The DCS series call stations utilize the latest network audio technology and support three-layer network transmission. They can serve as standard call stations for routine operational announcements or be used for emergency evacuation broadcasts following sudden emergencies.
Depending on usage habits and the specific location, users can choose between the gooseneck microphone (DCS-CSG) or the handheld microphone (DCS-CSH). All buttons on the call station can be freely configured and programmed to meet the user’s specific needs.
In addition to basic broadcasting functions, this call station supports secondary development of industry-specific broadcasting applications through interactive interfaces with dynamic operational databases across various sectors. For example, it can integrate with systems such as the Airport Operations Database (AODB) and the Integrated Business Exchange Platform (IMB) to enable dynamic automated broadcasting, personalized voice announcements for airlines, and the display of passenger flow status and other information.
The DCS series call stations utilize the latest network audio technology and support three-layer network transmission. They can serve as standard call stations for routine operational announcements or be used for emergency evacuation broadcasts following sudden emergencies.
Depending on usage habits and the specific location, users can choose between the gooseneck microphone (DCS-CSG) or the handheld microphone (DCS-CSH). All buttons on the call station can be freely configured and programmed to meet the user’s specific needs.
In addition to basic broadcasting functions, this call station supports secondary development of industry-specific broadcasting applications through interactive interfaces with dynamic operational databases across various sectors. For example, it can integrate with systems such as the Airport Operations Database (AODB) and the Integrated Business Exchange Platform (IMB) to enable dynamic automated broadcasting, personalized voice announcements for airlines, and the display of passenger flow status and other information.
The UR-D1 represents Relacart's latest fully self-developed digital wireless microphone system, combining powerful performance with user-friendly operation. With a 64MHz operating bandwidth and low intermodulation distortion, the system supports up to 20 simultaneously operating transmitters within an 8MHz bandwidth and accommodates over 60 compatible channels in a single frequency band, ensuring stable wireless transmission even in congested RF environments.
Featuring 24-bit/48kHz digital audio resolution, the system surpasses all analog alternatives in reproducing nuanced sound details. With a total system latency of <3ms and a flat frequency response across 20Hz–20kHz, it accurately captures the full spectrum of vocal characteristics—from deep resonances to high-frequency details—delivering transparent, high-fidelity audio. A dynamic range exceeding 120dB ensures exceptional signal-to-noise performance.
The system incorporates intelligent predictive diversity reception technology, which proactively monitors signal quality and utilizes algorithmic optimization to maintain uninterrupted transmission stability. This makes it ideal for complex multi-channel and large-scale audio applications.
For secure communication needs, AES-256 encryption is available to protect privacy in confidential settings such as classified meetings. This encryption also eliminates RF interference by restricting reception to authorized encrypted signals.
Designed for seamless integration, the WireMicX Wireless management system enables effortless remote maintenance, monitoring, and control for streamlined system operation.
The UR-D2 is Relacart's latest fully self-developed digital architecture wireless microphone system. Combining powerful functionality with operational flexibility, it features a 64MHz operating bandwidth and low intermodulation distortion, enabling 20 simultaneously active transmitters within an 8MHz bandwidth and supporting over 60 compatible channels in a single frequency band. This ensures stable wireless transmission even in crowded RF environments.
With 24-bit/48kHz digital audio quality, the system accurately reproduces sonic details beyond the capabilities of analog systems. A total system latency of <3ms and a flat frequency response across 20Hz–20kHz ensure precise vocal reproduction—from deep resonances to high-frequency nuances—delivering transparent, high-fidelity sound. A dynamic range exceeding 120dB provides exceptional signal-to-noise performance.
The system incorporates intelligent predictive diversity reception technology, which preemptively assesses signal quality and uses algorithmic optimization to maintain stable transmission. It fully meets the complex demands of multi-channel, large-scale audio applications. For secure transmission needs, AES-256 encryption is available—ideal for confidential meetings—ensuring privacy protection. This encryption also eliminates RF interference by allowing only encrypted signals to be received.
