Introduction to Livewire
Connect To More
IP-Audio System Design Reference & Primer
Version 3.0, May 2015
| Introduction to Livewire+
About This Manual
This document is your introduction to Livewire+. We explain the ideas that motivated it and how you can use and
benefit from it, as well as nitty-gritty details about wiring, connectors, switches, and the like. Since Livewire+ is
built on standard IP networks, we also help you to understand general network engineering so that you have the
full background for Livewire+’s fundamentals. After reading, you will know the essentials of Livewire+ along with
many of the details of the gear that vendors and the network guys that are often hanging around in radio stations
will be using.
This document covers topics common to all Livewire+ equipment and standards-based AoIP networking. It is only
a part of your full documentation package. You will also have manuals for each specific piece of equipment that
are used to build your system. From this document, for example, you will not learn how to install or operate an
Element console, but you will understand the nature of the network it plugs into.
(Permit us a small, shameless plug: Focal Press has available an excellent volume entitled Audio Over IP – Building
Pro AoIP Systems with Livewire™. This was co-authored by Telos founder Steve Church and respected technologist
Skip Pizzi. For a truly in-depth study, it can’t be beat.)
It was early in 2003 when Livewire™ was first introduced and came to market being new and fresh. Today, Broadcasters and Engineers have grasped this standards based technology to the point where you now find Livewire+
(the new, extensible protocol which incorporates and complies with the AES67 networking standard) incorporated into many other vendors’ equipment. Not to mention making Axia the fastest growing AoIP console company in
the world!
As always, we welcome your suggestions for improvement. Please feel free to contact us with your comments:
The Telos Alliance
1241 Superior Ave. E.,
Cleveland Ohio 44114 USA
Phone: +
Email: [email protected]
Table of Contents
About This Manual
A Note From The Founder of Telos
Livewire+ For Beginners
Why Ethernet? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
Compared to AES/EBU. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Audio Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
The Livewire+ Advertising System . . . . . . . . . . . . . . . . . . . . . . . 2
Control. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Livewire+ and PCs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Support for Surround . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Audio Quality. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Fidelity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
The Pac-Man of Protocols: Internet Standards. . . . . . . . . . . . . . . . . . 6
Converged Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
What Can You Do With Livewire+?
Make A Snake. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
A High-Performance Sound Card Replacement. . . . . . . . . . . . . . . . . . 7
Build An Audio Router. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Build A State-Of-The-Art Broadcast Studio. . . . . . . . . . . . . . . . . . . 8
Make a Flexible Two-Way Multi-Channel STL. . . . . . . . . . . . . . . . . . 10
Create A Facility-Wide Audio Network. . . . . . . . . . . . . . . . . . . . . 11
Create An Integrated National/Local Radio Network . . . . . . . . . . . . . . . 11
The Components Of Livewire+
xNodes. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Axia IP-Audio Driver for Windows® . . . . . . . . . . . . . . . . . . . . . . 14
iPlay (PC Router Selector). . . . . . . . . . . . . . . . . . . . . . . . . . . 16
Mixing Engines. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
StudioEngine. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
PowerStation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
QOR Mixing Engines. . . . . . . . . . . . . . . . . . . . . . . . . . . 18
| Introduction to Livewire+
Broadcast Consoles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Fusion Broadcast Console. . . . . . . . . . . . . . . . . . . . . . . . . 19
Element Broadcast Console. . . . . . . . . . . . . . . . . . . . . . . . 20
iQ and Radius Broadcast Consoles . . . . . . . . . . . . . . . . . . . . . 21
DESQ and RAQ Broadcast Consoles. . . . . . . . . . . . . . . . . . . . . 21
SoftSurface Virtual Consoles . . . . . . . . . . . . . . . . . . . . . . . . 22
PathfinderPC Routing Control Software. . . . . . . . . . . . . . . . . . . . 22
Provisions for Redundancy and Back-up. . . . . . . . . . . . . . . . . . 23
Timed Events. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Livewire+ Audio Router Control Protocol. . . . . . . . . . . . . . . . . . 23
iProbe Network Management Console. . . . . . . . . . . . . . . . . . . . . 24
iProFiler Automated Program Archiving . . . . . . . . . . . . . . . . . . . . 24
Networked Audio Processing. . . . . . . . . . . . . . . . . . . . . . . . . 24
Networked Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Networked Telephone Systems. . . . . . . . . . . . . . . . . . . . . . . . 26
Networked Web-Streaming Encoders. . . . . . . . . . . . . . . . . . . . . 27
Networked Time Management . . . . . . . . . . . . . . . . . . . . . . . . 29
Networked Intercom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Nuts & Bolts: Making Livewire+ Play
Livewire+’s Channel and Name system. . . . . . . . . . . . . . . . . . . . 30
Channels. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Text Name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
GPIO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
Sources vs. Destinations. . . . . . . . . . . . . . . . . . . . . . . . . . 31
Examples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
xNode Configuration & Web Access. . . . . . . . . . . . . . . . . . . . . . 38
Front Panel xNode Configuration . . . . . . . . . . . . . . . . . . . . . . 38
Plugs & Cables. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
CAT-5 for Audio?. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Designing & Building Your Livewire+ Ethernet System
Cabling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Twisted-pair Cable Categories. . . . . . . . . . . . . . . . . . . . . . . 47
Special Care for Ethernet Audio. . . . . . . . . . . . . . . . . . . . . . 48
To Shield or Not to Shield . . . . . . . . . . . . . . . . . . . . . . . . . 49
Unbalanced Connections. . . . . . . . . . . . . . . . . . . . . . . . . 49
More than Four Pairs in a Cable. . . . . . . . . . . . . . . . . . . . . . 49
Patch Panels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Wall Jacks. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
CAT-6 Jacks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
Architecture Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
Simple One-Switch Network . . . . . . . . . . . . . . . . . . . . . . . 50
Daisy Chained Multiple-Switch Network . . . . . . . . . . . . . . . . . . 53
Hierarchical Multiple-Switch Network . . . . . . . . . . . . . . . . . . . 54
Options for Redundancy. . . . . . . . . . . . . . . . . . . . . . . . . 54
Fiber. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
Radio Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
Designing For Security. . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
The Ethernet Switch
Livewire+ Ethernet Switch Requirements . . . . . . . . . . . . . . . . . . . 57
Some Switches We Like. . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Cisco Switches. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Axia xSwitch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Switch Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Testing, 1-2-3…
General Ethernet Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . 60
Prevention . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
The Basics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
Cable Testers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Sniffers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Diagnosing Problems using Livewire+ Components . . . . . . . . . . . . . . . 63
xNode Indicator Displays . . . . . . . . . . . . . . . . . . . . . . . . . 63
| Introduction to Livewire+
Network Engineering For Audio Engineers
Ethernet / IP Networks. . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Layering Model. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Making Packets. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
IP and Ethernet Addresses. . . . . . . . . . . . . . . . . . . . . . . . . . 67
Subnet mask. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Ethernet Switching. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Multicast . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Prioritization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
The Role of TCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
Virtual LANs (VLANs). . . . . . . . . . . . . . . . . . . . . . . . . . . 73
Ethernet Switching versus Routing . . . . . . . . . . . . . . . . . . . . . 75
Livewire+ Networks. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Quality of Service (QoS) . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Source Advertising. . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
Synchronization. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
Network Standards and Resources. . . . . . . . . . . . . . . . . . . . . 78
A Note about Protocol Design . . . . . . . . . . . . . . . . . . . . . . . . . 79
Frequently Asked Questions
General Questions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
Livewire+, Standards and Other Vendors . . . . . . . . . . . . . . . . . . . . 85
PCs and Livewire+ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
Building Livewire+ Facilities. . . . . . . . . . . . . . . . . . . . . . . . . 88
Ethernet Media. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
The Internet and Livewire+ . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Networked Consoles. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Analog Audio & AES On RJs And CAT-5. . . . . . . . . . . . . . . . . . . . . 93
Livewire+/Broadcast . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
Ethernet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
General Networking and Interest . . . . . . . . . . . . . . . . . . . . . . . 95
Cabling Information & Standards. . . . . . . . . . . . . . . . . . . . . . . 96
Cable and Contractor Supplies. . . . . . . . . . . . . . . . . . . . . . . . . 96
Cable Testers. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Ethernet Switch Vendors . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Network “Sniffers”. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Ethernet Radio Equipment. . . . . . . . . . . . . . . . . . . . . . . . . . 97
Appendix A: Livewire+ Tech Details
Livewire+ Packet Structures. . . . . . . . . . . . . . . . . . . . . . . . . 98
Standard Streams. . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Livestream . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Surround Streams . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Network Link Capacity. . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Multicast Address Translation. . . . . . . . . . . . . . . . . . . . . . . . 100
A Note From The Founder Of Telos
It’s been a tradition since Telos’ very first product, the Telos 10 digital phone system, that I share a few words with you at the
beginning of each manual. So here goes. In radio broadcast studios we’re still picking up the pieces that have fallen out from the
digital audio revolution. We’re not using cart machines anymore because PCs are so clearly a better way to store and play audio.
