Dante-enabled Product Catalog

The Model 207A eSports Console offers a unique combination of analog and digital audio resources specifically intended to support eSports-related live event, entertainment, and streaming broadcast applications. The unit is housed in a compact, steel enclosure that's intended for table-top use. Its compact size also makes it ideal for use in space-constrained locations. Color-configurable LED lights cast a distinctive "underglow" from the bottom of the enclosure. The Model 207A supports Dante® audio-over-Ethernet digital media technology with AES67 compatibility for integration into contemporary applications. The unit is extremely simple to deploy, is "pro" quality throughout, and provides an intuitive user experience. The Model 207A's audio quality is excellent, with low distortion, low noise, and ample headroom. Careful circuit design and rugged components ensure long, reliable operation. Key Features

• Dante audio-over-Ethernet technology
• Supports headsets with 5-pin XLR and 3.5 mm TRRS CTIA connectors
• Independent stereo earbud and headset headphone outputs
• Integrated masking noise source
• Unbalanced stereo line-input for compatibility with computer audio
• Main, talkback, and two computer audio output channels
• Compact unit supports one on-air location
• Excellent audio performance
• Uses STcontroller for configuration
• Power-over-Ethernet (PoE) powered

Dante audio interface, mixer, and MIDI control in one compact device, the Switch Back PLUS condenses functions into a single, unique tool. Connect via Cat6 cable to a PoE+ network switch, plug in your headphones/speakers, route channels, and start making music. Mixer: Control volume, panning, and stereo width of 64 Dante audio channels, with channel names on crisp OLED displays. Local inputs mix in at zero-latency. The mix is fed through a master volume control and output to a balanced stereo pair and two headphone jacks. Interface: Local inputs include XLR microphone input with switchable phantom power, high-impedance instrument input with adjustable gain, and two balanced line-level inputs. Route these and the main stereo mix to other devices on the Dante network. Unmuting four default-muted Dante channels provides intercom. MIDI control: MIDI shares the same network connection as Dante to send/receive CCs. Knobs can control software and external hardware.

Expand your Dante networked audio rig with I/O and monitor control. Add personal monitor control with headphone amplifier and line I/O to your Dante® audio network Easily connect synths, amp emulators, external analog gear, and playback devices with Apollo e2m's built-in stereo line inputs Use Power over Ethernet (PoE) and built in mic-stand mount to place Apollo e2m anywhere you need monitoring — on stage or in the studio Pair with Apollo e1x remote-controllable Unison™ preamps and Apollo x16D for a complete I/O solution  

Put Unison preamps on stage or in your studio with Dante.

• Expand any Dante® network with remote-controlled mic/line preamps using Power over Ethernet (PoE)
• Combine with Apollo x16D interface to get the authentic Unison™ mic preamp sounds of API, Avalon, Manley, Neve, SSL, and more
• Link multiple Apollo e1x units with included coupling bracket and integrated mic-stand mount
• Pair with Apollo e2m stereo headphone amplifier/line interface and Apollo x16D for a complete I/O solution

The Spectera Base Station is a 1RU hardware device that provides up to 64 channels (32 in/32 out) and utilizes up to two wideband RF carriers to accommodate all of the unit’s wireless transmission. It features redundant Dante and optional redundant Madi, assuring seamless integration into your audio network. Additional antennas can be added for redundancy, extended zone coverage, or additional spectrum capacity. The Spectera Base Station supports sample rates of 48 kHz and 96 kHz and features individual sample rate converters for all audio interfaces and wordclock I/O. Base Station must be activated with license per region.

• 19’’ 1 RU Base Station with redundant power supplies
• Provides up to 64 channels (32 in / 32 out) and allows for up to two independent wideband RF carriers in UHF or 1G4 (6 or 8 MHz) – country specific per region license
• 4 ethernet antenna ports for redundancy, extended zone coverage, and additional spectrum usage
• Redundant Dante® audio interface
• Two optional slots for redundant MADI (OM or BNC)
• Sample rates of 48 kHz and 96 kHz
• Individual sample rate converters for all audio interfaces
• Wordclock I/O BNC
• Synchronized clock for all audio over RF allows 3D recordings and playback
• Adjustable RF power from 10 – 100 mW (country specific)
• RJ45 network interface for full remote control with secured interface
• 11 Audio Link modes with selectable audio codecs (OPUS, SeDAC) or PCM
• Secure transmission of confidential content using AES 256 encryption
• Future expansion via Cascade port
• Selective headphone output with volume control on front panel
• OLED display, jog wheel navigation, and rugged metal housing
• Base Station activation via LinkDesk: Spectera license defines respective frequency spectrum per region (license sold separately)

