Intercoms and Talkbacks Update

  • By David Kirk

Intercoms and Talkbacks Update

David Kirk follows up from June’s issue on the latest developments from the manufacturers of broadcast communications technology…

2021 has seen a gradual rollout of new and upgraded hardware and software across the intercoms and talkbacks product category, with an increasingly strong focus on IP integration. Internet-based intercom broadcast intercom tools are also becoming increasingly commonplace for use where internet connectivity is reliable.

AEQ’s Phoenix Mobile is a compact IP audio codec and communications hub designed for remote production. It includes a high-visibility colour display, digital mixer, multiple analog inputs, hinged protective cover, 12 volt DC power supply/charger, and optional long life battery. SIP server services are offered at no additional cost. Optional communication modules can be used in conjunction with the onboard IP. POTS and ISDN telco modules are now available and others are in development. A USB connector on the rear panel can be used to establish connections over 3G or GSM networks with an external cell-phone-style data gateway. The hub can simultaneously provide co-ordination or talkback from studio to remote device using optional communications modules, in addition to the remote-to-studio main programme feed. ‘ControlPhoenix’ software allows control and management of Phoenix Studio, Phoenix Mercury and Phoenix Venus audio codecs through IP. The equipment can either be part of the same local area network as the PC that controls them, or be remotely situated. Local devices are automatically discovered when the software is launched.

Clear-Com has updated to its HelixNet digital network party-line and LQ series IP interfaces. Firmware release 4.2 adds visual communication, two-way radio capability, and new administration capabilities. LQ IP interfaces can connect to any type and brand of intercom or audio device over IP. The new update adds the ability to save and restore LQ unit configuration. New functions include a tally allowing users to correlate what they hear on their intercom station with individual channels. It also provides a visual cue. New administration functions include the ability to set minimum and maximum volume, disable channel mute, configure loudspeakers and set the headset toggle key. Operation of two-way radios is made easier by a new secondary talk action function which allows individual talk keys to initiate a visual call signal or LQ network control event. LQ can now support eight Agent-IC clients as well as up to eight Station-IC clients. “Customer feedback was a driving force in the 4.2 firmware update which has effectively taken us a step forward in maximising the functionality of HelixNet and LQ,” says Clear-Com Product Manager Kari Eythorsson.”

Also new from Clear-Com, FreeSpeak Edge incorporates a new 5 GHz chipset and OFDM to delivers 12 kHz audio bandwidth plus support for over 100 beltpacks and 64 transceivers. It can be combined with FreeSpeak II 1.9 GHz and 2.4 GHz systems to provide three bandwidths across a unified system. The beltpack includes asymmetrical concave/convex top buttons, eight programmable buttons, rotary controls on both sides, plus a master volume control and flashlight on the bottom. Microphone and speaker are included for headset-free or desktop operation. Each transceiver supports 10 beltpacks and includes attenuation plus external antennas for custom RF zones as well as wall and mic stand mounting options. Comrex has introduced a new cloud service called Gagl that delivers conferenced audio from multiple contributors to Comrex hardware IP codecs. Up to five users can send and receive audio from computers and smartphones by simply clicking a link using any common web browser. Their audio is conferenced (if there is more than one user) and delivered to a Comrex hardware codec such as ACCESS or BRIC-Link II. All participants can hear other participants, and the codec can send audio back to them. Gagl could be used as a hub to support multiple simultaneous connections. Its low latency allows use for call-in talk radio. Gagl can also be used to allow a single contributor to connect back to the studio. Additionally introduced by Comrex is the latest version of its Access Rackmount intercom. Access NX Rack features completely redesigned hardware, allowing for AES67, AES3 or analog audio connection. NX Rack is operated via a new HTML5-based web user interface and is backward-compatible with all Comrex IP audio codecs and the Comrex FieldTap smartphone app. Features include the CrossLock network management tool which creates a virtual private network between codecs. Switchboard is a tool allowing a Comrex codec to connect with a Comrex-maintained cloud-based server. Connections can be made between codecs without knowledge of IP addresses at either end of the link. Switchboard also provides presence and status information about connected Comrex codecs and can help make some connections through routers and firewalls.

Glensound’s Beatrice B4+ can receive audio from four separate network locations. Four separate talk buttons route the microphone input to either of the four network outputs. Button operation can be configured as momentary, latching or intelligent. Double tapping on any of the talk buttons produces an alert that will be received by another Beatrice unit. An LED indicates the incoming call alert which will also produce an audio indication on the headphones. The microphone input is on a three pin XLR; features include gain adjustment and a 48 volt phantom power output. The XLR connector can be specified with four or five pins to support headset connection. Each input has a separate level control adjust and a signal present indicator. The Beatrice B4+ can be remotely controlled via GlenController. Monitoring, setup and control are all available via remote control. 

Green-Go produces an Ethernet based digital intercom system. The Beltpack X and Wireless Beltpack X support 32 channels, a programme audio channel and an extra channel for direct user communication. A wireless antenna integrates the Wireless Beltpack X into an existing wired network. The system has no single point of failure because there is no central unit that carries the entire system. Multi-channel stations operate with any of the company’s belt packs or another multi-channel station connected in the same network, without the need of a matrix. The interface range consists of devices that can integrate Green-Go analog intercom systems into a digital Ethernet network, connect remote networks to a network and allow remote user login, or connect radio devices to a network.

