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A FEW WORDS ABOUT LIVEWIRE+ AND AES67
How Livewire+ works
Livewire+ has an audio advertising system. Every source has a text name and numeric ID. These are transmitted
from source devices to the network. Devices that play audio build lists of all available sources from which users can
select.
UsingxNode audio interfaces, you enter the names of your input sources via any PC with a web browser.
Withplayout PCs attached to the network, you open a configuration window.
Livewire+ networks employ two types of audio streams.Livestreamshave small, frequent packets optimized
for live audio that requires very low (circa1 ms.) delay, for microphones and headphone audio.Standard
Streamsare also real-time streams, but with bigger packets, and are used for audio streams which don’t require
super-low latency - like audio from CD players, or that exchanged with automation system PCs. Devices that
connect to Axia networks can transmit and receive both stream types; the user selects which type to generate when
a device is initially configured.
A sophisticated phase-locked loop clocking system allows Livewire+ to use very small buffers for least latency
and ensures that audio channels remain time-aligned (as needed for multiple mics in a studio or for TV sur-
round-sound mixing.)
Converged Networks
An Ethernet network used for Livewire+ audio can also be shared with other data transmissions, such as file
transfers and web browsing. An Ethernet system with a switch at the center may have a mix of audio nodes and
normal servers, PCs, etc., because the Ethernet switch directs traffic only to where it is needed.
Even on a single link, traffic can be mixed because we use modern Ethernet’s priority mechanism to be sure audio
packets have first call on the link’s bandwidth. A studio audio delivery system can use this capability to download
an audio file from a server, for example, while simultaneously playing another audio file live.
Livewire+ maximizes the benefits of converged networking in the broadcast facility. Many stations using Livewire+
have computer data, telephone, audio, and control on a single network that uses computer industry standard
wiring, spurring cost-efficiencies throughout the plant.
Audio Quality
A Livewire+ network is a controlled, high-speed environment, with no risk of audio drop-outs from network
problems and plenty of bandwidth for many channels of high-quality uncompressed audio. We use studio-grade
48kHz/24-bit PCM encoding. Axia digital xNode audio adapters deliver 138dB of dynamic range, with less than
0.0002% THD. Even analog xNodes have 100dB dynamic range, < 0.005% THD, and headroom to +24dBu.
Livewire+ is standards-based
Since the very beginning, The Telos Alliance has based its AoIP networking technologies on standards. IP (Internet
Protocol), the networking standard that is the underpinning of nearly all critical business networks (and of the
Internet itself ) is the basis for Livewire+ AoIP.
As charter members of the AES X.192 Working Group, we helped define the AES67 standard — and became the
first broadcast manufacturer to become AES67 compliant.
Livewire+ is so standards-based, in fact, that your audio can even be played by PC media playersthat support
standard protocols and uncompressed PCM audio. The Internet’s IP standard for streaming media, called RTP/IP,
is used for standard audio streams. RTP stands for Real-Time Protocol. It’s the Internet’s standard way to transport
streaming audio and video, just as TCP/IP is the standard for general data.