We’re replacing our analog mixing consoles with digital ones and routing audio digitally. But we’re still using decades-old
analog or primitive digital methods to connect our gear. Livewire+ has been developed by Telos to provide a modern PC and
computer network-oriented way to connect and distribute professional audio around a broadcast studio facility.
Your question may be, “Why Telos? Don’t you guys make phone stuff?” Yes, we certainly do. But we’ve always been attracted to
new and better ways to make things happen in radio facilities. And we’ve always looked for opportunities to make networks of
all kinds work for broadcasters.
When DSP was first possible, we used it to fix the ages-old phone hybrid problem. It was the first use of DSP in radio broadcasting. When ISDN and MP3 first happened, we saw the possibility to make a truly useful codec. We were the first to license and
use MP3 and the first to incorporate ISDN into a codec. We were active in the early days of internet audio, and the first to use
MP3 on the internet.
Inventing and adapting new technologies for broadcast is what we’ve always been about. And we’ve always been marrying
audio with networks. It’s been our passion right from the start. In our genes, if you will. As a pioneer in broadcast digital audio
and DSP, we’ve grown an R&D team with a lot of creative guys who are open-eyed to new ideas. So it’s actually quite natural
that we would be playing marriage broker to computer networks and studio audio. What you get from this is nearly as hot as a
couple on their wedding night: On one RJ-45, two-way multiple audio channels, sophisticated control and data capability, and
built-in computer compatibility.
You can use Livewire+ as a simple soundcard replacement – an audio interface connecting to a PC with an RJ-45 cable. But add
an Ethernet switch and more interfaces to build a system with as many inputs and outputs as you want. Audio may be routed
directly from interface to interface or to other PCs, so you now have an audio routing system that does everything a traditional
“mainframe” audio router does – but at a lot lower cost and with a lot more capability. Add real-time mixing/processing
engines and control surfaces and you have a modern studio facility with many advantages over the old ways of doing things.
Ok, maybe this is not as thrilling as a wedding night – perhaps kissing your first lover is a better analogy. (By the way, and way
off-topic, did you know that the person you were kissing was 72.8% water?)
While were on the subject of history… you’ve probably been soldering XLRs for a long time, so you feel a bit, shall we say, “attached” to them. We understand. But no problem – you’ll be needing them for microphones for a long while, so your withdrawal symptoms won’t be serious. But your facility already has plenty of Ethernet and plenty of computers, so you probably already
know your way around an RJ-45 as well. It’s really not that strange to imagine live audio flowing over computer networks, and
there’s little question that you are going to be seeing a lot of it in the coming years.
The 20th century was remarkable for its tremendous innovation in machines of all kinds: power generators, heating and air
conditioning, cars, airplanes, factory automation, radio, TV, computers. At the dawn of the 21st, it’s clear that the ongoing
digitization and networking of text, audio, and images will be a main technology story for decades to come, and an exciting ride
for those of us fortunate to be in the thick of it.
Speaking of years, it has been a lot of them since I wrote the Zephyr manual intro, and even more since the Telos 10. Amazing
thing is, with all the change around us, we’re still here and Telos is still growing in new ways. As, no doubt, are you and your
Steve Church
Livewire+ For Beginners
Livewire+ offers a revolutionary change in how studios can be built. But at the same time, it’s a natural continuation
of general trends and what you already know. This section explains the basics and puts audio over Ethernet into
Within the next few years, it is certain that the transition to digital now happening in our studios will be complete,
with all audio storage, mixing, processing and routing being digital. This transition is now in full swing. But
the sheer number of competing methods for integrating computers, legacy hardware and state-of-the-art audio
devices is (and long has been) bewildering. What we need is a connection method that gets the interconnection
job done easily, effectively, flexibly, and cheaply. So why not look to the computer and telephone worlds to find the
technology? We can then take advantage of the huge manufacturing scale in those industries and can piggyback on
the billions of dollars (and Euros, Yen, Yuan…) of R&D going on in those industries.
Why Ethernet?
Ethernet makes overwhelming sense. Today’s computers are near universally linked via Ethernet – and telephony
is moving that way as well, with VoIP rapidly and continually gaining market share. Even remote controlled stage
lighting is transitioning from the XLR-based DMX protocol to Ethernet. Ethernet cables, plugs, cards, and chips are
produced in the hundreds of millions so we get tremendous economy of scale. We get patch bays and cords, testers,
and all kinds of “structured wiring” components ready-made. Plugs are easy to install and jacks are efficiently small.
But much more important is that Ethernet allows us to combine many
channels of digital audio with whatever data transmission we might need on
a single cable. This data could be as simple as a start command for an audio
player or could be anything that computers and Ethernet do, such as file
transfer, e-mail, web communication, etc.
Further, we are in the line of future development. Since its invention over
30 years ago, Ethernet has been constantly evolving. It started as a 2Mbps
shared bus over coaxial cable and has grown to today’s modern (and very
common) 1 Gigabit star and switched system. 10 Gigabit is already widely
available on many Ethernet devices and is following the usual curve to low
cost as volumes increase. While copper is the most common Ethernet connection, fiber is popular as well and media converters allow the two to be interconnected. Ethernet switches cost
$6000 for 8 ports a half-decade ago; now high-end 24-port switches cost $500. And they include many advanced
features that were unheard of only a few years back.
There are radio links in many varieties, from WiFi for short-range to sophisticated long-range systems. There are
satellite links. And LASER links. Ethernet opens the door to a world of options.