Auri™ is the first complete assistive listening system for professional installation utilizing Auracast™ broadcast audio. Auri is an ideal public solution, addressing hearing assistance and personal listening preferences. Auri enables end users to use their Auracast™ compatible device or a venue-provided receiver to enhance their audio experience. Key features include:

• Multi-channel Broadcast
• Network Management
• Open or Secure Audio
• Compatible with Auracast™ enabled devices, such as hearing aids, headphones or earbuds

The Auri TX2N-D is an Auracast™ Broadcast Audio transmitter. It features balanced analog mic/line inputs and 2 channels of Dante, supporting mono, stereo, or dual mono configurations. Used with Auri RX1 Receivers it creates a fully compliant next generation assistive listening solution.

The WL64-4 rack mount Audio over IP (AoIP) audio monitoring unit operates in conjunction with the Nixer Remote iPad® App. The WL64-4 provides a “base unit” for the monitoring and mixing of AoIP sources from a Dante or Ravenna network. Each unit is configured at time of purchase to be Dante or Ravenna. The Nixer Remote App allows audio to be wirelessly streamed to an iPad® for local monitoring, freeing the Operator from a dedicated hardware front panel display and equipment rack. As standard, each WL64-4 can have four iPads connected wirelessly and each Operator can select and listen to sources independently as well as creating their own individual mixes. Each iPad is able to stream audio wirelessly to the internal speakers, headphone output or via Bluetooth. The audio can be selected in a simple “Listen” or “Mix” mode.

THE PERFECT LINK The MT 48 brings Neumann quality to the digital workflow. It is the first audio interface in its class that is able to capture the full dynamic range of Neumann microphones and is optimally matched to the KH-Line of studio monitors and Neumann headphones. The result is an ideal signal chain without any bottlenecks: uncompromising sound quality from the microphone diaphragm to your ear. SWISS PRECISION INSIDE The MT 48 was developed in close cooperation with the Swiss manufacturer Merging Technologies, whose high-precision AD converters offer a staggering dynamic range of 136 dB. This gives the MT 48 four times the resolution of comparable audio interfaces. The MT 48 comes fully equipped for the reference-class personal studio: two extremely low-noise mic/line preamps with up to 78 dB gain, two line/instrument inputs, four monitor/line outputs and two superb headphone outputs, each with state-of-the-art AD/DA converters. The MT 48’s connectivity may be expanded via ADAT / S/PDIF and AES67; a MIDI interface is also integrated. Four separate monitor mixes can be controlled directly on the unit via touchscreen. On-board processing includes a fully parametric 4-band EQ and a 3-way dynamics processor (gate/compressor/limiter) per channel as well as a high-quality reverb. So musicians get inspiring sound for top performances. And there is still plenty of DSP power for future developments.

This is the first of a new generation of the industry standard, DARK Dante/AES67 interfaces. The DARK DAWN range of interfaces are multichannel analogue and AES input/output devices intended to be installed in a studio setting where cosmetic appearance is important. The DARK DAWN 1616M has 16 analogue inputs, and 16 analogue outputs. The inputs are mic/line/ 48v switchable. Channel gain can be adjusted from the front panel or remotely via GlenController (Dante version) with UDP control via a separate network connection also possible. The front LED panel is machined from a single block of aluminium with no screwheads or connectors visible. It includes 16 individual channel meters, a single hi-res meter for setting individual channel level and a 1/4” headphone socket for the monitoring of all input or output channels. Loss of network or power supply is reported by GPO.

Engineered for commercial AV environments, the Infinium™ CT1 delivers ultra-low latency, high-fidelity audio to unlimited Auracast™ enabled devices—including hearing aids, cochlear implants, headphones, and Infinium™ receivers. Perfect for lecture halls, transportation hubs, theaters, and other large-scale venues, the CT1 is a modular solution featuring a discreet transmitter and a rack-mount-ready controller. Key Features

• Professional AV Controller.
• Seamlessly integrates into AV racks with both analog and Dante® inputs.
• Configure locally via the front panel or remotely through a web interface.
• Compact Transmitter.
• Installs easily with a single CAT cable carrying both audio and power.
• Delivers omnidirectional audio coverage up to 100 meters (328 ft).