Designed for easy installation and integration, the system can be seamlessly expanded via existing Dante networks, scaling from small conference rooms or classrooms to large campus and enterprise installations. Advanced features enable detailed configuration management while simplifying complex channel and audio setups. Additionally, the WireMicX Wireless management system supports remote system maintenance, monitoring, and control for streamlined operation.
The AIMIC-MC10 incorporates advanced acoustic algorithms, including Acoustic Echo Cancellation (AEC), Automatic Noise Suppression (ANS), and Automatic Gain Control (AGC). Its proprietary AI-enhanced noise reduction technology can identify and eliminate over 300 types of environmental noise in real-time. With 128 microphone elements arranged in a circular array and combined with sophisticated beamforming algorithms, it achieves true 360-degree coverage for accurate sound pickup. The standout feature of this product is its built-in local sound reinforcement function, enabling real-time audio amplification without the need for an external audio processor.
1. Adapt Variety Teaching Scenarios
Sound reinforcement and pickup are simultaneous at In-person and online class
2. Patented AI Noise Reduction Algorithm
AI noise suppression, AI reverberation cancellation and AI echo cancellation
3.Flexibly configurable audio pickup zones
Featuring 8 customizable pickup zones (amplification/communication/mute) with AI acoustic recognition, delivering precision sound
AlSpeech MA600D is a professional intelligent matrix microphone system featuring a 64-channel omnidirectional array with up to 6m pickup radius and 24 independently configurable pickup zones. Powered by the ClearSpeakAl algorithm suite for Al-driven noise reduction, dereverberation, auto gain, echo and feedback suppression, it supports desktop, ceiling and wall mounting, integrates Dante and analog audio, and delivers high-fidelity sound for reliable local sound reinforcement and remote conferencing across diverse meetings and event spaces.
The AISpeech MCS06 is an all-in-one ceiling mic-speaker designed for meeting rooms up to 60 m². It integrates a 32-element microphone array with dual 15 W loudspeaker drivers and supports Dante digital cascading. When paired with the AISpeech Dante Audio Adapter DU44, the system offers automatic network self-configuration upon power-up, eliminating the need for complex router setups. Built-in AI algorithms effectively handle double-talk and echo, while four configurable pickup zones—combined with advanced noise reduction an dereverberation technologies—deliver accurate voice pickup and crystal-clear conversations. Its integrated design significantly enhances collaboration efficiency in mid-sized meeting rooms.
The DCA901 opens new possibilities for capturing sound in broadcast and media production. This compact digital broadcast microphone array delivers eight channels of high-fidelity audio through a single network cable. Steerable pickup zones isolate the sounds that matter while minimizing ambient noise. With built-in DSP, fewer cables, and faster setup, it replaces bulky analog systems while supporting experiential, story-driven sound and modern production workflows.
Capture up to eight isolated audio channels from a single broadcast array microphone
Steerable lobes let you virtually adjust pickup zones, reducing mic count and eliminating the need to reposition gear
A single Dante or AES67 connection delivers audio, power, and control, simplifying routing and minimizing failure points
Built-in DSP handles EQ, compression, delay, and automixing, giving you more time to focus on creative sound design
Actively supports REMI workflows and alternate feeds with flexible routing and remote lobe control
Presets streamline deployments and ensure consistent configurations across shows or seasons
Supports seamless 5.1 immersive capture and stereo conversion for both modern and legacy formats
Low profile design installs cleanly in stadiums, studios, or mobile setups and blends into camera-ready environments
With 127 wide-range microphones, the Yealink CM50 provides superior audio coverage within a 5-meter radius*. Its advanced beamforming technology supports customized pickup zones, allowing precise control over audio capture and reducing unwanted background noise.
The CM50 integrates AI-based sound optimization to ensure immersive sound quality. With support for Dante protocol , it ensures easy integration with other Dante-compatible devices.