Ethernet has proven to be the PC of networking: Initially released with only basic capability – low speed and bussed
– it has been expanded to today’s fast, flexible, switched architectures.
| Section 1
The combination of huge R&D expenditures, open standards, massive economies of scale, technological evolution,
and flexible multi-service packet design is hard to beat. Not to mention the surprisingly appropriate name:
Ethernet was named by its inventor, Robert Metcalf. He had been involved in a radio data network in Hawaii called
ALOHA. The first Ethernet was a bussed coax that carried data packets similar to the way ALOHA had sent them
over the “ether.” As to the origin of ether… for many years after James Clerk Maxwell’s discovery that a wave
equation could describe electromagnetic radiation, the aluminiferous ether was thought to be an omnipresent
substance capable of carrying electromagnetic waves. In 1887 scientists Albert Michelson and Edward Morley
disproved its existence. The ingenious experiment that did so was performed at Case Western Reserve University,
just down the street from the Telos Alliance main office in Cleveland.
Compared to AES/EBU
For digital audio transport, AES3 is the main alternative to an Ethernet based system. Invented in the days of 300baud modems, it was the first practical answer to connecting digital audio signals. But it’s now 30 years old and is
showing its age. Compared to Livewire+’s computer-friendly, two-way, multi-channel + high-speed data capability,
AES3 looks pretty feeble with its 2-channel and one-way only limitation. Not to mention 50-year old soldered XLR
connectors and lack of significant data capacity. AES3 is a low-volume backwater, with no computer or telephone
industry R&D driving costs down and technology forward. Your 300-baud modem has been long retired; it’s well
time to progress to the modern world for studio audio connections as well.
That said, AES and Livewire+ may comfortably co-exist in your facility. You can use Axia interface nodes to connect
from one to the other. If you are using a house sync system for AES, Livewire+ may be synced to that system also.
Audio Routing
Low-cost mass-market Ethernet switches offer us something very interesting: Since their function is to direct packets from port-to-port, we can use them to move our audio signals from whatever source to whatever destinations
we want. This means we get a simple, flexible, facility-wide audio routing system, almost for free. Say goodbye to
racks of distribution amps or expensive proprietary mainframe audio routers.
An audio source entered into the system from any point becomes available for any number of receiving destinations.
The Livewire+ Advertising System
Livewire+ has an audio advertising system. Every source has a text name and numeric ID. These are transmitted
from source devices to the network. Receivers can build lists of all available sources from which users can select.
Configuration is made simple with hardware nodes; you enter the names, numbers, and other configuration
information through a web browser via an attached PC. With PC nodes, it is even easier, requiring only the opening
of a configuration window to make all necessary source parameter changes.
Most audio these days needs associated control. A delivery system needs a start input at minimum, but could well
benefit from a richer control dialogue such as text identifying what is playing that can be sent to the studio mixer
and to the HD Radio and RDS encoders. Satellite receivers have control outputs. Telephone systems need dialing,
line status, hold, transfer, etc. Even a simple CD player needs ready indication out and start in. Even the simplest
source, a microphone, needs to convey on/off status for the air lights. Most conventional controls have been done
with primitive GPIO parallel “contact closures.”
As a first step, Ethernet can transport GPIO data, reducing and simplifying cabling, and Livewire+ offers this basic
capability to replicate traditional start/stop control. But it continues from there; Livewire+ also supports sophisticated remote operation of studio equipment over Ethernet which allows the network to transport much more
advanced information than just simple start commands.
For instance, we can send the song title from a delivery system to a display on a mixing console’s fader channel.
Control of telephone systems and codecs can follow fader assignment and be accessible from any location. With
a high-bandwidth network linking everything and a flexible communication protocol, the door is open to many
interesting possibilities. Why couldn’t the satellite receiver identify its content with “metadata” tags? Then an
automatic system could store a program along with the information about it for later play. An on-air audio processor might respond to program type information to adjust its parameters. Microphones switched-on could activate
a logger. And these have all been accomplished with Livewire+ products from the Telos Alliance and our many
partners; there are still many possibilities yet to be explored.
Livewire+ and PCs
One of the advantages of a Livewire+ system is that PC-based audio may be directly connected to the network
without soundcards. This means radio station delivery systems can use the Ethernet connection they already have
to send and receive audio. Soundcard problems such as noise and multiple conversions are avoided – the audio
remains in digital form from the PC’s files to the network with no alteration or degradation. Received audio may
have originated from another PC or from a hardware audio node. Audio sent from a PC may be received by other
PCs or hardware nodes.
With nearly all audio in today’s radio stations being either played from computers or recorded into computers, isn’t
this a tremendous advantage? Not only do you save the soundcard, but also the port that it needs at the other end to
connect to your console or router. And you can pass control and other information over the same connection.
Support for Surround
Surround sound is a de facto standard in the home entertainment industry today. Blu-Ray, Broadcast TV, Streaming
Services, and even video games all contain surround audio on a large percentage of content.
The introduction of surround into radio
broadcast has also been made possible by
advances in multi-channel codec technology
making surround a possibility over European
DAB channels and the USA HD Radio system.
Continued migration to digital streaming
services originating from computers makes
the surround Internet broadcasts a possibility
as well.
A large number of home-entertainment
systems with surround sound are in use on a
daily basis, and many high-end cars also offer
surround audio systems. If you think about
it, the car is a great environment for enjoying
surround sound since speakers are installed all
around the vehicle to start with.
A surround program typically contains 5.1-channels: 2 front, 2 surround, 1 center, and 1 subwoofer. From a
strict channel count, this is really 6 channels. The “.1” channel is named that since it only contains 1/10th of the
bandwidth that the other, full-range channels do. It might also be required to keep a stereo version of the program
(called a downmix if it is derived from the original surround mix) with the 5.1-channel program. Sometimes
the surround program might be 7.1-channels, adding 2 more surround channels. So there could be 8 total audio
channels required. In TV broadcasting, this is quite common. Using a traditional approach, that’s a lot of plugs,
cables, router cards, and rack space!
| Section 1
On the other hand, Ethernet has plenty of bandwidth to carry the multiple channels surround broadcasting
requires. All eight channels plus associated control can easily be conveyed on one convenient Ethernet cable.
Expanding the audio paths and equipment (consoles, routers, etc.) in a hard-wired facility from a stereo
infrastructure to eight channels (or more!) would be either difficult or very expensive. Not so with Ethernet and
Livewire+. In fact, there is no additional cost for the core Ethernet switch because the one you need for stereo
would also be fine for surround, and audio from PCs can be multichannel.
We have designed Livewire+ with the future well in mind. As of this writing, Livewire+ already provides the
infrastructure for over 5,500 modern radio studios. By this simple fact, these facilities are also outfitted with
surround capability – ready to implement simply and with low cost.
Audio Quality
We’re always asked, “Is Livewire+ like audio on the Internet?” Yes and no. While Livewire+ uses internet transport
standards, it is intended to operate over switched Local Area Networks (LANs). Without the limitations of the
public internet and with 100% control over all parts of the system, we are able to achieve full studio quality.
So now the question is, “Is Livewire+, audio on the Internet and Audio over IP reliable?” The answer is that Axia
uses the same technology that underlies VoIP telephony.
Did you know that more than 70% of the Fortune 100 companies now use VoIP? Or that VoIP PBX systems now
outsell the old kind by a wide margin? With these systems, telephones plug into a standard Ethernet/IP network.
Contrast this with traditional PBX phone gear — proprietary devices which required you to purchase phone sets
and parts exclusively from the company that built the mainframe. You were locked into a single vendor, because the
technology that ran the mainframe was owned by the company that made the gear (kind of like the TDM router
IP is growing as a universal transport for almost any kind of signal. You see it now in television studios, business
teleconferencing, government communications, banking, etc. And it’s hardly unproven, even for applications
specific to broadcast studio infrastructure. There are plenty of people successfully using it – now.
