STAC VIP Overview

VoIP (Voice over IP) technology has taken the telephone industry by storm due to its flexibility and low costs. It’s time for radio broadcasters to reap the benefits of this change. STAC VIP smoothly integrates legacy POTS lines with VoIP technology to deliver a new way to manage telephone calls for talk shows, interviews and contests. STAC VIP can take traditional POTS calls, but breaks new ground by handling calls from "HD Voice"-capable telephones and Smartphones apps. Complete with the STAC IP Call Screening and Control Interface, the STAC VIP Caller Management system will even integrate with your VoIP PBX system. STAC VIP leads the charge in migrating old-fashioned on-air telephones to dramatically clearer wideband audio quality – just another reason to put Comrex on the line.


Features STAC VIP Control Surface : STAC VIP utilizes the familiar STAC Control Surface which provides clear indication of phone line and caller status. Super bright colored LEDs and an ergonomically friendly button layout give a concise overview and control of all callers at a glance. STAC CS integrates unique features like "Busy All Lines" and control of the "Auto Attendant" feature, which answers calls and plays out a custom message before parking calls on hold. STAC CS connects to the mainframe via IP, so it can be located anywhere IP connectivity is possible – even off-site.

Single and dual-line digital telephone hybrids


DH20 / 22 Overview

Are you looking for the very best, most affordable digital telephone hybrid available? You just found it. The DH20 and DH22 Digital Hybrids incorporate the latest in DSP technology to provide you with the best sound and easiest installation.


The DH20 and DH22 are high quality single-line and dual-line telephone interfaces, designed to interconnect a standard telephone line and your audio equipment, allowing you the ability to send audio to and receive audio from a connected telephone line. The DH20 and DH22 are ideal when the very best audio quality is required for talk shows, news feeds, production, recording studios, and internet applications.


Our newest DSP technology delivers better and more reliable sound quality. The DH20 and DH22 deliver the deepest and most stable hybrid null, ensuring maximum isolation between the send and caller audio. That means no echo and no "bottom of the barrel" sound.


In addition, we've added selectable automatic gain control (AGC) and caller ducking. With AGC, each call will be delivered to your equipment at the same level. Caller ducking reduces the caller's audio level when the announcer speaks, providing a more "controlled" environment. We've also added a speaker amplifier so you can connect a speaker directly to the unit and monitor callers without headphones.


DH30 Overview

With the variety of phone systems in the world today, it is becoming increasingly difficult to have all callers sound the same on-air. To help bring uniformity and high quality sound to a broadcast talk show environment, we have created the DH30. The DH30 provides a high quality interface between a standard telephone line and your audio equipment.


Designed with both AES/EBU and analog inputs and outputs, the DH30 can interface with the latest digital and analog consoles. To ensure quality sound in studios that have open speakers and microphones, the DH30 was created with an acoustic echo cancellation feature.


The DH30's digital signal processing (DSP) has multiple adjustable parameters to provide you with the best send and caller audio signals. New audio processing features such as the compressor and downward expander functions ensure that the best audio reaches your audience. All audio processing features are programmable via the front panel controls. For the deepest, most reliable hybrid null, and the best quality caller audio for your application, the DH30 is the answer.


Easy to use and install, the DH30 has simple front panel controls and remote control capability. These controls put the On, Off, Rec, and Cue functions right at your fingertips. The DH30 also includes a 2W monitor amp with volume controls right on the front panel. The Rec function sends a start/stop signal to your external recording device. The Cue function allows you to control which audio signal should be sent to the caller, allowing you to switch from the Send input to the Cue input at the touch of a button.                                                                                                   



 IP Streaming Server


Explosion 1: New Model

Low cost, high-performance solution for point-to-point IP audio conversion


The Comrex BRIC-Link is a low-cost, high-performance solution for audio-to-IP conversion. Leveraging many of the core technical aspects of Comrex's successful remote broadcast ACCESS product line, BRIC-Link provides for an elegant way of moving linear or compressed audio with very low delay. BRIC-Link is very simple to use, and can be used over a wide range of IP links. While it carries an entry-level cost, BRIC-Link maintains superb audio specifications and hardware reliability, making the system suitable for STLs and other mission-critical functions without the expense required of more full-featured codecs.

Create six mix-minus feeds from one bus on your console


Back in the good old days, radio and TV talent could be sent out into the field with a link back to the studio and the ability to monitor the station's off-air signal. The fact that they heard their own voices in headphones or ear pieces was not distracting as there was no delay between speaking and hearing. Goodbye analog, hello digital!

· The MMB takes one mix minus feed from the console, and creates 6 feeds for remote or telephone use.

· MMB's can be stacked for 12, 18 or more feeds.

· Separate level controls are provided for each input to the bridge.

· IFB (interruptible fold-back) allows cues to be sent to individuals or the whole group.




IFB can be set for "ducking" or complete muting of the program audio.

With huge delay times resulting from digital broadcast transmitters, you can forget about monitoring off-air. So, today's remote feeds are usually sent through digital codecs and there can be considerable processing time for audio in both directions. Your brain can only handle a relatively short echo of your voice while speaking. As a result, it has become absolutely necessary to keep the remote audio from being sent back to the remote site. This is done with mix-minus: A mix of all needed audio that the remote site needs to hear, minus the audio coming from the remote site itself.


It is usually easy to create one mix-minus feed on a broadcast console. Most consoles have an audition or auxiliary bus which allows the creation of an additional mix of certain console sources without affecting the main or program mix. If a telephone hybrid is fed by this mix, and a codec feed is required later, the console must be reconfigured for each mix-minus. This can create confusion and add complexity to a live broadcast. Also, if the hybrid and codec need to be on-air simultaneously, a single mix-minus is insufficient. Newer consoles may provide for more than one mix-minus feed but you always seem to need one more!


With the Mix-Minus Bridge (MMB), you can easily create up to six dedicated mix-minus feeds with each remote source having its own dedicated, full-time feed. No configuration is necessary before airing single or multiple remote sources. The MMB simply requires that you configure an audition or auxiliary bus on your console with a single mix-minus but it will remain permanently configured. There is also an interruptible fold back input (IFB) for sending producer cues to the field. Additionally, the MMB can be expanded to 12, 18 or more channels by stacking units interconnected by CAT5 cables.