buffer setting if you want to lower latency but the audio hardware and physical distance is going matter more there.

The developer states that no other software was suitable, and also that it's the first of its kind. Both of those statements are not accurate: there is nothing innovative in sublive that isn't in any of the apps listed below. Props to the developer though for scratching an itch.

https://en.wikipedia.org/wiki/Comparison_of_Remote_Music_Per...

[weepy]

It uses WebRTC for the video, but the audio latency of WebRTC is too large and uncontrollable.

As stated in the post, the audio uses a custom C++ UDP solution. As far as I know it's the first video calling app with very low latency audio.

[telesilla]

How do you deal with firewalls in that case? Sonobus has a similar problem that if you don't have accessible NAT you can't connect. You need a relay or central server which can get really expensive!

[weepy]

Basically similar issue to Sonobus. Relay could work buy it probably adds latency and it certainly adds complexity. I may know a way to improve some NAT configurations but need to do more research.

[audray]

Yeah one reason I like Jacktrip, you can optionally use servers to connect. Low latency is nice but it doesn't help in a performance when you can't connect to your group of musicians