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Barix Intros New SIP Audio Endpoint

Barix's most feature-rich SIP interface solution to date, the new SIP Audio Endpoint is designed to enable the cost-effective extension of SIP-based VoIP telephone systems.
(Image credit: Barix)

The What: Barix's most feature-rich SIP interface solution to date, the new SIP Audio Endpoint is designed to enable the cost-effective extension of SIP-based VoIP (Voice over IP) telephone systems with functionality including intercom, paging, music-on-hold and other audio applications.

The What Else: Barix has a proven history of providing robust yet inexpensive interface hardware that bridges analog audio inputs and outputs with SIP phone systems, turning equipment ranging from amplifiers and doorbells to paging stations and meeting room systems into SIP-enabled client endpoints. The SIP Audio Endpoint builds on this tradition by offering new codec support and expanded functionality on a modern hardware foundation. The first model in the new series, the mono, bi-directional MA400 SIP Audio Endpoint, features a microphone or line-level analog audio input and 5-watt (8-ohm) or line-level analog output.

The Bottom Line: The SIP Audio Endpoint supports a broad range of audio codecs including Opus, G.711, G.722, and GSM. Contact closures allow triggering from physical interfaces such as call buttons, while features such as DTMF tone dialing support maximize integration possibilities.

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