24/7 Audio Call Software

The ideal audio call solution for SIP answering machines, remote communication 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.

Enjoy Maximum Flexibility in 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.

Special Features

• 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.