Everyones talking about using AoIP (Audio over IP) in the broadcast world, but what does it mean to the operators actually making the production - the intercom users? And after years of interoperability taking the lowest common format of analogue tie-lines (also known as 4-wires), has intercom today already moved into the digital world by sharing digital audio resources with other types of hardware?
Until recently there were several standards for transporting digital audio between hardware; some of them common to the manufacturers, some standard amongst a number of different manufacturers and some just totally bespoke. Todays standardisation on Dante and AES67 helps simplify matters, but how?
Interoperability has always been part of the intercom world; analogue 2-wire beltpacks allowed different areas to work together over microphone cables or tie-lines, as an early example. Then the intercom world explored the benefits of telephony to interconnect operators across sites and even countries using POTS (analogue telephony). This was basically extending 4-wires across telephone copper cables - it was expensive and had balance issues resulting in feedback of your own voice, but it worked sufficiently well. Add simple GPIOs (general purpose interfaces, contact closures etc.) and you had some basic signalling between intercom systems. Digital Intercom had to use multiple lines to carry both voice routing and signalling data - anyone remember voice over data modems?
When ISDN came along the digital interconnection started to become much easier to manage, and ISDN interfaces for everything audio became the fashionable item to have. It was still costly, and the ISDN lines often had to be booked in advance, but it worked in a similar way to POTS, just with embedded data and caller ID.
Within the studio complex MADI grew to be the digital audio interface of choice, allowing multiple streams of AES3 audio from, for example, a sound desk mix-minus bus to the intercom system over a single cable, rather than 64 pairs of analogue 4-wires. MADI (AES10) worked well between devices and was effectively the forerunner of todays AoIP communications.
Early adopters in IP connectivity from a manufacturing point of view include Trilogy Communications, which launched the worlds first IP connected intercom in 2002. The system used standards-based IP networks to allow intercommunication between intercom matrices, which permitted Trilogy to install several hundred systems world-wide that were all interconnected. At this early stage intercom bandwidth was controlled by the use of compression codecs; in this case G.722 was the best one that could be used, with an audio bandwidth of around 7.5kHz, which has since been proven to be adequate for talkback needs.
Advances in technology saw the introduction of IP connectivity of the intercom panel, again using standards-based IP networks. This meant that panels could be sited anywhere within the Wide Area Network and put more and more onus on the IT engineers rather than broadcast engineers, which was an issue in itself. Most IP panel connections shunned compression of the audio to achieve lower latency, since its no good telling a cameraman to switch shots and having the intercom arrive some seconds too late! IP panels were also fighting for bandwidth with other uses of the IP network, be it an office network or something also serving video, so we were introduced to the world of VPNs.
Various interoperability Audio over IP standards started to appear, and the joys of a plug-fest where different manufacturers joined their hardware together to test interoperability became fashionable. The EBU worked first on video interoperability then proceeded to work on audio interoperability between intercom manufacturers. Differing standards included Ravenna, AVB and Dante, all fighting to be the technology to use as a joiner between disparate audio systems. It was obvious that a single standard was needed and various groups were working together to achieve something that we knew would just be plug and play.