Bringing in remote guests over WebRTC

Studio's remote-caller bridge is a browser-based WebRTC channel that brings a remote guest into your mixer as if they were on a local mic. No Zoom, no Squadcast handoff, no separate recording app — the guest opens a link, you hear them, the multitrack records them.

How a remote-caller session works:

1. **Open the Callers panel** in your session (sidebar). Click "+ Invite caller."
2. **Generate an invite link** — the platform creates a per-session link with a per-caller token. Copy it.
3. **Send the link** — email, text, calendar invite, however you reach your guest. The link is valid until the session ends or you revoke it.
4. **Guest opens the link** — in any modern browser (Chrome, Edge, Safari, Firefox on desktop or mobile). Their browser asks for mic permission.
5. **Guest hears a "Ready" tone** when they're connected. You see their channel light up in the Callers panel with a level meter.
6. **Drag the caller's channel into the mixer** — they get a full channel strip just like a local mic (gain, gate, EQ, compressor, pan).

What the guest sees:

A minimal page with a mic-level meter, a "Mute" button, a "Disconnect" button, and a status line ("Connected to {{org name}} — {{episode title}}"). No software install. They see no other guests by default unless you enable group-mode.

Audio quality:

- The bridge negotiates Opus codec at up to 64 kbps mono — broadcast-quality voice over a normal home internet connection (~70 kbps upload).
- Echo cancellation is on by default; if your guest has good headphones they can disable it in their browser controls for a more natural sound.
- A 30-second pre-roll is recorded the moment the guest connects — useful for catching a great first reaction that happened during the level check.

Recording the caller cleanly:

The multitrack records each caller's audio on its own track (not just the mixed-down feed). So if you discover in post that one guest's audio was too quiet, you can re-balance just their track without affecting the others. This is the workflow that Zoom + DAW handoff can't give you.

When the network gets bad:

- The platform shows a "Network quality: poor" indicator on both sides.
- A guest with a dropping connection gets bumped to a backup audio mode (lower bitrate, mono). Recording continues.
- If the guest fully disconnects, their channel goes silent (not gated — actually silent) and a marker is dropped on the recording timeline so you can find that point in post.

Group-mode (multiple remote guests):

You can invite up to 4 remote callers per session. Each gets their own channel. They can optionally hear each other (for natural conversation) or be isolated (for one-at-a-time interviews where the host introduces each caller).

Privacy + retention:

The guest's audio is recorded the same way a local mic is — to the org's storage, retained per the org's retention policy. The guest is told (at connect time) that their audio will be recorded; they need to acknowledge before they can speak. The acknowledgement is captured in the recording log for compliance.