Internet streams are usually compressed for transmission over public links with limited, variable bandwidth and
low reliability. Livewire+ audio is not compressed – we use studio-grade 48kHz/24-bit PCM encoding. Our audio
interface nodes have more than 100dB dynamic range, < 0.005% THD, and headroom to +24dBu. LANs offer a safe,
controlled environment where there is no risk of audio drop-outs from network problems and plenty of bandwidth
for many channels of high-quality audio without compression.
Indeed, we often hear from Livewire+ users that they notice an improvement in fidelity when they transition from
other systems. This is probably due to the direct connection of PCs, the 64-bit accumulator in the Axia mixing
engine, and to the careful design of the audio stages of our audio interface nodes, rather than the network itself.
But, in any event, the network takes nothing away from audio performance.
In packet-based systems, delay is an important issue and certainly has an effect on your talent’s perception of
“quality.” Packetizing audio for network transmission necessarily causes delay, and careful design of the system
is required to reduce this to acceptable levels. Internet audio delay is often multiple seconds because the receiving
PCs need long buffers to ride out network problems and the delays inherent in multiple-hop router paths. However,
with fast Ethernet switching on a local network, it is possible to achieve very low delay. To do this, we must have a
synchronization system throughout the network. This also avoids sample or packet slips that cause audio dropouts.
Internet streaming does not use this technique, so even if it were to have guaranteed reliable bandwidth, you still
couldn’t achieve the very low delay we need for professional studio applications.
For Livewire+, we generate a system-wide synchronization clock that is used by all nodes. Within each node, a
carefully-designed PLL system recovers the synchronization reliably, even in the case of network congestion.
Hardware nodes provide this clock and in each system, there is one master node which sends the clock signal to the
network. If it should be disconnected, or stop sending the clock for any reason, another node automatically and
seamlessly takes over. Or, should you wish, a Livewire+ network can easily be synchronized to input provided by an
external “house clock” or the IEEE-1588 clock required for AES67 signals.
In broadcast studios we care very much about audio delay in the microphone-to-headphones path for live
announcers. Maximum delay must be held to around 10ms or announcers will start to complain of comb-filter or
echo problems. We need to consider that this is a total “delay budget” and that multiple links and some processing
will often be in the path. So we’ve decided to have a link delay around 1ms end-to-end for anything in this path,
allowing us a few links or maybe a couple of links and a processor before we get into links: one from the mic node to
the mix engine and one from the engine to the headphones out node. Thus, 2ms total.
1-3 ms
3-10 ms
Audible shift in voice character (comb filter effect)
10-30 ms
A slight echo turning to obvious slap at 25-30ms
30-50 ms
Disturbing echo, disorienting the announcer
> 50 ms
Too much delay for live monitoring
Here are the air-talent reactions to delay in a test conducted by Jeff Goode at WFMS in Indianapolis
In our experience, delays to around 10ms are not a problem. From 10-25ms announcers are annoyed but can work
live; anything above 25-30ms is unacceptable.
Another way to think about delay: Audio traveling 1 foot (0.3 meters) in air takes about 1ms to go this distance.
And another data point: A common professional A-to-D or D-to-A converter has about .75ms delay.
But, as is universally the case in engineering, there is a tradeoff – otherwise known as the “if you want the rainbow,
you gotta put up with the rain” principle. To have low delay in a packet network, we need to send streams with
small packets, each containing only a few accumulated samples, and send them at a rapid rate. Bigger packets
would be more efficient because there would be fewer of them and they would come at slower rate. But they would
require longer buffers and thus impose more delay. Big packets would also have the advantage that the necessary
packet header overhead would be applied to more samples, which would more effectively use network bandwidth.
With Livewire+, we enjoy our rainbow and avoid the rain by having different stream types: Livestreams use small
and more frequent packets, while Standard Streams have bigger and less frequent packets.
Livestreams require dedicated hardware and achieve the required very low delay for microphone-to-headphone
paths. PCs are not able to handle these small packets flying by so quickly, therefore they use the Standard Streams.
These are compatible with Internet standards and can be directly received into the network from PCs running
delivery software. The network delay in this case is around 5ms and the PC’s latency is likely to add perhaps
50-100ms more. Since PCs are playing the files and they are not in live paths needing Livestreams, this is not a
problem. Our only concern is how long it takes audio to start after pressing the On button, and delays in this range
are acceptable.
Standard Streams can also be sent from the network to PCs for listening and recording. Again, this small delay is
not an issue – especially given that PC media players have multiple seconds of buffering. However, off-the-shelf
PC hardware with a special operating system and software optimized for real time audio is able to handle the fast
streams. Indeed, we use this approach for our studio mixing and processing engine.
| Section 1
All Livewire+ hardware devices transmit both stream types and can receive both stream types. There is no
inefficiency from having both available because all streams stop at the Ethernet switch and take no system network
bandwidth unless they are subscribed to by a receiver or node. Each receiver takes only the one it needs, taking the
low-delay version if available, or the higher-delay version if not. The selection happens transparently with no user
action needed; users just select the channel they want and audio is delivered by whichever method is appropriate
for the equipment they are using.
Livewire+’s low-delay streams are also fixed-delay. The delay is constant, regardless of the system size or anything
else. In fact, a source being received at multiple Nodes will have a differential delay of less than 5µs - less than 1/4
sample at Livewire+’s 48kHz rate.
The Pac-Man of Protocols: Internet Standards
We use the Internet’s IP standard for streaming media called RTP/IP for Standard Streams. RTP stands for
Real-Time Protocol. It is the internet’s standard way to transport streaming audio and video, just as TCP/IP is the
standard for general data. Both use the same underlying IP packet structure, but each has a header and transmission
method appropriate to the content.
Since we adhere to Internet standards, your audio may be played by PC players such as Windows Media and VLC,
which support standard protocols and uncompressed PCM audio.
Converged Networks
In the telephone and networking worlds, IP has become the “Pac-Man” of protocols, eating up everything in sight.
Major networking companies like Cisco and HP are dedicated to the idea that a facility needs only one network
for data, telephones, and media. Many companies have jumped on this idea supporting the products that these
companies are building today.
Meanwhile, PBX companies like Alcatel-Lucent, Nortel, Mitel, AT&T and Siemens have plunged into IP transport
for their telephone products. This is bringing converged networks, serving all needs from PBX companies too.
Skepticism from a few years ago, which questioned whether Ethernet would handle convergence with services like
telephone and live media while sharing the network with computers, has been patently disproved since then. (A
network technology called ATM was proposed as a better solution. But it was expensive, difficult to administer,
and would have required a “fork-lift” upgrade to existing systems. So it never caught on and has pretty much faded
from sight for local networks, although it has a role at the core of some Telco networks.) Ethernet’s switching,
priority mechanisms, and increasingly fast speed has put most concerns to rest, and all the vendors who offer VoIP
telephones connect them over Ethernet, not ATM.
Ethernet might just as well be said to be the Pac-Man of local networks too. It has nearly a 100% share of new LAN
installations and is the network that all VoIP phone systems we know about use for connection to the desktop.
An Ethernet network being used for Livewire+ audio may be shared with any other data transmissions such as file
transfers, web browsing, and the like. An Ethernet system with a switch at the center may have a mix of audio nodes
and normal servers, PCs, et cetera. 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 could use this capability, for example, to download an
audio file from a server while simultaneously playing another, live.
Livewire+ adds to the convergence possibilities in a broadcast facility. In fact, nearly all new broadcast facilities
today have computer data, telephone, audio, and control on a single network that uses computer/telephone
industry standard wiring; many of our customers have had this benefit for years.
