AI Said
How it works — core concept
Your existing mixer is the key piece here. You don't need to change how RadioBOSS plays your scheduled audio — you simply add the caller's audio as
one more input channel on your mixer, exactly like your microphone or CD player. You control their volume, on/off, and EQ from the mixer just like any other source.
Caller → Phone Interface → Mixer (spare channel) → Main Mix → RadioBOSS (or encoder) → Broadcast
Studio Output → Mix Minus → Phone Interface → Caller hears you
The critical detail is
mix-minus — you send everything
except the caller's own voice back to them, so they don't hear an echo.
Option 1: Software-based (Low–Medium Cost)
Best for internet/cloud calling; works with Skype, Google Meet, Telegram, SIP.
Software needed
- Voicemeeter Banana (free virtual mixer) or Virtual Audio Cable – routes audio between apps and RadioBOSS
- Calling app: Skype, Google Meet, Telegram, or a SIP client
- RadioBOSS — use its MIC/Line Input section to bring the caller feed into the broadcast
How to connect
- Caller audio from app → virtual cable → Voicemeeter → spare input channel on your mixer
- Your main mix (minus caller) → virtual output → send back to the calling app
- RadioBOSS continues playing your scheduled audio as normal; its output goes to the mixer
- When you press MIC On in RadioBOSS (or route the mixer output to your encoder), the caller is heard live
Pros & Cons
Low cost (Voicemeeter is free)- No extra hardware beyond your computer
Relies on internet stability; PSTN-style landline quality not available- Managing mix-minus virtually takes careful setup
Option 2: Hardware Phone Patch / Hybrid (Professional Quality)
Best for real telephone lines (PSTN) or SIP with broadcast-grade audio. These units handle the line, echo cancellation, and mix-minus automatically.
Recommended hardware
Table
| Product | Type | Notes |
|---|
| JK Audio Innkeeper / ComPatch | Analog hybrid | Connects standard phone line to mixer; built-in echo cancel; balanced XLR I/O |
| JK Audio RemoteMix | Portable hybrid | Good for single lines; works with mobiles via adapter |
| Telos VX / Zephyr | VoIP/SIP system | Multi-line broadcast phone system; AES67/AoIP options; professional studio standard |
| Studio Technologies Phone Interface | Digital/SIP | Clean digital I/O; great for modern setups |
| Broadcast console with built-in phone | All-in-one | e.g., Axel Oxygen has integrated telephone hybrid |
Connection diagram
PSTN/SIP line → Phone Hybrid Unit → Audio Out → Mixer Channel N (fader controls caller level)
Mixer Aux/Send (Mix-Minus) → Phone Hybrid → Line In → back to caller
RadioBOSS output → Mixer Channels 1–2 → Main Mix → Encoder/Broadcast
Your Mic → Mixer → Main Mix → Broadcast
Pros & Cons
- Broadcast-grade audio; dedicated echo cancellation
- Simple fader control from your existing mixer
- Works with standard phone numbers
- Hardware cost (£300–£2,000+ depending on lines)
Option 3: Hybrid — Software + Simple Hardware
If you want phone-in capability without full broadcast gear:
- Smartphone → Bluetooth/USB interface to mixer (e.g., RØDECaster approach works with any mixer with a spare channel)
- SIP softphone on PC → USB audio device → mixer channel
- Use RadioBOSS's MIC input or feed your whole mixer output to RadioBOSS's encoder
Proven Setups — Examples
Example A: Community Station (Software-Focused)
- PC running RadioBOSS + Voicemeeter Banana
- Skype receives listener calls; audio routed via virtual cables to Voicemeeter
- Caller audio → Voicemeeter → USB interface → Mixer Channel 3
- RadioBOSS output → Mixer 1–2; Mic → Channel 1
- Mixer output → RadioBOSS MIC input → to stream
- Mix-minus set up in Voicemeeter so caller hears program without echo
Example B: Semi-Pro Station (Hardware Hybrid)
- JK Audio ComPatch connected to a standard BT/Openreach line
- ComPatch output → balanced XLR to Mixer Channel 4
- Mixer Aux Send (program minus caller) → ComPatch input
- RadioBOSS plays scheduled music → Mixer 1–2
- Host mic → Channel 1; Caller → Channel 4
- Mixer main out feeds your stream encoder — RadioBOSS runs as normal plus the live mix passes through
Example C: VoIP/SIP Professional
- Telos VX SIP phone system registered to station's SIP trunk
- VX audio over network → Dante/AES67 interface → mixer
- Each call appears on its own fader
- RadioBOSS handles automation; mixer handles all live elements
Key Tips & Best Practices
- Mix-minus is non-negotiable — if you send the full main mix back to the caller, they'll hear themselves delayed (echo). Use an auxiliary send on your mixer that excludes the caller's channel.
- Keep RadioBOSS independent — let it continue playing your playlist; treat the caller just like another live source. You can even use RadioBOSS's Voice FX or MIC button to "duck" music automatically when the line is live.
- Latency matters — keep buffer sizes low in your audio driver (10–30ms range) so conversation feels natural.
- Test without broadcast first — have a friend call in while you monitor locally; fix echo/latency before going live.
- Recording — consider recording the full mix separately if you want to archive shows.
Summary
- Is it done? Yes — many stations use RadioBOSS with live phone-ins daily.
- Cheapest path: Voicemeeter Banana + Skype/SIP + your existing mixer → free–low cost.
- Best quality path: JK Audio or Telos hybrid + your mixer → broadcast-grade, reliable, proper phone lines.
- RadioBOSS role: stays exactly as your playout system; the mixer integrates everything together.
Would you like me to help you work out the exact cabling and audio routing for your specific mixer model and computer setup?
