Manual

32
Even in “old-fashioned” analog telephone circuits, it’s possible to create a 10mS round-trip delay on a long
distance call. Now add in the requirement that modern VoIP-based systems have inherent windowing and
buffer delays, and its easy to pile up over 100mS round-trip on a call. A delay of this length will typically not
impede interactive conversation, but will certainly create an intolerable “slapback” environment if the caller
hears his own voice delayed.
The telephone network employs digital echo cancellers at various nodes along the path of a phone call
to avoid this scenario. And when they malfunction or are “untrained” at the start of a call, the effect is a
dramatic echo in the caller’s ear.
Many users installing a studio-based phone system for the first time make the mistake of applying audio to
the outgoing “send’ port that contains the main program feed - the same audio used to feed the transmitter
or webstream. Since this mix contains the caller’s own audio, and there’s an inherent delay in modern digital
systems, the “slapback” effect is immediate.
The solution here is mix-minus-- a term used for a special mix of audio that explicitly excludes one source--the
audio coming from the place the mix-minus is being sent. To put it another way, mix-minus is the entire studio
mix minus one audio source.
So how do we create this special audio mix? On modern studio systems, this is usually well defined and
easy to do. Many consoles feature channels dedicated to telephone interface, and part of the channel is an
automatically-created mix-minus output.
In less full-featured consoles, a mix-minus can often be created with an auxiliary or “audition” bus function.
By selecting all relevant incoming sources on the bus except for the telephone fader, you can do this easily.
The following figure shows the block diagram of a single mix-minus feed being generated on a mixing
console.