What Can You Do With Livewire+?
Imagine everything that you can do with a PC connected to a network: Share files, send and receive emails, chat,
surf the web, listen to audio, etc., etc.. PCs and networks are designed to be general-purpose enablers. You have a
similarly wide range of possibilities for audio applications using Livewire+. Here are examples, starting with the
most simple, and continuing to the most interesting.
Make A Snake
Concert sound guys need to get a lot of audio from the stage to their mixing consoles in the center of the house.
They call the multi-conductor cables they traditionally use for this function a “snake”. Livewire+ lets you put
such a snake on a diet! A single Ethernet cable connects multiple audio channels. Add a switch at each end and
you can have as many nodes as you want. Use Gigabit Ethernet and you can have hundreds of channels. Add fiber
optic media converters and cable to extend the distance between units to many kilometers. Maybe you need to get
something from here to there?
A High-Performance Sound Card Replacement
Livewire+ can talk directly to PCs, making the network look like a soundcard to delivery systems, editors, etc. Axia
xNodes have excellent audio performance: Balanced I/O with more than 100dB dynamic range, < 0.005% distortion, headroom to +24dBu, etc. They make excellent multi-channel “soundcards” for professional applications. You
can position the xNode at a distance from the PC, and you get balanced audio on connectors that are a lot more
reliable than mini phone jacks.
With the addition of an Ethernet switch you can feed your audio to multiple computers and/or have multiple I/O
boxes – which take us to the next application…
| Section 2
Build An Audio Router
A system with Livewire+ nodes, one or more Ethernet switches, and PC-based routing controller software make an
excellent facility-wide audio router. PCs send and receive audio directly to the network without soundcards or audio ports, thus lowering costs and eliminating conversion steps. telephone, codec, streaming encoders, time-management solutions and audio processing equipment from Telos Alliance brands (Telos, Omnia, 25-Seven, Axia and
Linear Acoustic) is now able to connect directly, as are numerous products from other broadcast manufacturers
such as International Datacasting, AudioScience, RCS, Digigram and BSI, to name a few (for a complete list, see
To interface conventional analog and AES signals, Livewire+ interface nodes come in a number of versions. One
(called xSelector) operates like a traditional audio router X-Y control panel, but with a difference: audio in and out
is available on the same box.
A PC-based router control package, called PathfinderPC, is available that makes your whole system, with all of its
disparate parts, capable of being controlled and supervised as a single entity. You can control which outputs are
connected to which inputs just as if the system were a single-location box.
Since there is no requirement for a mainframe, the base cost is low – you can make a small system at very reasonable
cost and expand it over time. Indeed, the total cost of a large system will be much lower then older approaches due
to the use of commodity switches at the core. Just as using standard PCs to play audio makes much more sense than
any proprietary approach, building routers from common computer industry parts makes similar sense. Indeed,
this approach gives you a true “audio network” quite unlike other approaches.
Build A State-Of-The-Art Broadcast Studio
Plug an audio processing engine and a control surface into the network and you have a modern radio studio with
many advantages over the old way:
♦♦ Simplified and unified cabling for audio, control, general data, and telephone.
♦♦ No multiple conversions. With most studio audio coming from or going to PCs, audio is kept in the
networked digital domain. Audio may be monitored on any PC with a player such as Windows Media,
VLC player, etc.
♦♦ Integrated data means you are ready for synchronized text and metadata, which is needed for HD-Radio
in the USA and DAB in Europe, as well as for streaming audio channels. It will also be possible for audio
processor parameters to be controlled by metadata, depending upon source characteristics.
♦♦ Tighter integration with delivery systems means that mixing, scheduling, and playing can work together.
For example, song titles can appear on the mixer surface, start and other control functions may be
conveyed over the network, and logging can confirm that an audio piece was really played on the air.
♦♦ Troubleshooting and repair are transformed. Extensive diagnostics are available over the same network
that connects the audio. A suspect surface or engine may be swapped by re-plugging only one Ethernet
♦♦ Low-cost power. Computers replaced cart machines because they are a lot more powerful, convenient,
reliable, and cheap. The technical side of radio broadcasting is tiny compared the computer and networking industries. We get tremendous value by plugging into the massive R&D and production scale offered
by the computer world. Leveraging low-cost mass-produced computer components makes the same
sense for studio mixing and audio distribution as it did for cart machine replacement.
♦♦ Surround-ready. As one would expect from its flexible computer technology-based origins, Livewire
easily handles common audio formats such as surround sound, which is quite prevalent in television.
In the example below, a Livewire+-based system is being used as a studio console. Sources such as microphones and
CD players are interfaced to the network with a node in the studio, while sources such as network feeds interface
with a node in an equipment room.
Certain peripheral equipment connects directly to the network. Audio from the delivery PC goes to the network via
an Ethernet connection and control is also over the network. Telos Zephyrs or Telos telephone hybrids can connect
directly to the switch to make the audio and control connections much simpler, only one RJ-45. The network also
supports file transfers to the delivery system from a server. The studio operator surface controls a rack-mount mix
engine, which has a single Ethernet connection for both control and audio.
10 | Section 2
Make a Flexible Two-Way Multi-Channel STL
Studio and transmitter sites may be linked with “Ethernet STLs”. Livewire+ nodes provide audio interface to
Ethernet point-to-point radios. These are off-the-shelf today from such companies as Dragonwave, Cambium,
Ubiquiti and many more. In addition to the audio, anything that can be carried over Ethernet can be conveyed over
the radio link, such as VoIP telephones, email, file transfers, and transmitter remote control.
Ethernet radio systems are available that can connect at speeds of 100Mbps and better. This data rate would support more then a dozen stereo uncompressed audio channels in each direction, with capacity remaining for VoIP
telephone and facilities control. Seems these radios would make an interesting two-way RPU also. For co-owned
stations that are not co-located, these could be an effective way to link studio facilities.
Many of these systems are optimized for speed vs low error rate and therefore may not work. Axia
has evaluated several units and we can offer guidance if you are interested in pursuing this option.
Create A Facility-Wide Audio Network
That Includes Integrated Studio Consoles
Combine all of the above for maximum power, convenience, and flexibility. You get facility-wide audio routing,
state-of-the-art studio mixing, a single wiring infrastructure for audio, computer data, control, and telephone.
Audio processors with Livewire+ ports may easily have multi-channel outputs, such as for simultaneous analog
FM, HD Radio, and low-delay monitoring feeds. A single Ethernet would serve for all needed inputs and outputs.
With a data capability alongside the audio, it would be possible to control processing parameters depending upon
which audio source is active.
Create An Integrated National/Local Radio Network
Imagine a satellite transmitting IP packets. Now live audio, audio can be stored for later play, and identifying data
can be delivered. Wouldn’t this transform radio networks into something much more interesting, useful, and
powerful? Including an Internet-based return path adds a whole new dimension.
The Components Of Livewire+
Livewire+ is not only a technology. It is a solution, made for broadcast. Here are the essential pieces that put
Livewire+ to work for you.
A Livewire+ system usually has a mix of hardware nodes and PCs with driver software that lets them send and
receive Livewire+ audio streams. There will also be one or more Ethernet switches, unless you are making only
a very simple 2-box snake or a PC soundcard replacement. This section gives an overview; currently available
switches are covered in a later section.