The ultimate high-tier, high channel count wireless receiver, Shure ANX4 receives up to 16 channels of Shure Axient Digital or 24 channels of Shure ULX-D wireless audio in a single rack space. Delivering maximum flexibility, ANX4 does not ship preconfigured with active wireless channels, allowing you the opportunity to purchase channels in single- and four-channel increments and activate the exact number of channels you need. Licenses can be activated in one ANX4 and later removed and activated in a different ANX4 as needs change. ANX4 channel licenses can be purchased through the same authorized suppliers ANX4 hardware units are sold (or at Shure.com/anx4) and managed via your ShureCloud user account.
All the exceptional features found in award-winning Axient Digital and ULX-D receivers that have made them standards on stages, in facilities, on tour, and on location throughout the world are available in the ANX4**. Whether you operate ANX4 in Axient Digital or ULX-D transmission mode, ANX4 maximizes your budget, rack space, and rack weight in delivering large wireless audio channel counts in a single rack design. What’s more, this 1RU format also enables streamlined antenna configurations for large channel count environments. *ANX4 receiver does not ship preconfigured with active wireless channels. Channel licenses are sold separately and required for operation. **System features are dependent upon activated transmission mode: Axient Digital or ULX-D (ANX4 does not support both systems at one time).
The TeamConnect Ceiling Medium Ceiling Tile (TCC M CT) allows for even easier installation of the TCC M into suspended grid ceilings, thereby reducing installation time and costs for integrators on site. The tile comes in a 60 cm square solution and a 2 ft square solution, primarily for the US market.
The TCC M allows integrators the versatility to offer elevated meeting experiences across a range of flexible and hybrid meeting spaces and learning environments. With its adaptive, intelligent beamforming technology, the TCC M ensures Trusted Sennheiser Audio Quality every time. Seamless integration with single cable mode and numerous design and installation options ensure TCC M transforms any room into a hub of collaboration.
Relacart has recently launched an innovative ceiling microphone array – a solution integrating cutting-edge AI technology and advanced acoustic engineering, designed to deliver exceptional audio quality in any space. Our ceiling microphone is not just a technological innovation, but also the optimal audio solution for users in venues like conference rooms and classrooms. It ensures every word is accurately conveyed and no sound detail is overlooked.
The RXT-4DV is a 4” desktop touch panel with a ReAX™ JavaScript based control system and SmartSpeak™. Ideal for conference rooms, educational facilities, hospitality, digital signage and more. The RXT‑4DV has a beautiful 720 x 720 1:1 aspect ratio touch screen with 160° viewing.
SmartSpeak™ takes microphone technology to another level with a Dante/AES67 beamforming microphone with auto gain, noise cancelation, feedback suppression, and anti-tap. Camera tracking using Aurora's WordGate™ technology is like no other. It uses AI to detect speech intelligibility in over 100 languages along with Sound Pressure Levels (SPL) to trigger the camera presets. This eliminates false triggering caused by clapping, coughing, and other unintended noises.
To further the capability of this all-in-one unit is the integrated VPX 1Gbps 4k60 AV over IP transmitter. On the side is a USB-C and mini-HDMI input connector. On the rear is a USB-C power pass-through connector which can move 100 watts to the side input connector for laptop charging. There is also a USB-C display output connector on the rear for secondary screen usage.
The 2D/3D graphics engine with 4K H.264 streaming decoder provides a powerful solution for complex graphics and video, ensuring an enhanced user experience.
To further enhance the experience, the RXT‑4DV has a 4-watt powered speaker, proximity sensor, & ambient light sensor. The internal web server allows control from remote devices like cell phones and computers.
The RXT-4DV is a powerful ‘all-in-one’ solution for your presentation and automation needs. Core Studio tool is available free of charge for code and interface creation.
he RXT-4D is a 4” desktop touch panel with a ReAX™ JavaScript based control system. Ideal for conference rooms, educational facilities, hospitality, digital signage and more. The RXT-4D has a beautiful 720 x 720 1:1 aspect ratio touch screen with 160° viewing. It is also designed to work as a Dante™/AES67 beam forming microphone with advanced hardware for feedback suppression, auto gain, & noise cancelation.
