• Audiotx Communicator Client
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Being a freelance commercial voice talent today requires many different methods of providing product to his clients. While ISDN audio codec technology and the Internet have made the voice-over (VO) world a much smaller and faster place, the need for compatibility and dependability is paramount. To work locally and internationally via ISDN, voice talent the world over are not only the talent but also audio engineers/editors and telephone company technicians. The financial investment in equipment and its options can be quite expensive and daunting. The need for a compatible and dependable, yet affordable and easy-to-use audio codec can make the technical job of being a voice talent a great deal easier. That's what led me to AudioTX.

The AudioTX Communicator is an MPEG Layer II, III and G.722/G.711-compatible, software-based, full-bandwidth, bi-directional ISDN and IP/Internet audio codec for the PC. The user-interface operating screen is simple and straight-forward, incorporating audio metering of incoming and outgoing signals, framing indicators, input/output source selection and level controls and an easy-to-use, user created phonebook. Once connected and framed with the far-end codec, the software can also provide incoming Caller ID information if so provisioned.

Their AudioTX Communicator system will then always use that setting for the return audio, regardless of the settings used by the system on the other end of the line ie me and my magnificent bandwidth busting LOSSLESS setting. And regardless of who dialled who first. This is making me think that things are shifting. The AudioTX Communicator is an MPEG Layer II, III and G.722/G.711-compatible, software-based, full-bandwidth, bi-directional ISDN and IP/Internet audio codec for the PC. The user-interface operating screen is simple and straight-forward, incorporating audio metering of incoming and outgoing signals, framing indicators, input/output source. The Communicator can operate over Ethernet or private wide-area networks. Using a standard PC network card or other installed networking hardware, the AudioTX Communicator can provide live audio links over IP networks. Using existing installations of hubs and routers makes the scale of a given network irrelevant.

I use a P-4, 2.5MHz PC that incorporates not only AudioTX, but also an audio recorder/editor and a good word processor for reading directly from the monitor. However, an affordable Pentium-II or Athlon 400MHz (or above) system with 128MB of RAM, Windows Me and PCI bus running Windows 98-SE, Windows 2000 or XP is the minimum requirement for the host PC. Hard-drive capacity should be determined by operating system and audio file storage requirements for your needs. However, a 10GB drive will accommodate the PC operating system and AudioTX with ample storage space to spare. I also use a professional audio card with S/PDIF I/O. However, most moderately priced, full-duplex PCI audio cards, such as those from Sound Blaster, will work nicely.

Installation of the company's software was typical of most Windows installations, and updates are posted on the AudioTX website. There is a 30-day evaluation trial version of the most recent version. When updating, the new version overwrites the old, so there is no confusion and the phonebook names and settings and all operating parameters for both operating modes are preserved. When purchased, the software is supplied with a small USB or printer port dongle, which acts as the software license.

After setting up and configuring the ISDN line or IP connection and installing the software, entering the phonebook information into a list of my most often used connections was easy. To initiate a link, I highlight the name I've programmed into the phonebook and click the dial/connect button at the bottom of the interface window. When receiving a call, the software determines the configuration of the caller (such as algorithm, audio mode and bit- rate) and adapts. The software synchronizes the path and passes audio. It takes from 2 to 15 seconds for audio delivery to begin.

The only problems I've encountered are from older codecs of varying brands that haven't been updated with their manufacturer's most recent firmware. However, in most cases, that's usually only a single parameter change on the part of the far-end device to remedy.

This software offers two separate operating modes: an ISDN mode and an IP/Internet mode. The two modes are operated from a similarly styled window switched by a tab at the top of the interface screen. The two modes share a common audio card, a unique and useful audio file transfer system and several global parameters found under the main window's pull down menu under settings. There are four headings under the settings menu. The first is global, which includes identification settings when connected, host computer speed for above and below 1GHz operation, G.711 settings for the United States or Europe and auto connect on start-up, which asks for a connection type and destination when starting the software. These need only be set once for most operations.

The second settings heading is audio, which includes audio quality for MPEG II and MPEG III (fast coding or studio quality), audio buffering for the sound card, and a sample rate conversion enable/disable setting to convert incoming sample rates.

The next heading is answer settings. The two modes, ISDN and IP/Internet, are switched by a tab at the top of the screen. The ISDN side handles incoming calls and provides a parameter to accept or deny incoming data and voice calls. It also stores the ISDN numbers to use if multiple ISDN lines are available, and the type of compression, sample rate and audio mode used to answer calls.

The IP/Internet side of this screen is quite similar, with the exception being that under a section called accept incoming calls a DNS ID is entered instead of telephone numbers. Also, an internal port for incoming data is selected. The last heading under the settings menu is SOCKS/Proxy version settings and username/password info if required.

