The new SIP Audio Endpoint is Barix' most flexible and feature-rich SIP interface solution to date. The SIP Audio Endpoint enables integrators to seamlessly and cost-effectively extend SIP-based VoIP (Voice over IP) telephone systems with functionality including intercom, paging, music-on-hold, SIP addressable amplifier or loudspeaker and other audio applications. Example use cases include the ability to make announcements through a particular loudspeaker by dialing it from a SIP phone, or having an analog talk station automatically dial a specific VoIP extension. The SIP Audio Endpoint supports a broad range of audio codecs including Opus, G.711, G.722 and GSM. Contact closures on the device allow triggering from physical interfaces such as call buttons, while features such as DTMF tone dialing support maximize integration possibilities. The device is powered by Barix’s high-performance IPAM 400 and is designed to meet today’s increasing IT security requirements. The mono, bi-directional MA400 SIP Audio Endpoint, features a microphone or line-level analog audio input and five watt (eight ohm) or line-level analog output. The stereo M400 SIP Audio Endpoint encodes or decodes the signal to or from a line-level analog in- or output.