The AVN-CU1 adds to the Sonifex range of Dante commentary units with a simple to use, single microphone/headphone combination with a full feature set and a low price point.

The HDSPe AoX PCIe x4 audio interface cards combine dual network ports with headphone output, AES3, MIDI, and internal expansion headers for up to four MADI ports. Whether used in a Thunderbolt™ expansion chassis or built into DAWs or audio processors, these cards send and receive up to 512 channels of audio and provide numerous mixing and routing capabilities.

The vast number of channels can be managed by the user in two steps. Using a driver dialog or web interface, the user first adjusts the network vs. optional MADI channel count, and then pre-selects which inputs reach the 256 channel TotalMix engine. The remaining 256 channels are passed straight through. All 512 channels are visible to the DAW for recording. TotalMix offers its basic functionality without effects, but with all the great benefits such as presets, layouts, loopback, sub-mixes, and many more.

Lake People Phone-Amp G103D4 is a headphone amplifier intended for DANTE networks. The G103D4 model comes with a DANTE 4/4 input and two independent stereo amplifiers. 24 Bit / 192 kHz D/A converters with a Signal-to-noise ratio >105 dB and THD+N as low as 0,002 % are driving the powerful headphone amplifiers. These are best suited to drive any headphones from low to high impedance and with low or high sensitivity. To adapt the amp to demanding headphones, to further reduce the noise level or to maximize the travel of the volume control, the gain is internally adaptable by Dip-Switches to optimize the amp; The DANTE digital audio signals are input via a standard RJ45 connector. Lake People G103D4 will accept 4 audio channels. The unpacked audio signals are input to two high quality D/A converters. The recovered analogue signals are buffered, attenuated in the analogue domain and then sent to powerful headphone amplifiers. If required, additional gain of +6 dB or +12 dB can be set by Dip-Switches inside the amp. Each amp section offers a 1/4” and 3.5 mm phone jack which may be used simultaneously. Power supply for the amp is accomplished either by PoE (Power-over-Ethernet) or by a USB wall plug adaptor (not supplied) with USB-B or USB-C connector. The case, front and back panel of G103D4 is made of rugged black anodized aluminum, the lettering is laser engraved. The volume knobs are also made of solid aluminum.

AX Center It's the audio interface that dreams are made of, as it unites the distinguished audio quality of pristine Digital Audio Denmark AD & DA conversion with the renowned transparency and musicality of the DAD mic preamps. It's all about capturing the source truthfully and get it into your DAW exactly as it is. But that's only the beginning of this journey through the various applications covered by AX Center. On the analog side, two XLR / 1/4" Jack combo connectors on the front for connecting microphones and instruments, you can add two 8 ch. mic/line A/D cards for 16 mic inputs and you have the perfect interface for a small studio or location recording. Or add one 8 ch. D/A card for an additional 8 analog outputs and you have the perfect interface for mixing in 7.1.4 ATMOS format. With one 8 ch. line A/D card and one 8 ch. D/A card, you have the perfect mastering converter. And all along you have two headphone connections where all sources in the system can be easily monitored. Included on the rear panel are a host of connectivity, covering Thunderbolt, Dante AoIP, MADI, ADAT, S/PDIF and Word Clock. In total - and fully expanded - you can max out AX Center with a staggering 984 input and output channels.

The AVN-AH8 multi-channel headphone amplifier is a 1U rack-mount which distributes 2 sets of stereo audio to up to 8 different sets of headphones, fed from the Dante AoIP network. A typical application might be to provide common headphone feeds for guests in a radio studio, coming directly from the AoIP network.

Each output has front panel unbalanced 1/4” & 3.5mm jack sockets and a rotary level control. The front panel potentiometers adjust the headphone volumes from mute (fully anticlockwise) to 0dB of gain when fully clockwise. This is useful if the Dante stream level is low or high and requires adjusting.

There are 8 parallel 1/4” jacks on the rear panel, for ease of wiring if the unit is installed in an enclosed rack.

Each of the 8 professional headphone outputs can each select between 2 stereo Dante sources which are routed via Dante controller. The level of each stereo Dante stream can be adjusted ±12dB using one of the two front panel rotary controls. The unit supports AES67 operation and is Dante Domain Manager compliant.

The AVN-AH8 front panel contains a power (PoE) LED, a DC LED, a network link status LED, and a clock LED. A recessed reset switch is also provided.

The unit is powered via Power over Ethernet (PoE) or a 12VDC input via a locking 2.5mm DC Input, 2A minimum rating.

