The problem is that networks themselves are lossy.
Lossless audio ensures all the data is preserved. But if the audio transport itself introduces losses or corruption, then your audio integrity suffers anyway.
Lossless audio is a base requirement for professional audio. It gives you bit-exact samples and no perceptual coding.
But it's not enough on its own. Public networks like the internet can introduce 1-2% of packet loss, especially where WiFi is used. A stream can be perfectly uncompressed and still produce an audible glitch every time the link hiccups — and that will ruin your take. So the audio transport HAS to ensure 100% integrity.
Kormix is built around this understanding. Here's how.
Each session uses a dedicated gateway on high-performance cloud infrastructure in many locations around the world — started when you start it, destroyed when you finish. Not a shared server you queue behind, and not a peer-to-peer connection that works or doesn't depending on what your router thinks of their router. Kormix automatically picks your closest region to minimise latency.
Nothing on that path re-encodes your audio. What leaves your DAW is raw PCM — the same samples, up to 32-bit float and 96 kHz. It's encrypted before it leaves your computer, and the gateway passes it through byte for byte: it doesn't decode it, re-encode it, mix it down or store it. No compression or other processing touches your mix — not in the plugin, not on the server, not in the browser.
Video and talkback audio flow independently. With completely separate routing paths, camera and talkback audio never compete with the main audio stream.
Every link loses packets. Home broadband, hotel wifi, a café, a transatlantic hop at eight in the evening. Transport protocols have to handle this loss somehow.
So the question is never does it lose packets. It's what the software does about it.
Give up on it
A packet that hasn't arrived by the time it was due is abandoned, and playback carries on without it. Sometimes the gap is concealed, sometimes you hear it. Either way those samples are gone for good.
Ask for it again
The receiver notices what's missing, requests it, and keeps requesting until it arrives. Nothing is abandoned. The replacement lands inside the reserve it was already holding, and you hear nothing at all.
The receiver buffers audio to a certain size, depending on your internet connection: this is the window in which corrupt or missing packets are re-requested. When something goes missing, the round trip to ask for it and get it back is typically a fraction of that reserve, so you hear nothing. When the link is having a genuinely bad moment, the reserve is what gets spent — and if it runs out, playback waits. What never happens is the audio being thrown away.
The same goes for a packet that arrives damaged. Encrypted audio carries an integrity check, so corruption is detectable — and the usual thing to do with it is discard it, leaving a permanent hole from a packet that actually arrived. Kormix checks each packet for integrity, and if it fails, that packet is re-requested. The replacement slides into the same reserve, and playback never notices.
Latency is an unavoidable reality of sending data across the internet. And the transport needs to buffer a certain amount to ensure integrity of delivery.
But this is hard to measure correctly, for different protocols over different links. You are working on audio, not measuring networks. Kormix takes out this pain. From connect time, Kormix measures the connection latency and automatically configures the available buffers according to a set number of presets, from short and tight, for very stable links, to bulletproof for a bad one. Kormix measures the round trip when you connect, sizes the buffer from it, and re-checks as conditions change. Nothing more to configure. Leaving you to work on the music.
Kormix performance presets - Indicative figures
| Setting | Best for | Typical ear-to-ear latency |
|---|---|---|
| Minimum | short, stable links where you want it tight | ~50 ms |
| Balanced Default | standard sessions; low latency with solid repair room | under 100 ms |
| Maximum | bad links, where you can't afford to glitch | ~1.5 s |
Distance is the part you can't configure your way out of — the buffer is sized from the round trip, so a longer link enlarges it whatever you pick.
These are ear-to-ear figures — what your collaborator hears against what you played, counting the buffering at both ends. That's a stricter measurement than the one-way figure usually quoted for this kind of thing, and we quote it that way on purpose. Your own interface and DAW buffer size move them. The Balanced preset is the default because it balances low latency with enough repair headroom for real-world links.
Just send the session web link to your collaborator. No install, no plugin, no account, no card — it opens in the browser they already have, on a laptop or on a phone.
They receive the full stereo stream, lossless FLAC-encoded audio: compressed for the wire, bit-exact on the other side, the same samples you sent. This means there's no compromise on audio quality - your guests get the full fidelity that you send. A guest can also turn on their own camera and microphone and be in the room properly, giving you the communication channels you need, when you need them.
Video encoding and decoding happen on the graphics hardware, on both macOS and Windows — not on the cores your plugins want. Talkback is a conversation rather than a stem, so it gets echo cancellation, noise suppression and automatic gain, none of which goes near your program audio.
Kormix records each incoming stream separately, not the mix — and before your faders, so muting a channel doesn't gut its recording. Drag the take straight into your DAW. If someone drops off and comes back, their recordings are still there.
Encrypted on both legs. Your session runs on a gateway that exists for it alone and is destroyed when it ends. No audio or video is stored anywhere — there's nothing to leak because there's nothing kept.
Download Kormix, start a session and send someone the link. The free tier is a real tier, not a demo — and the audio path is identical on every plan.