Riedel introduced two new comms-related products in 2021: Artist and DisTag. Artist is a decentralised scalable digital intercom network that can be scaled from a single frame to a remotely connected network of nodes. Each node module contains client cards that accept and distribute different signal types such as SMPTE 2110-30/31 (AES67), VoIP, Dante, AVB, MADI, AES3 and analogue audio. The system can be expanded from 8 x 8 to 1024 x 1024 ports. Using inter-node trunking it connect several thousand subscribers within a single system. Scaling is achieved by adding new client cards to an existing node or adding additional nodes. “DisTag is a lightweight, wearable device that alerts users if people around them aren’t maintaining a safe distance,” adds Riedel founder Thomas Riedel. “Worn around the neck or carried in a pocket, it generates haptic, visual, and acoustic signals whenever the mandatory minimum distance to other people is about to be breached.” DisTag offers three signal levels: a two-stage vibration alarm, a two-stage LED signal and a two-stage sound signal. The proximity limits of the warning signals can be individually defined and adjusted in accordance with local regulations for social distance. 

RTS’ DBP is a four-channel/four-button wired beltpack that runs on 802.3af and 802.3at PoE. Up to six DBP devices can be operated from the same port when used in partyline mode. Up to 40 DBPs can be connected to an RTS OMS. “DBP offers both digital partyline and matrix keypanel modes,” says RTS Product Manager Angelo Piga. “Users such as broadcast networks and industrial facilities can expand their comms inventory cost-effectively while continuing to leverage the scalability of their matrix equipment.” 3.5 mm TRRS and XLR ports are provided for connecting headsets, with three different XLR options available: 4-pin female, 4-pin male and 5-pin female. Incoming call notifications are via audible alerts or haptic vibration. DBP also supports Bluetooth audio.

Sonifex recently introduced two new audio-over-IP commentator units. Model AVN-CU2-Dante has two mic/line inputs with a wide, adjustable gain range and +48 volt phantom power as well as two stereo headphone outputs with lockable jack sockets. It uses PoE via Neutrik EtherCon connectors, with primary and secondary ports for power and network data redundancy. SFP cages are also available for networking and there is an additional four-pin XLR 12V DC input. The unit supports up to 16 input and output audio-over-IP channels and up to 16 simultaneous input and output audio-over-IP streams. Stream setup to and from the unit is initially via Dante Controller software with more detailed configuration performed via the built-in web GUI. An RGB display split into multiple sections allows metering and control of up to 24 audio sources. Colour-coded rotary encoders can be used to adjust signal level and pan. An integral mix matrix can be used to mix physical and audio-over-IP sources to four-wire and audio-over-IP destinations. The unit has six pushbutton rotary encoders and 12 key-cap buttons which can be assigned to control input and output levels and panning. Each rotary encoder has a separate colour-coded meter section showing the channel name, detailed level metering, left/right panning and a limiter indication, on a bright daylight reading display. Model AVN-CU4-Dante is a dual version of the AVN-CU2-Dante suitable for operation by three or four commentators. It provides four mic/line inputs and four stereo headphone outputs with lockable jack sockets.

Studio Technologies’ StudioComm System with Dante Support comprises the Model 792 central controller and 793 control console. It can be used to monitor high-channel-count Dante digital and analog audio sources. “Dante has found wide acceptance in professional applications but analog signals continue to play an important role in audio workflows,” says Studio Technologies’ President Gordon Kapes. “That fact, combined with the range of audio formats, can make monitoring a challenge.”

The StudioComm System can handle 16-channels, providing direct support for surround audio. Audio level control and processing is performed in 32-bit depth. Also new from Studio Technologies are two party-line intercom kits that use Dante Audio-over-IP networking. The Party-Line Intercom Kit 1 includes a Model 5421 Dante intercom audio engine and four 372A intercom beltpacks. Party-Line Intercom Kit 2 includes a 5421 but with four 373A intercom beltpacks.

Telos Infinity’s VIP (Virtual Intercom Platform) makes pay-as-you-go cloud-based media production workflows available on devices such as smartphones, laptop or desktop computers or tablets. Control can be direct from the device or via third-party hardware such as Elgato’s Stream Deck.

Unity Intercom offers a communication system, itself called Unity Intercom, which connects remotely over Wi-Fi or a cellular network. It supports features such as local I/O integration via USB devices and Dante. A Unity Intercom system comprises software which runs on an Apple Mac, plus apps that run on mobile devices. Unity Cloud is an alternative way to use Unity Intercom from the company’s cloud-based subscription servers. It supports a single group of users sharing six party-line channels, set up online via a web portal. It is intended for setting up remote comms in a hurry or for specific events. Unity Connect can send and receive 64 audio streams over the internet. Unity Server is a lightweight client that runs in the background on a Mac computer, handling the audio and user permissions for each of the connected Unity clients. The server is used to add and manage user accounts and to configure additional features such as Unity Tally and Audio I/O.