aixtream
COMMUNICATION
The ideal audio call solution for SIP answering machines, remote communcation hubs or integration into PBX systems. You get full control over your communication network — managing multiple SIP connections and account has never been so easy.
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ENJOY MAXIMUM FLEXIBLILITY COMMUNICATION
- Use established SIP infrastructure, ad-hoc WebRTC conferences or direct RTP and SRT streaming.
- Aixtream supports an amazing range of audio I/O, including MADI, AES67/Ravenna, Livewire and Dante. Stay flexible, no matter from where or how you call.
- Call profiles ensure your call won't be rejected due to incompatibility, no matter what system lies on the remote end.
- Keep your audio safe! Redundant streaming ensures maximal audio quality even under difficult circumstances.
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functions and
modules
Get even more out of your radio streaming!
WebRTC
Deliver and receive audiofrom and to anywhere in the world with nothing but a phone or PC.
Virtual Audio Ports
aixtream's virtual audio router makes complex input/output routing and audio reuse possible.
Recording
Need to record the call? All possible while the call is live.
Smooth Delay
Maximize the security andsafety of your call by encapsulating it in a Virtual Private Network.
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product specification
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our testimonials
WHAT DO OUR CUSTOMERS SAY?
Udo Botzenhardt, 1. FSV Mainz 05
Until now we had to use various pieces
of equipment for our requirements.
Thanks to Ferncast and the fernReport
we can now integrate experts and our
fans into the program using a small and
simple-to-use device and so enhance our
live reporting on 05ER.fm
