Audio around your telephone systems
SIP/ VoIP based audio solutions for message on hold, paging and intercom.
The Instreamer can stream audio over IP networks into phone systems smooth and easy. The software converts any audio signal into G.711, PCM or high-quality MP3 streams and can send the stream to up to 32 different destinations across the LAN or over the internet. Supporting peer-to-peer connections with other Barix devices, the software can also be used for integration into wide audio infrastructures with Shoutcast or Icecast servers.
- MP3 Encoding (low Latency)
- G.711, G.722, PCM linear Encoding
- Shoutcast/Icecast Source capability
- SNMP Trap generation
- IP Streaming via TCP, UDP, RTP, Multicast
- Line Level input (Stereo)
- RS-232 Serial Port
With the SIP Client application it is easy to expan an existing VOIP phone system to make announcements. It can dial a single or group of loudspeakers from any connected SIP phone. SIP Client works with most VoIP solutions including Cisco and Asterisk. The client device receives a phone number in addition to its IP address, and can communicate either on a peer-to-peer connection or over a proxy-based connection. Ringtone and busy tone guarantees that the call remains confidential and the loudspeaker and its audience really receives the information. Featuring background music, audio rebroadcasting, priority audio port and special DTMF commands to quickly manage the device, the SIP Client expands the functionality of your SIP system. Even controlling the relays (e.g. to open door) of a Barix device via SIP phone is no problem with this solution.
A central management software portal with corresponding audio playout devices. Users with administrative access can create and schedule playlists featuring music, ads and promos from the head office, and additionally configure and update player devices across the network. At the playout locations, Barix devices deliver high-quality audio content to speaker systems, with targeted advertisements seamlessly fading in and out of music streams and playlists.
- Centralized management software
- Audio players with high-quality music reproduction
- Seamless switching between live streams and stored playlists
- Dynamic ad insertions with seamless fade-in/out
- Alarm generation, network reporting for monitoring and troubleshooting and more
Barix’new MA400 SIP Opus Codec enables remote contribution links with SIP and the high-quality, open audio format Opus in a cost-effective solution. The advantages of the royality-free Opus codec are its quality, efficiency and low latency. Combined with the SIP functionality to establish seamless links across VoIP and other IP based communications systems, it is ideal for remote contribution applications. The SIP functionality in the Barix unit allows to dial another device or auto replies to a call and automatically negotiate a transmission link for Opus based audio streaming. Particularly interesting is the MA400 SIP Opus Codec for remote contribution back to the studio such as from sporting events.
SIP Opus Codec
- Bi-directional mono Opus codec
- Microphone / line level input
- Audio level supervision with SNMP trap generation
- Support of multiple coding standards
- Line level output (balanced)
- Amplified output 5W (8 Ohm)
- Power over Ethernet (PoE)
Implement a zone capable paging solution without the need for a SIP client for each paging point. Page routing is done in the Barix paging gateway and server. The big advantage of this solution: there is no complicated SIP server configuration required and with only one SIP client port required per simultaneous paging link. The solution overcomes the problems of prohibitive implementation cost for SIP solutions where the client needs to pay individual client licenses or each connected zone.
SIP Zone paging
- Only one SIP server client
- Up to 80 zones addressable
- Page to all or page to selected zones
- 7 zones per call selectable by phone keypad
- 4 simultaneous paging sources
- Chime support
- Background music channels
- Emergency notification port on receivers
SIP Client Application is the right software to integrate Barix devices into an existing VOIP phone system. SIP Client works with most SIP based VoIP solutions including Cisco Unified Communications Manager or Asterisk. Using the Session Initiation Protocol (SIP), the client device receives a phone number in addition to its IP address, and can communicate either on a peer-to-peer connection or over a proxy-based connection (PBX). Ringtone and busy tone guarantees that the call remains confidential and the recipient really receives the information. Featuring background music, audio rebroadcasting and special DTMF commands to quickly manage the device, the SIP Client is easy to integrate into an existing telephone system. Four different relay ports can be configured to open doors, windows, or to activate machines while simultaneously speaking.