Dante-enabled Product Catalog
Speaking the language of the interpreter is what the DICENTIS Interpreter desk is designed to do. By observing and interviewing expert interpreters and technicians worldwide, the desk has been optimized for the real-life needs of interpreters.
Speaking the language of the interpreter is what the DICENTIS Interpreter desk is designed to do. By observing and interviewing expert interpreters and technicians worldwide, the desk has been optimized for the real-life needs of interpreters.
The DICENTIS Interpreter desk is part of the DICENTIS Conference System, which supports up to 100 interpreted languages. The conference system uses internationally recognized encryption standards for all audio and data. It offers the best protection against eavesdropping, unauthorized access and data tampering. High-quality, real-time audio and control is ensured by OMNEO* IP technology, which also allows for easy system integration with third-party solutions. OMNEO also enables the use of existing network infrastructure.
The DICENTIS Interpreter desk is fully compatible with the DICENTIS platform and complies with the simultaneous interpreting equipment standard ISO 20109. It enables the creation of passive overflow rooms adjacent to the main meeting room in case there are more participants than can be accommodated at the main meeting location.
Using the DICENTIS Meeting application, all languages and interpreter desks can be remotely configured and managed.
The IP116-D Announcement Control System (aka ACS) is the heart of AtlasIED's next generation GLOBALCOM®.IP enterprise wide communications management system.
The IP116-D Announcement Control System (aka ACS) is the heart of AtlasIED's next generation GLOBALCOM®.IP enterprise wide communications management system. It manages the flow of announcements and messages between network endpoints such as digital communication stations, zone controllers, network power amplifiers, IP-based loudspeakers and other types of IP endpoints. Additionally it serves as a gateway for external I/O devices, VoIP telephone systems and analog audio to be incorporated into the ecosystem. Each model is 'plug and play' ready with GCK application software installed and ready to be easily configured via an intuitive browser-based interface.The IP116-D model has the ability to interface with external systems via eight (8) built-in logic inputs and eight (8) built-in relay outputs as well as two (2) balanced line level audio inputs.The IP1116-D Controller supports VoIP communications with Audio over Ethernet (AoE) functionality using sixteen digital channels of Dante™ digital audio transport over the same network infrastructure simultaneously. Our VoIP is Wide Area Network (WAN) enabled allowing communication to be routed across an entire enterprise or to another LAN location.
The IP108-D Announcement Control System (aka ACS) is the heart of AtlasIED's next generation GLOBALCOM®.IP enterprise wide communications management system.
The IP108-D Announcement Control System (aka ACS) is the heart of AtlasIED's next generation GLOBALCOM®.IP enterprise wide communications management system. It manages the flow of announcements and messages between network endpoints such as digital communication stations, zone controllers, network power amplifiers, IP-based loudspeakers and other types of IP endpoints. Additionally it serves as a gateway for external I/O devices, VoIP telephone systems and analog audio to be incorporated into the ecosystem. Each model is 'plug and play' ready with GCK application software installed and ready to be easily configured via an intuitive browser-based interface.The IP108-D model has the ability to interface with external systems via eight (8) built-in logic inputs and eight (8) built-in relay outputs as well as two (2) balanced line level audio inputs.The IP108-D Controller supports VoIP communications with Audio over Ethernet (AoE) functionality using eight digital channels of Dante™ digital audio transport over the same network infrastructure simultaneously. Our VoIP is Wide Area Network (WAN) enabled allowing communication to be routed across an entire enterprise or to another LAN location.
The 5432DZM Digital Zone Manager provides processing and management of up to 32 paging zones assigned from anAtlasIED 5400 Announcement Control System.