Many equipment manufacturers are offering equipment with Livewire+ onboard, and we expect many more in
the future. Of course, this capability is a part of most new Telos Alliance products, meaning you can now get Telos
telephone hybrids and systems, Z/IP ONE IP codecs, Omnia and Linear Acoustic audio processors, 25-Seven time
management gear and Z/IPStream web-streaming equipment with Livewire+ connectivity.
These interface traditional audio to the Livewire+ network. Some are used to interface GPIO such as for starting
CD players or lighting on-air signs. Configuration and monitoring is via a networked web browser.
Analog xNode: Four stereo / eight mono balanced inputs and outputs with more than 100dB dynamic range, <
0.005% distortion, headroom to +24dBu. Software controlled gain lets you trim adjust to accommodate different
levels. Front panel OLED audio level metering, can be PoE-powered.
AES/EBU xNode: Four AES3 inputs and outputs. An input can be used to sync your Livewire+ network to your
house AES clock, if desired.
Mic + Line xNode: Four microphone inputs with very high-grade pre-amps, Phantom power, and four stereo /
eight mono balanced line outputs. Like the other xNodes, it can also run on PoE connections.
Mixed-Signal xNode: This multi-purpose xNode provides one switchable Mic/Line analog input, two analog line
inputs (dedicated), 3 analog line outputs, 1 digital AES3 input and 1 AES3 output, and 2 GPIO ports.
General Purpose Input/Output xNode: This GPIO interface for parallel closures has eight DB-15 connectors, each
with five inputs and five outputs. Interfaces control to CD players, delivery systems, on-air lights, etc. that need
simple parallel control. Many Axia mixing engines, such as the PowerStation and QOR mix engine families, also
offers identical GPIO functionality in addition to audio I/O and Ethernet switch ports.
SDI xNode: Essential for using AoIP networks in TV plants. Supports up to two separate, independent SDI paths;
de-embeds up to 16-channels of audio (8 stereo pairs) from the incoming SDI streams and makes that audio
available to the Livewire+ network. Additionally, it is possible to take audio from the incoming streams, shuffle
pairs, and create two unique outgoing SDI streams with matched audio/video latency.
14 | Section 3
X2 Router Node: Emulates the function of traditional X-Y audio router controllers. The OLED on the left presents
a list of available audio sources, which are selected with the adjacent knob. The OLED on the right presents a list
of possible output destinations. The unit can be used for equipment room monitoring and production studio or
newsroom audio interface.
xSelector Node: An X-Y router controller with a twist: it has its own pair of local inputs and outputs. Not only can
it be used to choose any network audio stream and route it to the output destination, xSelector can also contribute
audio streams to the network. Many users find these indispensable in NOCs, terminal rooms or Master Control
racks; they are also an excellent way to outfit news bullpens or other audio contribution stations where only a single
input and output are needed.
There are also a number of switch panels that work with Axia PathfinderPC routing control software to enable
routing scene changes with a single button press.
Axia IP-Audio Driver for Windows®
This is the software interface between your PC audio applications and the Livewire+ network.
24-in/24-out Driver: This is a driver that interfaces 24 stereo inputs and outputs and is available through one of our
delivery system partners. It provides these functions:
♦♦ Interface for audio sent to Livewire+ from audio applications such as delivery systems and
other audio players.
♦♦ Interface to receive audio from Livewire+ into applications such as audio editors.
♦♦ A GPIO function to convey “button-press’ data from the network to applications, such as from a control
surface fader start button to an audio player.
8-in/8-out Driver: This is an eight-input/output version of the OEM Driver described above, available from our
delivery system partners.
4-in/4-out Driver: This is a four-input/output version of the Driver, available for direct purchase from any Axia
distributor. Like its OEM cousins, it works with any delivery system and editor that support standard Windows
audio. It provides these functions:
♦♦ The Axia Windows® Driver connects PC audio directly to the network via Ethernet and
without sound cards.
♦♦ Provides four stereo inputs and four stereo outputs via the Livewire+ network.
♦♦ Audio applications see the Livewire+ network as if it were a standard sound card. A sample rate converter
and clock generation functions are included.
1-in/1-out Driver: A single-channel version of the retail 4-in/4-out Driver described above, ideal for workstations
or audio capture/editing applications.
If you have a Linux plant, a Linux version of the Driver is available for specific OS distros from our partner, Paravel
16 | Section 3
iPlay (PC Router Selector)
The next application in the Livewire+ Windows Suite is an interface to display and select Livewire+ streams –
essentially a software version of the Router Selector. The selected audio is sent to any audio application that works
with standard Windows sound cards. The Preview function lets you listen directly without another application.
Sources are listed for selection with a mouse click. They may be filtered by category.
There is a capability similar to the radio buttons on the hardware Router Selector. Dragging a listed source to one of
the buttons allows it to be used to quickly select a desired source.
In addition, Livewire+ streams are based on standards and can be played with standard Internet audio players such
as Microsoft Windows Media and VLC players, with the use of SDP files.
Mixing Engines
With all audio sources in your facility available on a single Ethernet jack, the door is open to new ways of mixing
and processing audio signals. We are now able to build a low-cost, but very powerful mixing/processing engine that
subscribes to networked audio streams, modifies them and presents the resulting streams back to the network on
that same jack. One mixing engine is required for each console mixing surface.
The Axia StudioEngine is a powerful stand-alone processor for Axia Fusion and Element mixing consoles. It has no
I/O of its own, taking all audio from the Livewire+ network via Gigabit Ethernet and performing internal mixing,
with mixed output sent back via the same Livewire+ port. The StudioEngine performs all the mixing and signal
processing functions that would have been performed in the past by an audio console. Of course, a Livewire+-based
routing system may be used with any traditional console, but integration brings many advantages.
Each StudioEngine can perform all the mixing and processing functions needed by even the largest console, with
per-channel mix-minus feeds, multiple outputs and monitor feeds, EQ, mic processing, et cetera. There’s plenty of
headroom to support future features.
The front panel display on the StudioEngine provides confidence feedback. The selector knob allows you to easily
perform basic configuration. As with all Livewire+ components, web-based interaction is used for more advanced
PowerStation is an “integrated”, all-in-one mixing engine for Axia Fusion and Element mixing consoles. It includes
audio I/O, GPIO, a console power supply, and a zero-configuration Livewire+ network switch into one easy-todeploy package.
18 | Section 3
There are two PowerStation models, a Main and an Aux. Each PowerStation Main provides:
♦♦ 4 Analog inputs and 6 Analog outputs
♦♦ 2 AES/EBU inputs and 2 AES/EBU outputs
♦♦ 2 Microphone inputs with selectable Phantom power
♦♦ 4 GPIO machine-control logic ports, each with 5 inputs and 5 outputs
♦♦ An integrated network switch with 14 100Base-T Ethernet ports and 2 1000Base-T (Gigabit) ports with SFP
♦♦ Heavy-duty power supply
♦♦ Industrial CPU designed for harsh-environment reliability
Simple Networking built in to the PowerStation switch allows daisy-chain connection of up to 4 PowerStation-based studios without the use of an external core switch. Connecting a PowerStation Aux adds auto-switching,
redundant backup power — as well as doubling audio I/O and GPIO capacity. If you require more I/O, xNodes can
be added at will to expand capacity.
QOR Mixing Engines
Like the PowerStation integrated engine above, QOR mixing engines are self-contained units with DSP mixing
engine, audio and logic I/O ports, and a network switch with Livewire+ ports and Gigabit trunking. There are
two models, which work with Axia iQ, Radius, DESQ and RAQ mixing consoles. If more I/O than that built-in is
required, xNodes can be added to expand those capabilities. Like the larger Axia mixing engines, these also provide
automatic mix-minus, IFB/talkback paths and source EQ.
