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On-air Telephone Interfaces
Radio broadcasters rely heavily on their telephones. Whether used for news, interviews, contests, requests or talk radio, the ability to easily manage phones on-air is a long-standing requirement, and that's unlikely to change. As technical director of Comrex, I've had a long-time understanding of the state-of-the-art in this domain and that was only enhanced by Comrex's acquisition of the Gentner line of telephone interface equipment in 2002. I'll describe here the current paradigm in telco interface and introduce where this stuff is heading ‗ and it's heading there fast.
Hybrids aren't just electric cars
The essential piece of studio telephone integration is the telephone hybrid. As shown in Fig. 1, the two-wire nature of the local telephone loop necessitates that the conversation back and forth must take place over the same circuit. Fig. 1A shows this as being analogous to two people speaking into a single tube. The downside of this arrangement is that each speaker will hear himself echoed in the tube louder than he'll hear the far end speaker.
Fig. 1B is closer to reality, showing the telephone network as separating the send and receive channels. But in the case of analog phone lines, these are reconverted to a single two-way circuit for delivery to the customer. In both the 1A and 1B scenarios, a hybrid must be used to convert the studio-side, two-wire circuit back to four-wire for use on-air. The output of a hybrid will consist of only caller audio, with all the host audio digitally subtracted from this output. It can be assumed with modern digital hybrid circuits that any remaining host audio is more than 20dB below that of the caller. Lack of a proper hybrid arrangement will result in studio host audio coloration since the hybrid feed-through will add with the main host microphone in the console and produce an effect often described as talking into a barrel. It's important to note that the main function of these systems is not to alter the send audio, remove garbage audio introduced by poor quality telephones or even far-end echo, but simply to keep the host audio unmolested by telephone feed-through.
Fig. 1C shows the best possible scenario, a four-wire connection at the studio side that keeps a completely independent path in each direction. In this case the actual hybrid function (two-to-four wire conversion) is not required to provide studio send-receive isolation. This is typical of systems that utilize ISDN, T1 or voice over IP (VoIP) interfaces at the studio (more on that later).
But the term hybrid has come to mean much more than this conversion. Telephone feeds suffer from noise and hum, as well as widely varying caller levels (especially these days due to use of poor quality cell phone microphones). So it's essential that a hybrid (whether it utilizes a two or four wire interface) contain line filtering and AGC to combat these evils. Also, a talk-show host may want to dominate a conversation, so some kind of ducking function is required, detecting host voice activity and reducing caller audio level to a more submissive level.
Call management systems
Taking the concept further, once you move beyond one or two phone lines, you'll need the call management capabilities of a multi-line (or talk show) system. This typically consists of a mainframe that connects to the telephone lines, and one or more control surfaces to answer, park, screen and put calls on-air. In addition, studios taking callers will need some kind of screening capability to talk to callers in advance and inform on-air talent of their intentions. A block diagram of such a system is shown in Fig. 2.
Modern trends in these systems revolve around enhancing remote control capability. Higher-end talk show systems will include the ability to put these capabilities on a Web page, and this will usually combine the remote control and screening functions. The Comrex STAC talk show system's Web-based screening page is shown in Fig. 3, which is delivered by a Web server built into the mainframe of the unit. By use of a Web browser, calls can be screened and routed from any location that has Internet access, including remote broadcasts. Also, this removes the requirement for the client computer to have any specialized software installed to view and control the system. Multiple locations can monitor the caller queue and screening information as they listen to the show.
Additional features offered by some screening software may include reflection of caller ID information and logging screening information into a database.
Many options are available to deliver multiple on-air phone lines. BRI-ISDN lines are certainly useable as voice lines, and have the advantage of pre-separated send and receive paths. Telos Systems has pioneered use of these for talk show systems with much success. But in my view the long-term outlook for BRI-ISDN service is uncertain, as fewer and fewer non-broadcast applications require it.
As the number of phone lines brought into a facility increases, the cost-benefit tips toward a multiplexed digital delivery such as T1 or PRI-ISDN. For maximum flexibility with use on the widest range of telco interface equipment, individual lines from the T1 may be converted back to analog POTS lines in the channel bank. While effective, this approach isn't entirely elegant, since you are intentionally introducing an additional digital-to-analog as well as a four-to-two wire conversion that must be reversed by the hybrid within the talk show system. The market for direct-interface T1/PRI broadcast telephone integration is dominated by the higher-end Telos 2101.
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