The 5432DZM Digital Zone Manager provides processing and management of up to 32 paging zones assigned from anAtlasIED 5400 Announcement Control System. It provides line level audio outputs to a paired four (4) channel amplifier unit. Digital Signal Processing (DSP) is provided on each of the four (4) output channels. This includes: gain control for the paging input, page routing, automatic ducking of background music, equalization (high and low pass filters, up to 7 parametric bands per amplifier channel, and up to 40 milliseconds of delay). The paired amplifier outputs comes back into the 5432DZM and then is distributed to up to eight (8) loudspeaker line outputs per amplifier channel. Each of the 32 loudspeaker lines is supervised.The 5432DZM provides automatic backup amplifier switching by switching groups of eight (8) loudspeaker loads from the primary power amplifier to a backup when a failure is detected. It also contains integrated supervision that monitors each signal path and reports any failures to the 5400ACS.
The 5404DZM Digital Zone Manager provides processing and management of four (4) paging zones assigned from anAtlasIED 5400 Announcement Control System.
The 5404DZM Digital Zone Manager provides processing and management of four (4) paging zones assigned from anAtlasIED 5400 Announcement Control System. It also provides line level audio outputs to a paired four-channel amplifier unit. Digital Signal Processing (DSP) is provided on each of the four (4) output channels. This includes: input level controls, paging routing, automatic ducking of background music, equalization (high pass filters, low pass filters, up to 40 milliseconds of delay, plus up to 7 parametric bands per amplifier channel).AtlasIED’s patented ambient analysis and control provides automatic level adjustment with two (2) Sensor Input Channels per amp channel.The 5404DZM provides backup amplifier switching by switching up to four (4) loudspeaker loads from a primary power amplifier to a backup when a failure is detected. It also contains integrated supervision that monitors each signal path and reports any failures to the 5400ACS.
Powerful and flexible, the HT-9500M fully digital conference system allows both medium and large-scale meetings to be managed with complete control and reliability.
Powerful and flexible, the HT-9500M fully digital conference system allows both medium and large-scale meetings to be managed with complete control and reliability. The conference system combines PC control, software package, and digital processing unit in one integrated system, whose central controller is designed digitally. A comprehensive powerful options, like, the management of multi-mode microphone, video tracking, video switching, recording, monitoring and etc. are displayed clearly on the menu, so operator could find the required function promptly and correctly. Besides, the conference controlling is done by a management software, which could work well on independent windows PC when Ethernet cables are adopted.
The conference system is composed by a DCN-7300 unit and two kinds of microphones, including one for representatives and one for chairman.
With latest technology, our HTDZ digital conference system is managed to integrate discussion, voting, simultaneous interpreting (SI), registering, video tracking and software management together, so it could be installed on varied conference rooms, from the smallest up to the very largest ones.
The conference system is composed by a DCN-7300 unit and two kinds of microphones, including one for representatives and one for chairman. Both these two kinds of microphones have built-in speaker and directional cardioid microphone with light ring, allowing excellent audio sensitivity and wide frequency response. Meanwhile, since the routing for the system is quite easy, a fast and correct installation could be guaranteed.
As many as 64 microphones could be installed on our conference system. The conference system has four independent lines, each of them allowing an installation of 30 microphones.
When installing HT-08EX, 60 microphones could be installed on the system (six network interfaces). And as many as 30 microphones are allowed to be installed on a single interface.
The conference system is composed by a DCN-6600 unit and two kinds of microphones, including one for representatives and one for chairman.
With latest technology, our HTDZ digital conference system is managed to integrate discussion, voting, simultaneous interpreting (SI), registering, video tracking and software management together, so it could be installed on varied conference rooms, from the smallest up to the very largest ones.
The conference system is composed by a DCN-6600 unit and two kinds of microphones, including one for representatives and one for chairman. Both these two kinds of microphones have built-in speaker and directional cardioid microphone with light ring, allowing excellent audio sensitivity and wide frequency response. Meanwhile, since the routing for the system is quite easy, a fast and correct installation could be guaranteed.
As many as 64 microphones could be installed on our conference system. The conference system has four independent lines, each of them allowing an installation of 30 microphones.
When installing HT-08EX, 60 microphones could be installed on the system (six network interfaces). And as many as 30 microphones are allowed to be installed on a single interface.
DCN-2288R wireless microphone conference system support different meeting modes, like, Limit (1/2/3/4), FIFO (1/2/3/4) and Chairman Only.