QOR.32 is suited for use with iQ or Radius consoles of up to 24 faders (or 3 frames), but can also be used with
smaller DESQ and RAQ consoles where large amounts of local I/O are desired. It includes:
♦♦ 4 mic inputs with selectable Phantom power
♦♦ 16 Analog inputs
♦♦ 8 Analog outputs
♦♦ 2 AES/EBU inputs and outputs
♦♦ 8 GPIO ports, each with 5 inputs and 5 outputs
♦♦ Zero-configuration Ethernet switch with 6 100Base-T Livewire+ ports (4 with PoE) and 2 Gigabit
Livewire+ ports (RJ-45 & SFP).
♦♦ Redundant, auto-switching backup power can be added by connecting a QOR Backup power supply.
QOR.16 is designed for personal “desktop” mixer installations and small-studio applications. It has half the I/O
count of the QOR.32.
♦♦ 2 mic inputs with selectable Phantom power
♦♦ 8 Analog inputs
♦♦ 4 Analog outputs
♦♦ 1AES/EBU input and 1 AES/EBU output
♦♦ 4 GPIO ports
♦♦ Zero-configuration Ethernet switch with 6 100Base-T Livewire+ ports (4 with PoE), and 2 Gigabit
Livewire+ ports (RJ-45 & SFP) for connection to a larger network.
Broadcast Consoles
Of course, operators still need to have control interfaces. The Axia consoles (and virtual console software!)
described below work with the mixing engines just shown. They, too, connect with a single Ethernet plug.
Designed for the needs of live programming, Axia consoles provide your on-air staff with a familiar and comfortable set of controls in an uncluttered and intuitive format. With a lot of broadcast experience under our belts, we
worked carefully to keep the basic functions simple and trouble free, but still include all the sophisticated functions
of large traditional consoles supported in a deeper layer.
Fusion Broadcast Console
Axia’s most sophisticated broadcast console to
date, designed after more than a decade of
experience with AoIP console design. Fusion is a
modular console, with faders presented in “gangs”
of 4 per module. There are a large variety of
module types; some that directly-control Telos
phone systems, others that provide full-plant
intercom services over the same IP network upon
which all your other plant audio rides.
Metering, time, timer, and essential status info
are clearly presented on an adjacent DVI LCD
monitor of your choosing. Pressing the Option
knob on fader channels brings up all the fancy stuff. All sources in your Livewire+ system are listed and available for
selection, and there is pan, EQ, L/R select, send bus access, mic processing, headphone EQ, and more.
20 | Section 3
In addition, there are “expert user” modes that allow
engineers to configure very advanced fader/source
behaviors, based on channel-strip status. Monitor
assignments, bus assignments, even mix-minus
sources can all be altered automatically, without operator intervention, based upon channel on/off status
or the state of the source’s assignment to the Preview
bus – a power-user’s dream. No longer is the operator
saddled with the heavy lifting of complicated bus
assignments and mix-minus generation, as the
console and mixing engine relieve him or her of those
But it goes further. As you would expect from Telos and Axia, our control surfaces have a smart approach to
mix-minus for phones and codecs. Every channel has the ability to provide a mix-minus output automatically.
Operators simply select a phone or codec source and the backfeed is automatically generated based on preferences
established when the user profile was configured. There is a single button that selects a Phone Record mode when
users need to record phones off-air for later play.
High-resolution OLED displays on the overbridge above each fader show the active source, and icons indicate
status when needed, along with full-time confidence meters.
Fusion can save profiles for each user, allowing different preferences, layouts and defaults for a variety of shows and
In addition to console functions, Fusion provides controls and displays that interact with phone systems, codecs,
editors, PC delivery systems, and other broadcast gear. Together with the StudioEngine or PowerStation mixing
engines, Fusion was designed to meet all the console/control needs of the most demanding live and live-assist radio
Element Broadcast Console
Axia’s most-popular, longest-running broadcast
console, installed in more than 4,500 studios
around the world. Like the Fusion, Element is a
modular console that comes in multiple frame
sizes and supports as many as 28 faders in a
single frame or up to 40 in multiple linked
frames. Metering and status info are likewise
presented on an adjacent monitor.
Also like Fusion, every channel is equipped
with automatic mix-minus, and there is enough
horsepower in our mixing engines to provide
one for every fader in operation, were such a
situation to occur.
Clear, bright LCD displays complement each
fader strip and display active source and status
icons when needed. Show Profiles can be saved
for individual users or shows.
iQ and Radius Broadcast Consoles
Where Fusion and Element are well-suited for large
operations, iQ and Radius consoles are intended for
medium-sized control rooms. Each come with a
matched QOR mix engine (explained in the previous
section); they require no external monitor for meters,
clocks etc., as this is provided via high-rez OLED
displays (iQ) or multi-segment LED displays (Radius).
The iQ and Radius main units each contain 8 faders,
but may be expanded up to 24 faders using a combination of accessory frames which may also provide
features such as built-in multi-line phone system
control, user keys which can be programmed to execute
GPIO commands or router salvos, etc.
Although visually different the iQ and Radius are
functionally the same; you may choose whichever meter
style suits your preference. Each should be paired with a
QOR.32 mixing engine in order to provide the I/O needed
to accommodate a good-sized studio with associated audio
and GPIO connections, etc.
DESQ and RAQ Broadcast Consoles
Axia DESQ and RAQ are compact consoles designed for tight spaces, such as dubbing stations, remote road kits,
turret-based news assembly booths, or voiceover booths. Each provides two stereo mixing buses and 6 faders to
work with.
Like their bigger brothers, DESQ and RAQ feature
automatic mix-minus, source EQ and user profiles
that can recall frequently-used configurations.
They work with the same QOR mixing engines
used with iQ and Radius mixers, but with an added
benefit: a single QOR engine can power two DESQs,
two RAQs, or a combination of one of each. This
allows the construction of networked small rooms
with a very high cost-to-benefit ratio, as the cost of
equipping each mixing surface with a mix engine
drops dramatically, and since these smaller rooms
are also now part of your greater network, they can
access and use any audio source from any room
in your plant — previously a task too expensive to
accomplish for small studios.
22 | Section 3
SoftSurface Virtual Consoles
Perhaps the idea of a virtual console – a mixing engine controlled
by software only – interests you, for use in giving talent access to
mixing capabilities in round-table discussions, in-studio performances or in-person interviews. Axia SoftSurface is such a product,
with the ability to run on any Windows-based PC, laptop or
touchscreen tablet. Using SoftSurface, talent can take complete
control of a StudioEngine or PowerStation mixing engine, using it
just as they would a physical console, with access to all the same
controls and options. Imagine carrying a 40-fader console around
in your hand! Compelling, no?
PathfinderPC Routing Control Software
Distributed Livewire+ systems can be controlled just as if they were traditional centralized audio routers. In this
case, you will need a way to control the multiple nodes as if they were a single device.
We offer a PC software package called
PathfinderPC, developed in cooperation with a partner, Software
Authority, which specializes in router
controls. It is a client-server system
that serves as a front end for X-Y style
router switching. The server communicates will all of the Livewire+ nodes
in your system, and offers a common
point of control to clients. Multiple
clients can connect to the server to
provide any number of control points.
Each client may be optimized for a
particular style of operation or control features. For example, a master control client will probably be very different
from a controller within a studio.