The 2D/3D graphics engine with 4K30 H.264 streaming decoder provides a powerful solution for complex graphics and video, ensuring an enhanced user experience. To further enhance the experience, the RXT-4D has a 4-watt powered speaker, proximity sensor, & ambient light sensor. The internal web server allows control from remote devices like cell phones and computers.
The RXT-4D is a powerful ‘all-in-one’ solution for your presentation and automation needs. Core Studio tool is available free of charge for code and interface creation.
The supplied mounting plate and paintable cover allow discreet placement on the ceiling or wall in boardrooms and conference rooms where low visibility is preferred.
Two-way wireless transmission allows for remote control of audio and transmission of a secondary program audio to each transmitter.
Multiple units can be combined to support configurations of up to 40 compatible channels and 80 in Europe and select countries (double in HD mode).
Supports standards-based security protocols to manage access and protect data, including 802.1x, secure boot/storage, and AES encryption for wired and wireless control and audio data.
Selectable RF output power to allow optimize spectrum usage by selecting the appropriate wireless output power for the desired operating range.
Diagnostic LEDs offer easy visual reference to system activity and indicate power, wireless connection, and network audio status.
A CUI can be accessed on a PC connected to the same network enabling access to all device properties for optimization and user specific settings.
The Microflex® Wireless neXt system is more than a wireless microphone. It’s an all-in-one audio solution for hybrid classrooms, training rooms, and lecture halls that provides superior audio for conferencing and presentations at the same time. Despite its comprehensive feature set, the MXW neXt system is easy to set up in minutes.Bluetooth functionality not currently available.
Current Microsoft Teams and Zoom Rooms certification solutions include MXWAPXD2, MXW6X, and MXN5W-C.
Relacart latest innovation, the ACM-600 AI Ceiling Microphone Array, combines cutting-edge AI technology and advanced acoustic engineering to deliver unparalleled audio quality for any space. This ceiling-mounted solution is not just a technological breakthrough but a versatile tool for environments like conference rooms, classrooms, and auditoriums, ensuring every word is captured with precision and clarity.
SpectraBand SpectraBand technology delivers a tuning range of 470 – 1525 MHz in a single unit – an industry first. Experience ultimate flexibility as you find open spectrum away from the crowded UHF TV band. The included A20-Monarch antennas can receive the entire 470-1525 MHz range, keeping your kit small and tidy. NexLink Control your A20-Mini transmitter using the built-in long range remote control link. NexLink’s advanced long distance modulation reaches even farther than the A20-Mini’s long-range audio transmission.
Network away You’ve got options. Configure your A20-Nexus for Dante audio, use the built-in web server to control the receiver using the web browser on any computer, tablet, or phone, and even power the A20-Nexus remotely using Power-over-Ethernet (PoE). Real time spectrum analysis The days of waiting for a frequency scan to complete are over. Instantly visualize the fluctuating RF spectrum around you in real time – while running all channels of the audio. GainForward With the A20-Mini’s GainForward Architecture, easily set the Mini’s gain, low-cut, and limiter directly on your A20-Nexus or mixer-recorder. This immediate, local control of gain means never having to adjust the gain on the transmitter again.
Go cableless . Thanks to its half-rack size, the A20-Nexus is equally at home in a rack, in a bag, or docked to an 8-Series mixer recorder for a cableless kit. Long range The A20-Nexus and A20-Mini deliver the longest transmission distance of any system on the market.¹ The state-of-the-art, 100% digital long-range modulation offers the same great audio quality (10 Hz – 20 kHz) as our Standard modulation scheme with a longer range for both line-of-sight and heavy multipath transmission. Shoot for the stars The A20-Nexus joins the growing lineup of Sound Devices Astral™ wireless products. With industry-leading features like SpectraBand, GainForward, NexLink, long range modulation, real time spectrum analysis, and more, the Astral Series pushes the bounds of wireless audio.