The ISDN line requirements for full-bandwidth (20Hz to 20kHz) and full duplex (bi-directional) operation are minimal and identical to those of any other stand-alone codec; a single standard BRI (Basic Rate Interface) with voice and data activated on both B channels. Most phone companies have become quite familiar with this simple ISDN line and configuration so normally ordering, installation time and hassle is minimal.

The ISDN will connect in all modes to most all popular codecs having MPEG Layer II, III and G.722/G.711 algorithms through an inexpensive ISDN internal PCI terminal adapter/modem you install in the PC. I use the Eicon DIVA PCI ISDN modem with an external NT-1 adapter. In my case, the ISDN line is connected directly to the NT-1 via an RJ-45 connector then to the EICON ISDN TA in the PC via a CAT-5 cable. Depending on which ISDN TA/modem you choose for your system, you may not need an external NT-1. However, I found it more cost-effective for my situation to purchase an external NT-1 than one incorporated on the ISDN TA/modem itself.

The IP/Internet mode requires at least a 140kb/s broadband connection to accomplish full bandwidth, bi-directional audio across a network or the Internet. This mode works on an internal (in house) network, the Internet or ATM network and (currently) only to another Communicator software codec on the far end. When using it in IP/Internet mode, the software connects through a 10/100baseT Ethernet card directly to the Internet through an external ADSL/DSL/cable modem connection, or indirectly through an internal network. It will not work with external serial connected modems or dial-up connections.

Calling and receiving using the IP/Internet mode is almost the same as using the ISDN mode, except the user enters the IP address of the remote system instead of an ISDN number. The only drawback to the IP/Internet mode I've found was when I set the desired connection bit rate higher than the capacity of my ADSL connection. Occasionally, it would lose data and the audio signal would chatter a bit. However, there are three user-selectable buffer settings in the phonebook for each individual entry to help remedy such situations, but no one can predict the traffic on the Internet at any given time. Combine that with the nature of ADSL and its understandable how this can happen. A 140kb/s or higher broadband connection is the minimum required for the IP/Internet mode. Not only is it probably the most inexpensive of the ADSL connections, but it also provides a solid 128kb/s connection with some overhead. My broadband connection (ADSL) is rated at 160kb/s up and 250kb/s down.

I use the ISDN mode and the IP/Internet mode and, aside from a minor amount of additional latency in IP/Internet mode—again due to the nature of ADSL and traffic on the Internet at any given time—the operation and quality of the signal is identical, if not even slightly better to my ear as compared to the most popular stand-alone codecs. Also, the G.711 compression mode works well for standard phone patch sessions. This is possible if the ISDN line is configured for voice and data, and the software is set to receive incoming calls.

In the ISDN and IP/Internet modes, there is a useful feature that allows file transfers during a live connection. If both ends of the link are using the Communicator software, it enables either end of the link to send .WAV files across the link. When sending a file, the internal sound card is muted until the transfer is completed or aborted. The add file button brings up a browser screen so files stored on the computer or internal network can be accessed.

To transfer a file, add the file or files you wish to transfer to the playlist, then click the play button. To remove a file from the file list, simply highlight the file and click the remove button adjacent to the file list display window.

Being a voice-over talent, I most commonly transmit in mono, 128kb/s, 48kHz, MPEG Layer II but quite often receive in stereo to be able to read to a music or SFX track. I'm also often asked to do dialog with other talent on the far-end and being recorded by the far end or via a bridge link to a third studio. Also, I'm often required to be able to receive in a different algorithm than that which I'm transmitting and all with no problems.

I chose the AudioTX Communicator for several reasons. For my purposes, it was far easier to use than most stand-alone products. It incorporated everything I needed, not only for the present and all its varied requirements, but also for what I feel will be future requirements for my business. I could install and operate it from the same computer I use to read text from e-mailed or downloaded scripts and use to access the Internet. Also, even with the computer, ISDN TA/modem and professional audio card, the cost (minus the ISDN line and associated charges or a broadband Internet connection) is about a third to half of most stand-alone codecs. I like to refer it as a “codec for dummies.”

Editor's note:Field Reports are an exclusive Radio magazine feature for radio broadcasters. Each report is prepared by well-qualified staff at a radio station, production facility or consulting company.

These reports are performed by the industry, for the industry. Manufacturer support is limited to providing loan equipment and to aiding the author if requested.

It is the responsibility of Radio magazine to publish the results of any device tested, positive or negative. No report should be considered an endorsement or disapproval by Radio magazine.

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