32-XLR Mic Inputs 20-XLR Auxiliary Outputs XLR- Main Out Headphones Out Digital Inserts 17 Motorized Multi-Use Faders OLED Name Strips External Plug-In Processing 4 Multi-Effects Engines 10" Touchscreen Rec / Play to USB Key Sends on Faders 16 DCA Groups Automix

The Model 362A Listen-Only Beltpack is a self-contained 2-channel user device that supports headphone monitoring of two audio channels associated with the Dante® audio-over-Ethernet media networking technology. Each Model 362A is intended to be used by one user, providing them with the form factor, reliability, and features needed to be effective in a variety of applications.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

A 1U 19inch rack mounted AoIP monitoring and mixing unit.

4 models are available the RL64, RL128, RL192 and RL256 and each unit can be configured at time of purchase to be Dante or Ravenna

RL Series units allows two modes of operation a mixing mode which can mix up to 256 channels at 48kHz (128 channels at 96kHz) plus microphone input (via headphone connector) and AES input to create a 259 into 2 mixer. Alternatively, it can be set into a simple Listen mode where touching a channel instantly routes to the LR output and cancels any previous selection.

The outputs of the LR bus are presented to the user via the inbuilt 4mm x 9mm full range speakers or via the 3.5mm headphone jack. The speakers are automatically muted when a headphone jack is inserted. Simultaneous outputs are also available in the form of the stereo balanced line outputs and AES output accessed via XLR connectors on the rear of the RL Series Unit. The mix output is also routed to the AoIP output.

The RL Series units are operated and controlled via the large letterbox 6.6” diagonal (168mm) 1440 x 240 24-bit colour LCD and capacitive touch panel.

A simple and elegant menu system has been designed to be intuitive to navigate allowing the user to access all the features of the RL Series unit very quickly.

The high-resolution screen is easy to read presenting comprehensive metering, routing and status information to the user.

Firmware updates of the RL Series unit are achieved via a dedicated USB micro AB connector on the front panel. There is also a dedicated 3.5mm jack socket for the headphone output

Stacks Image 30

On the rear of the RL Series unit there are 4x Primary and Secondary ports for up to 4x AoIP inputs. 2x balanced line outputs on 3 pin male XLRs and AES output also on 3 pin male XLR. These outputs can be configured to follow the main volume settings of the speakers/headphones or they can also be set to follow the main Mute control.

Additionally, there is an AES input via 3 pin female XLR connector

The RL Series unit can be powered via either of the two 2.5mm DC input connectors requiring 24V @ 2A. These inputs also have a screw lock to prevent accidental unplugging. Using both inputs provides redundancy.

Quantum2 XL is a complete Commentary Unit including 5 mic/line inputs and 4 headphone outputs. Additional sources to its embedded mixers are USB Audio, AES/EBU or AoIP/AES67 (DANTE or Ravenna optional) inputs.

Quantum2 XL is an all-in-one box solution for remote commentaries linking two bidirectional audio streams with four independent voices, sounds or mixed audios to the studio or Host Broadcast Center.

Perfect sounds are encoded using a choice of advanced and low delay compression algorithms for sport and music.

Highly reliable transmissions are achieved using Diversity and Bonded streams over a group of network interfaces (2x Ethernet, 2x LTE and Wi-Fi).

Additionally, a video downlink stream can be decoded internally and shown on a general purpose monitor, enriching the information supplied to the talents.

A video video uplink is also possible for the transmission of images from the commentators to the studio.

It also allows to insert audio into the received video and to forward a single synchronised video and audio stream to the studio.

Quantum2 W is a complete Commentary Unit including 3 mic/line inputs and 3 headphone outputs. Additional sources to its embedded mixers are USB Audio, AES/EBU or AoIP/AES67 (DANTE or Ravenna optional) inputs.

Quantum2 W is an all-in-one box solution for remote commentaries linking two bidirectional audio streams with four independent voices, sounds or mixed audios to the studio or Host Broadcast Center.

Perfect sounds are encoded using a choice of advanced and low delay compression algorithms for sport and music.

Highly reliable transmissions are achieved using Diversity and Bonded streams over a group of network interfaces (2x Ethernet, 2x LTE and Wi-Fi).

Additionally, a video downlink stream can be decoded internally and shown on a general purpose monitor, enriching the information supplied to the talents.

A video video uplink is also possible for the transmission of images from the commentators to the studio.