Adopting Zenwin's original digital conference technology, in compliance with the IEC60914 international standard
Exquisite and elegant countertop structure, ergonomic and modern, mainly used in high-end conferences and other professional occasions
Using zinc/aluminum alloy material, anodizing process, CNC precision machining high gloss edge, high fidelity microphone head conference unit adopts a φ 16 capacitive dual microphone head, integrated with a double mesh windproof cover, which can reduce environmental noise and wind sound, and has a good pickup distance;;
The chairman has a priority button function, which can close the representative unit that is speaking; The priority button (Priority) and approval button (Permit) can be used to control the speaking order and the atmosphere of the meeting;
High resolution display screen: supports dynamic clock display, call service, message reception, image display, volume adjustment, customized interface, check-in, voting, voting, election, scoring and other functions.
Circular handle connection technology ensures that the failure or replacement of one microphone will not affect the operation of other units in the system, and a connection failure between microphones will not affect the system's operation, thereby making the system more reliable;
Support line "hot plug" and can add microphones to the system at any time
An effective sound collection angle of 100 ° can prevent interference from adjacent microphones and suppress howling.
Microphone adjustable elevation angle -50 ° to 45 ° (horizontal)
Adopting Zenwin's original digital conference technology, in compliance with the IEC60914 international standard
Exquisite and elegant countertop structure, ergonomic and modern, mainly used in high-end conferences and other professional occasions
Using zinc/aluminum alloy material, anodizing process, CNC precision machining high gloss edge, high fidelity microphone head conference unit adopts a φ 16 capacitive dual microphone head, integrated with a double mesh windproof cover, which can reduce environmental noise and wind sound, and has a good pickup distance;;
The chairman has a priority button function, which can close the representative unit that is speaking; The priority button (Priority) and approval button (Permit) can be used to control the speaking order and the atmosphere of the meeting;
High resolution display screen: supports dynamic clock display, call service, message reception, image display, volume adjustment, customized interface, check-in, voting, voting, election, scoring and other functions.
Circular handle connection technology ensures that the failure or replacement of one microphone will not affect the operation of other units in the system, and a connection failure between microphones will not affect the system's operation, thereby making the system more reliable;
Support line "hot plug" and can add microphones to the system at any time
An effective sound collection angle of 100 ° can prevent interference from adjacent microphones and suppress howling.
Microphone adjustable elevation angle -50 ° to 45 ° (horizontal)
Adopting Zenwin's original digital conference technology, in compliance with the IEC60914 international standard
Exquisite and elegant countertop structure, ergonomic and modern, mainly used in high-end conferences and other professional occasions
Using zinc/aluminum alloy material, anodizing process, CNC precision machining high gloss edge, high fidelity microphone head conference unit adopts a φ 16 capacitive dual microphone head, integrated with a double mesh windproof cover, which can reduce environmental noise and wind sound, and has a good pickup distance;;
The chairman has a priority button function, which can close the representative unit that is speaking; The priority button (Priority) and approval button (Permit) can be used to control the speaking order and the atmosphere of the meeting;
High resolution display screen: supports dynamic clock display, call service, message reception, image display, volume adjustment, customized interface, check-in, voting, voting, election, scoring and other functions.