DCN-2288R wireless microphone conference system support different meeting modes, like, Limit (1/2/3/4), FIFO (1/2/3/4) and Chairman Only. Single audio receiver allows messages’ receiving from one thousand microphones simultaneously, and when linked with central processor, video tracking for 256 microphones could be realized. Meanwhile, five microphones could be activated at the same time.
Signal’s transmission could be realized with FM (frequency modulation), featuring high fidelity, high signal-to-noise ratio, and low latency. Unique state-of-art condenser microphone cartridge is introduced, which is resistant to whistle, highly sensitive, allowing excellent audio transmission effect.
Wireless transmission is introduced for both audio channel and control channel with no need for extra cable routing. Besides, the system could be installed anywhere as soon as there is a signal coverage, so conference room’s displaying is more convenient.
The system features ultra-low power, so continuous speaking for more than eight hours or more than thirty hours stand-by is supportable. AA battery and original lithium battery provided by our company are all supportable. Meanwhile, coupled charging case is designed with intelligent power management integrated circuit, so the battery could be well protected for prolonged service life and safe us.
Built-in USB is installed on its audio receiver, allowing both the audio recording and playing function. Besides, when used together with a Bluetooth device, the audios inside the device could also be played with our system.
Built-in Dante audio transmission interface is installed.
The system allows an average working distance of 60m, but 80m is also possible.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
The core of TAIDEN infrared wireless conference system – dirATC (Digital Infrared Audio Transmitting and Control Technologies) - is a new technology originated by TAIDEN. It digitally encodes and modulates audio and data signals for transmission via infrared light, achieving both multi-channel audio signals and bidirectional data transmission and control.
The core of TAIDEN infrared wireless conference system – dirATC (Digital Infrared Audio Transmitting and Control Technologies) - is a new technology originated by TAIDEN. It digitally encodes and modulates audio and data signals for transmission via infrared light, achieving both multi-channel audio signals and bidirectional data transmission and control. Digitalization combined with wireless transmission via infrared technology is an ideal solution for a wireless conference system.
TAIDEN digital infrared wireless conference system is composed of a main unit, one or more transceivers and conference units.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
HCS-5100Plus Series New Generation Digital Infrared Language Distribution System adopts TAIDEN new generation digital infrared processing chip, it conforms to the international standard for digital infrared. This system supports at most 40 distributing channels, making it the world's most advanced IR language distribution system.
DCN-3388R wireless microphone conference system support different meeting modes, like, Limit (1/2/3/4), FIFO (1/2/3/4) and Chairman Only.
DCN-3388R wireless microphone conference system support different meeting modes, like, Limit (1/2/3/4), FIFO (1/2/3/4) and Chairman Only. Single audio receiver allows messages’ receiving from one thousand microphones simultaneously, and when linked with central processor, video tracking for 256 microphones could be realized. Meanwhile, five microphones could be activated at the same time.
Signal’s transmission could be realized with FM (frequency modulation), featuring high fidelity, high signal-to-noise ratio, and low latency. Unique state-of-art condenser microphone cartridge is introduced, which is resistant to whistle, highly sensitive, allowing excellent audio transmission effect.
Wireless transmission is introduced for both audio channel and control channel with no need for extra cable routing. Besides, the system could be installed anywhere as soon as there is a signal coverage, so conference room’s displaying is more convenient.
The system features ultra-low power, so continuous speaking for more than eight hours or more than thirty hours stand-by is supportable. Battery of the system could be charged with DC power adapter, which could be installed directly on microphones, or charging case. Charging case is designed with intelligent power management integrated circuit, so the battery could be well protected for prolonged service life and safe us.
Built-in USB is installed on its audio receiver, allowing both the audio recording and playing function. Besides, when used together with a Bluetooth device, the audios inside the device could also be played with our system.
Built-in Dante audio transmission interface is installed.
The system allows an average working distance of 60m, but 80m is also possible.
The PGM1 is a paging station microphone that can be used with the MTX5-D or MRX7-D.
The PGM1 is a paging station microphone that can be used with the
MTX5-D or
MRX7-D.