It also allows to insert audio into the received video and to forward a single synchronised video and audio stream to the studio.

1. Exquisite and elegant design, ergonomic, very modern. 2. Original D-net IOT digital conference IoT system technology, in line with IEC60914 international standard 3. With priority button function, you can turn off the representative unit that is speaking. 4. The microphone is equipped with an ultra-quiet electronic touch switch, which can avoid the mechanical sound amplification of the switch, and can control the operation of the microphone. The microphone has a bright luminous halo to indicate the working status. 5. The unique audio processing circuit improves the clarity of the entire system. 6. Fixed square array microphone pole with variable directivity. 7. The chairman unit can use the priority button (Priority) and the approval button (Permit) to control the speaking sequence and control the meeting atmosphere. 8. The self-developed technology of D-net IOT digital conference IOT system realizes five adjustable modes of directivity of the pickup microphone: omnidirectional, cardioid, super-cardioid, sharp-hearted, Figure- 8-pointing type; 9. Two-color LED indicator light, green when speaking; 10. The unit comes with a 2m Category 6 RJ45 network cable; 11. Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 12.1 Ø 3.5 mm stereo headphone jack can be connected to headphones. 13. The unique ID number of the microphone unit can be manually assigned to the conference unit by the conference system.

Features： Adopt all-digital conference technology, in line with IEC60914 international standard; The exquisite and minimal structure design highlights the modern atmosphere and conforms to ergonomics; Brand new dual microphone design, longer pickup distance and wider range; Special electret capacitive single microphone core design, clear and bright sound quality, high sensitivity; Original microphone directivity variable technology, users can adjust the microphone directivity according to different environments, locations, meeting methods, and the needs of the speaker, so as to achieve an optimal sound pickup effect, and support 5 modes: omnidirectional, heart -Type directivity, super-cardioid directivity, sharp-heart-type directivity, figure 8 directivity, etc.; Full digital audio technology, built-in high-performance CPU, faster processing speed, better sound quality; Ring hand-in-hand connection technology, the failure or replacement of a microphone will not affect the work of other extensions of the system, and a connection failure between the microphones will not affect the system work, thus making the system more reliable; Support line "hot swap", microphone access system can be added at any time The chairman unit has the priority function of full control of the conference order, and the connection position is not restricted. The chairman unit can short press the function button in the application mode to allow the delegate unit to turn on the microphone. In the conference mode, the chairman unit can short press the function button to turn off all the microphones of the delegate units. If there is a delegate unit waiting to be turned on, it will be turned on immediately. With the camera tracking host, after using the computer preset, the camera can be automatically tracked; There are 4 speaking modes: normal mode, first-in first-out mode, free mode, application mode; Two-color LED indicator, red when speaking; With the sign-in function, the host computer initiates and checks the sign-in situation; Anti-interference from mobile phone microphone core, stable connection, durable and reliable; Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 1 Ø 3.5mm stereo headphone jack can be connected to headphones; The conference unit has a unique serial number, and the conference system can manually assign an ID to the conference unit.

1. Exquisite and elegant design, ergonomic, very modern. 2. Original D-net IOT digital conference IoT system technology, in line with IEC60914 international standard 3. With priority button function, you can turn off the representative unit that is speaking. 4. The microphone is equipped with an ultra-quiet electronic touch switch, which can avoid the mechanical sound amplification of the switch, and can control the operation of the microphone pickup. The microphone has a bright luminous halo to indicate the working status. 5. The unique audio processing circuit improves the clarity of the entire system. 7. The chairman unit can use the priority button (Priority) and the approval button (Permit) to control the speaking sequence and control the meeting atmosphere. 8. The unit comes with a 2m Category 6 RJ45 network cable; 10. Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software, so that speakers with different volumes can speak more smoothly; 9.1 Ø 3.5 mm stereo headphone jacks can be connected to headphones. 10. The unique ID number of the microphone unit, the conference system can manually assign the conference unit.