Circular handle connection technology ensures that the failure or replacement of one microphone will not affect the operation of other units in the system, and a connection failure between microphones will not affect the system's operation, thereby making the system more reliable;
Support line "hot plug" and can add microphones to the system at any time 8. An effective sound collection angle of 100 ° can prevent interference from adjacent microphones and suppress howling. 9. Microphone adjustable elevation angle -50 ° to 45 ° (horizontal) 10. Screen resolution 480 * 480
Adopting Zenwin's original digital conference technology, in compliance with the IEC60914 international standard
Exquisite and elegant countertop structure, ergonomic and modern, mainly used in high-end conferences and other professional occasions
Good resistance to GSM interference; Using a detachable microphone rod, easy to operate and convenient for product maintenance;
Adopting a capacitive microphone with a dual color ring indicator light, it is easy to distinguish between speaking and waiting states;
The chairman has a priority button function, which can close the representative unit that is speaking; The priority button (Priority) and approval button (Permit) can be used to control the speaking order and the atmosphere of the meeting;
Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software to make speakers speak more smoothly at different volumes;
The microphone is equipped with a super quiet switch, which can control the operation of microphone pickup. The microphone has a bright luminous ring to indicate the working status;
Built in high-performance DSP audio chip, optimized audio signal, restored vocals, and excellent sound quality
Two color LED indicator light, speaking in green;
The unit comes with a 2-meter Category 6 RJ45 network cable;
The unique ID number of the microphone unit is assigned to the conference unit by the conference system host when setting the ID;
DMS-301C is a wireless 5G digital conference chairman unit with a 2-inch LCD screen displaying device status. It is easy to operate and has permission buttons and speaking buttons. Participants only need to lightly press the speaking button to speak, with clear sound quality. It adopts professional DSP processing design, making the speaker's voice clear and thorough. It supports functions such as discussion speaking and camera tracking, and has built-in 128 bit AES and 64 bit PIN code dual encryption technology, supporting HD Sound audio and video transmission protocol.
The representative unit adopts the latest six core ARM Cortex-A53 chip and is also equipped with a multi-core neural distributed GPU acceleration processor to ensure stable and reliable acquisition and transmission of distributed transmission signals;
Support HD Sound digital audio and video transmission technology, sending and receiving 1024 audio channels simultaneously through a regular Ethernet cable;
Using the top-level ADC chip AK5392 with 24bti/192KHz, the dynamic range and signal-to-noise ratio achieve CD sound quality
Adopting multi-channel digital filter transmission technology to solve crosstalk, noise, distortion, and noise caused by mutual interference between units;
POE switches are directly powered and connected to the switch using standard Ethernet cables, making system setup simple and flexible.
Support partition sharing technology, one conference host supports 4-6 conference rooms;
Support speech discussion, camera positioning and tracking functions
Adopting a fully copper shielded design to resist mobile phone RF interference, it has super strong resistance to mobile phone, electromagnetic, and high-frequency interference.
Adopting intelligent neural distributed algorithm technology to completely eliminate the phenomenon of host crashes caused by multiple units turning on at the same time;
Adopting the IEEE1588 precision clock protocol for clock synchronization and zero delay transmission of audio and video signals.
Using uncompressed audio transmission, 48K sampling rate, providing clear and perfect speech
Adopting soft control buttons to effectively avoid mechanical button sound impact;
Adopting Zenwin's original digital conference technology, in compliance with the IEC60914 international standard
Exquisite and elegant countertop structure, ergonomic and modern, mainly used in high-end conferences and other professional occasions
Using zinc/aluminum alloy material, anodizing process, CNC precision machining high gloss edge, high fidelity microphone head conference unit adopts a φ 16 capacitor microphone head, integrated with a double mesh windproof cover, which can reduce environmental noise and wind sound, and has a good pickup distance;
The chairman has a priority function and can close the representative unit that is speaking; The priority button can be used to control the speaking order and the atmosphere of the meeting;
The pure speech conference unit is equipped with a 0.96-inch color screen displaying conference mode, conference unit ID number, and chairman/representative function;
Independently adjust the microphone gain, which can be adjusted on the microphone or computer management software to make speakers speak more smoothly at different volumes;
The microphone is equipped with a super quiet soft control switch, which can avoid mechanical sound amplification caused by the switch and control the operation of microphone pickup. The microphone has a bright luminous ring to indicate the working status;
Built in high-performance DSP audio chip, optimized audio signal, restored vocals, and excellent sound quality
Two color LED indicator light, green for speaking, flashing red when applying for speaking and setting ID;
The unit comes with a 1.5-meter Category 6 shielded Ethernet cable;
The ID number of the microphone unit can be assigned to the conference unit ID through the ID switch of the conference host;
Third party applications that use Dante Ready™ allow customers to purchase and activate Dante audio channels after installation using the Dante Activator tool in Dante Controller 4.5 or higher. Dante Activator is tied to an active Audinate user account and accepts standard credit card payments.