- Dante network support
- 8 buttons can be assigned to any functions *Output destinations and input sources are specified via the MTX-MRX Editor
- Up to 4 units can be connected per system
- PoE powered
- Dimensions: 132(W) × 56(H) × 154(D) mm (excluding mic)
- Weight: 1.2 kg (including mic), 1.0 kg (excluding mic)
- Dante Ch: 0x1
- 44.1kHz-48kHz
The PGX1 is a paging station extension that can be used with the PGM1.
The PGX1 is a paging station extension that can be used with the PGM1.
- Function expansion for the PGM1
- 8 buttons can be assigned to any functions *Output destinations and input sources are specified via the MTX-MRX Editor
- Up to 2 units can be connected to each PGM1
- Dimensions: 132(W) × 56(H) × 154(D) mm
- Weight: 0.9 kg (excluding connecting bracket)
- Dante Ch: 1x0
- 44.1kHz-48kHz
The Yamaha MCP1 is an advanced wall-mount control panel for controlling MTX and MRX systems.
The Yamaha MCP1 is an advanced wall-mount control panel for controlling MTX and MRX systems. Its dot-matrix display and universal character support facilitates dynamic labelling in many languages including Japanese, Chinese and Arabic. Up to 36 parameters can be assigned to either the main Home page or any of the six sub pages. Any of these pages can be secured from unauthorized access by using a programmable PIN code. Up to 16 units of MCP1 can be added to an MTX/MRX system.
- Multi-language displays supported
- Four user levels for access management
- Up to 16 units can be connected per system
- PoE powered
- Back can adapter supplied
The TesiraFORTÉ DAN VT4 is a digital audio server with 4 analog inputs and 4 analog outputs, and includes Acoustic Echo Cancellation (AEC) technology on all 4 inputs.
The TesiraFORTÉ DAN VT4 is a digital audio server with 4 analog inputs and 4 analog outputs, and includes Acoustic Echo Cancellation (AEC) technology on all 4 inputs. Each model also includes up to 8 channels of configurable USB audio, a 2-channel VoIP interface, and a standard FXO telephone interface. TesiraFORTÉ DAN VT4 includes Dante digital audio networking.
- 32 x 32 channels of Dante (DAN model only)
- 12 mic/line level inputs with AEC, 8 mic/line level outputs
- Gigabit Ethernet port › RS-232 serial port
- 4-pin GPIO
- 2-line OLED display with capacitive-touch navigation
- Rack mountable (1RU)
- System configuration and control via Ethernet
- Internal universal power supply
- SIP VoIP interface via RJ-45 connector
- Standard FXO telephone interface via RJ-11 connector
- Signal processing via intuitive software allows configuration and control for signal routing, mixing, equalization, filtering, and delay
- CE marked, UL listed, and RoHS compliant
- Covered by Biamp Systems’ five-year warranty
- Dante Ch: 32x32
- AES67 Enabled
The TesiraFORTÉ VT is a digital audio server with 12 analog inputs and 8 analog outputs, and includes Acoustic Echo Cancellation (AEC) technology on all 12 inputs.
The TesiraFORTÉ VT is a digital audio server with 12 analog inputs and 8 analog outputs, and includes Acoustic Echo Cancellation (AEC) technology on all 12 inputs. It includes up to 8 channels of configurable USB audio, a 2-channel VoIP interface, and a standard FXO telephone interface. TesiraFORTÉ DAN VT adds Dante digital audio networking.
- 32 x 32 channels of Dante (DAN model only)
- 12 mic/line level inputs with AEC, 8 mic/line level outputs
- Gigabit Ethernet port
- RS-232 serial port
- 4-pin GPIO
- 2-line OLED display with capacitive-touch navigation
- Rack mountable (1RU)
- System configuration and control via Ethernet
- Internal universal power supply
- SIP VoIP interface via RJ-45 connector
- Standard FXO telephone interface via RJ-11 connector
- Signal processing via intuitive software allows configuration and control for signal routing, mixing, equalization, filtering, and delay
- CE marked, UL listed, and RoHS compliant
- Covered by Biamp Systems’ five-year warranty
Standalone commentary unit (CU), or AoIP connected with 8 channels via Dante protocol. Scalable architecture: simple routing to DanteTM IP devices; integrated in IP intercom system, or connected to IP commentary system matrix.