Features： 1. Adopting all-digital conference technology, in line with IEC60914 international standards; exquisite and minimalist structure design, highlighting the modern atmosphere, in line with ergonomics; special electret capacitive dual microphone core design, clear sound quality, high sensitivity; 2. The self-developed technology of D-net IOT digital conference IOT system realizes five adjustable directional modes of pickup microphones: omnidirectional, cardioid directional, super-cardioid directional, sharp-heart directional, Figure- 8-pointing type; 3. The exquisite and elegant 3.5-inch touch screen is used for the information interaction between the speaker and the conference manager, and can display the following information at the same time: a. Speaker information and time b. Microphone directivity setting c. Sign in and observe the voting status d, microphone gain adjustment e. Information interaction between the speaker and the conference administrator f. System language setting and unit information display 4. Ring-shaped hand-in-hand connection technology, the failure or replacement of a microphone will not affect the work of other extensions of the system, and a connection failure between the microphones will not affect the system work, thus making the system more reliable; 5. Support line "hot swap", microphone access system can be added at any time 6. With the camera tracking host, after using the computer preset, the camera can be automatically tracked The 7.3.5-inch wall control touch screen is used for conference host management and settings, which is convenient and efficient, and can display the following information at the same time: a. Time-limited mode: The speech time limit can be set to 1-60 minutes, and the speech will be automatically closed when the time is up. After reaching the power-on quantity, when another representative unit wants to speak, it will automatically enter the request-to-speak registration queue. b. First-in-first-out mode: First-in-first-out speaking mode (factory preset position), after reaching the number of power-on, the speaking unit opened first is closed by the subsequent speaking unit. c. Application for speaking mode: the speaking of the representative unit needs to be approved by the chairman unit or the operator. In this mode, the chairman unit always keeps the normally open state. 8. Two-color LED indicator light, green when speaking; 9. The unit comes with a 1.8-meter conference dedicated 8-core cable, and a 1-meter analog backup audio cable (with a T-Nong head) 10. Anti-RF microphone core of mobile phone, stable connection, durable and reliable; 11.1 Ø 3.5 mm stereo headphone jack can be connected to headphones. The conference unit has a unique serial number, and the conference system can manually assign an ID to the conference unit.

Features： • Adopt all-digital conference technology, in line with IEC60914 international standard; • Exquisite and minimalist structure design, highlighting the modern atmosphere, accord with ergonomics; • Brand new dual microphone design, longer pickup distance and wider range; • Special electret capacitive single microphone core design, clear and bright sound quality, high sensitivity; • The exquisite and elegant 3.5-inch touch screen is used for the information interaction between the speaker and the conference manager, and the following information can be displayed at the same time: a. Speaker information and time b. Microphone directivity setting c. Sign in and observe the voting status d, microphone gain adjustment e. Information interaction between the speaker and the conference administrator f. System language setting and unit information display • All-digital audio technology, built-in high-performance CPU, faster processing speed, better sound quality; • Digital and analog dual backup function to ensure that the system is stable and reliable, and every important meeting is uninterrupted; • Ring-shaped hand-in-hand connection technology, a microphone failure or replacement will not affect the work of other extensions of the system, and a connection failure between the microphones will not affect the system work, thus making the system more reliable; • Support line "hot swap", microphone access system can be added at any time • The chairman unit has the priority function of full control of the conference order, and the connection position is not restricted; • The chairman unit can short press the function button in the application mode to allow the representative unit to turn on the microphone; • The chairman unit has two touch buttons, no mechanical button sound, long life, and waterproof function, one is to switch the microphone button, the other is the function button; • With the camera tracking host, after using the computer preset, the camera can be automatically tracked; • There are 4 speaking modes: normal mode, first-in first-out mode, free mode, application mode; • Two-color LED indicator, red when speaking; • The unit comes with a 1.8-meter dedicated 8-core cable for conferences, and a 1-meter analog backup audio cable (with a Deng Nong head); • Anti-cell phone interference microphone core, stable connection, durable and reliable; • 1 Ø 3.5 mm stereo headphone jack for connecting headphones; • The conference unit has a unique serial number, and the conference system can manually assign an ID to the conference unit.

The Model 209 Talent Console offers a unique combination of audio, remote control, and tally-output resources for use in podcast and other specialized audio applications. The unit is housed in a compact enclosure that's intended for mounting under a tabletop, within custom furniture, or as part of an on-air set.

The Model 209 supports Dante® audio-over-Ethernet digital media technology and AES67 for integration into contemporary applications. Dante Domain Manager™ software application is supported.

The unit is simple to deploy, is "pro" quality, and provides an intuitive user experience. The audio quality is excellent, with low distortion, low noise, and high headroom. Careful circuit design and rugged components ensure long, reliable operation.

The Model 209 integrates directly into Dante audio-over-Ethernet environments. A PoE connection, microphone, and a pair of headphones or earpiece create a complete audio "position." Remote control inputs and tally outputs allow the unit to integrate into complex applications.