Standalone commentary unit (CU), or AoIP connected with 8 channels via DanteTM protocol. Scalable architecture: simple routing to DanteTM IP devices; integrated in IP intercom system, or connected to IP commentary system matrix.
Standalone mono or stereo sound mixer with mixing, routing, tone and dynamics control. 3 commentator inputs and a dual-mono or stereo line level input. Listening of 8 remote and 2 local sources.
- Operates as an intercom panel at the same time as a Commentary Unit.
- Configurable as interpreter desk up to three languages.
- 3 1Gigabit IP ports per unit for redundancy, daisy chain and auxiliary data or video transport.
- Dual power supply: 48VDC via PoE or external local power supply.
- Software Configuration and remote control.
- Rugged and ergonomic mechanics, suitable for indoors and outdoors locations.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio through:
- Seamless microphone integration
- Comprehensive connectivity from one DSP model
- Simplified software configuration
The EX-8ML is an under-table endopoint with 8 Mic Ins, 48V phantom, Logic In/Out and Mic LED power.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio through:
- Seamless microphone integration
- Comprehensive connectivity from one DSP model
- Simplified software configuration
The EX-UH is an under-table endopoint with USB, RJ-9 and 3.5 mm TRRS analog I/O.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio through:
- Seamless microphone integration
- Comprehensive connectivity from one DSP model
- Simplified software configuration
The EX-4ML is an under-table endopoint with 4 Mic Ins, 48V phantom, Logic In/Out and Mic LED power.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio.
The ControlSpace® EX conferencing solution is a Dante™-based processor and under-table endpoint system designed to streamline the implementation of conference room audio through:
- Seamless microphone integration
- Comprehensive connectivity from one DSP model
- Simplified software configuration
By passing all audio signals through a powerful DSP that’s built to make sure every word is heard, TesiraFORTÉ CI is the ideal model to help you get the most from soft codec conferencing.
Each TesiraFORTÉ® model is designed and optimized to address a different business challenge, with the ability to accommodate future AV technology advancements. The Dante-equipped (DAN) models allow you to interface with other audio devices that communicate over Dante.
By passing all audio signals through a powerful DSP that’s built to make sure every word is heard, TesiraFORTÉ CI is the ideal model to help you get the most from soft codec conferencing.
With the USB port enabling direct integration with soft codec technologies, and AEC technology ensuring superior conferencing audio, all TesiraFORTÉ CI models are built for truly collaborative communications.
Specialized Features
At its best in conference rooms or council chambers, the sweet spot for the AI model is installations requiring high-quality audio solutions, such as those with voice lift and mix-minus.
Each TesiraFORTÉ® model is designed and optimized to address a different business challenge, with the ability to accommodate future AV technology advancements. The Dante-equipped (DAN) models allow you to interface with other audio devices that communicate over Dante.
At its best in conference rooms or council chambers, the sweet spot for the AI model is installations requiring high-quality audio solutions, such as those with voice lift and mix-minus. The TesiraFORTÉ AI and DAN AI are best suited for those environments where a standalone DSP is needed for localized audio per room.
For customers who conference over analog telephone lines, the TesiraFORTÉ TI models are a cost-effective solution that provide quality audio in conference rooms, without requiring infrastructure updates.
Each TesiraFORTÉ® model is designed and optimized to address a different business challenge, with the ability to accommodate future AV technology advancements. The Dante-equipped (DAN) models allow you to interface with other audio devices that communicate over Dante.
For customers who conference over analog telephone lines, the TesiraFORTÉ TI models are a cost-effective solution that provide quality audio in conference rooms, without requiring infrastructure updates.
Whether you plan to link multiple rooms, add them to an existing Tesira AVB network, or keep the units separate, the TI models facilitate broad communication over standard telephone lines.
Specialized Features
- AEC on all inputs
- Standard telephone interface via an RJ-11 connector