Scenes (presets) can be created and recalled to allow changes to the local studio or to the global network. A “virtual
patch bay” function provides an intuitive way to manage routes. The server and clients run on Windows PCs.
Because Livewire+ nodes put audio level information onto the network, PathfinderPC clients are able to display
level metering. This is indicated with on-screen cross point icons – green dots indicate the presence of audio. Users
may also select accurate multi-segment meters for audio sources they want to check carefully.
You can use PathfinderPC to make “virtual routers,” which can be subsets of the real routers. For example, if a
Livewire+ system has 128 different sources and destinations on the network, but you may only wish to use a small
number of these points in a particular studio area. You can create a virtual bay that specifically includes only the
sources and destinations required by this studio. This virtual router can have its own set of “snapshots” (scene
changes). The virtual router also allows you to map multiple points to a single virtual point. For example, you can
make a virtual source and destination that contains both the audio inputs and outputs for a particular device, and
also the GPIO points. Thus when the route is made, both audio and GPIO is routed simultaneously.
PathfinderPC supports non-Livewire+ routers including video routers and machine control routers. Thus, you can
make routing points in the virtual bay which will simultaneously route audio, video, GPIO, and Machine Control.
This makes the software ideal as a master centralized router control package. Software Authority continues to
expand our list of supported products, and the software is designed to allow us to add support for additional
protocols and routers quickly.
PathfinderPC supports the use of tie lines or gateways between routers. For example if a system has both an analog
video router and an SDI video router, one or several tie lines can be wired through Analog to SDI converters
between the two routers. PathfinderPC will then combine the routing tables and automatically use the tie lines
when necessary to get analog sources to the SDI router. The complexity is hidden from the end user. This capability
allows Livewire+ terminals to extend an older and already filled router.
An even more powerful program, PathfinderPRO, allows clustering of redundant servers for large facilities.
PathfinderPRO allows the multiple servers to mirror data and ensures that, should a primary server be taken
offline, a backup server is instantly available to take its place, virtually guaranteeing 100% uptime.
Provisions for Redundancy and Back-up
PathfinderPC includes a silence detector that allows you to place a “watch” on a particular Livewire+ destination.
If the audio level falls below a specified threshold for longer than a specified period of time, the system will switch
to a backup audio source. This lets you build automatic redundancy into a signal path. If the primary and backup
sources and destinations in the silence detector are assigned to different Livewire+ units and these units are wired
to different AC power sources, the signal path can be maintained even in the event of a failure of a terminal or
power source.
In addition, multiple PathfinderPRO servers can be simultaneously monitoring the Livewire+ network, building
redundancy into the control system as well. The Livewire+ system is an ideal tool for building a redundant audio
chain. Since every audio unit is an independent device, the server can automatically switch audio to a different unit
if the usual one fails. With careful planning, you can even arrange your system so that the primary and backup audio
units are connected to different LAN switches which are chained together using the inherent Ethernet redundancy
protocols. Thus audio is continuously and reliably passed, even in the event of a LAN switch failure.
Timed Events
PathfinderPC has a simple timed-event system built into the server with which you can program events to happen
at specified times. Individual routes or snapshots (scenes) can be triggered at a particular time and date or on a
rotating schedule on certain days and times of the week. Events can also be created which will monitor a GPIO and
initiate a snapshot (scene change) or route whenever a GPIO condition changes.
For more sophisticated timed operation, external automation systems can access and manipulate the routing
tables provided by the Pathfinder server using the protocol translator. Multiple protocols may be simultaneously
translated and connection may be on either IP/Ethernet or serial ports.
Livewire+ Audio Router Control Protocol
We provide a documented protocol for those who want to develop their own controllers. Applications designed
for controlling traditional audio routers can implement Livewire+ Audio Router Protocol directly or use a software
gateway between this protocol and their native protocol. The first solution may be preferable, as it enables applications to fully control every Livewire+ unit and is free from potential problems with the gateway reliability. To avoid
multiple TCP/IP connections, the gateway solution may be used. In this case, there must be gateway/translator
software developed for each protocol that has to be supported.
Livewire+ Routing protocol assumes multiple audio input/output nodes. Every node has its unique IP address, N
input ports and M output ports.
We offer a software interface that emulates a traditional router and does the mapping and translation. Input to
this module can be either serial port or TCP/IP over a network. Network configuration of Livewire+ devices can be
communicated to this program using command line or a text configuration file. There is a TCP/IP server waiting on
every Livewire+ node. The client simply writes text commands to the appropriate device.
24 | Section 3
iProbe Network Management Console
Axia iProbe is an intelligent network maintenance and
diagnostics tool that makes managing, updating, and remote
controlling a Livewire+ system very easy. There’s an
auto-documentation feature that generates configuration
docs for every device. The organizer lets you group Audio
Nodes into logical groups for easy management, upload
software to single or multiple devices, make device configuration backups and more. The status of every device
connected to the Livewire+ network is immediately
available from a single point of control. iProbe can even
perform automatic software updates for specified devices,
querying them to determine installed software versions and auto-downloading newer releases from our servers for
immediate or delayed update.
iProbe also provides a centralized command interface for each connected network device; individual device Web
configuration pages can be quickly accessed, and individual audio sources auditioned directly.
iProFiler Automated Program Archiving
iProFiler logging software simultaneously captures up to eight stereo audio channels to time-stamped MP3 audio
logs directly from the Livewire+ network — no audio cards required. iProFiler monitors GPIO channels you select
and begins recording when those channels activate. Record mode can be set for logging, skimming, or a combination of both. Logged audio can be stored locally or remotely, and may be auditioned via LAN, WAN, or Internet, if
you choose.
Networked Audio Processing
With the addition of Livewire+ connections to the latest audio processing gear, it is possible to route program
streams directly from control room to processor without any interim conversions and in fact, without leaving
the network at all. Many Telos Alliance audio processors with Livewire+ capability are available now, with plans
for more in the works. Analog and AES/EBU I/O is provided for compatibility with legacy audio chains, but with
Livewire+ I/O, it is possible to build an audio chain that remains completely within the networked digital domain.
Omnia.11 is, perhaps, the most powerful FM and HD/DRM
audio processor available yet. Thanks to an incredibly
powerful processing platform and very sophisticated, yet
easy-to-use processing tools, Omnia.11 delivers thunderous
bottom end, sparkling highs, and crisp, clear voice reproduction. The firmware in Omnia.11 takes advantage of software
capabilities never before possible: AGCs, compressors, and
limiters analyze music in real time and adjust internal
parameters for optimum performance across a broad range
of material. Features include touchscreen front panel
controls, Ultra-Multiband Limiter System with self-adjusting attack/release, Ultra LoIMD Distortion Controlled Clipper, Single Sideband Suppressed Carrier technology for
reduced multipath distortion and many more audio-shaping tools.
Linear Acoustic AERO.soft provides enterprise-wide
audio and loudness management, deployed on one or
more AERO.soft servers. As AoIP is being adopted at an
ever-increasing rate in TV facilities, AERO.soft
connects via Livewire+ to provide engineers with
trusted Linear Acoustic AEROMAX® loudness control,
UPMAX® upmixing, and ITU-R BS.1770-3 metering,
and the very latest Dolby encoding and decoding,
guaranteeing both compliance with worldwide audio
loudness regulations, and the highest audio quality.
AoIP connectivity via Livewire+ or AES67 enables I/O
of any flavor and combination or integration with third
party video and/or audio encoding software.
Omnia ONE is a single-rack-space, full-featured
broadcast audio processor for AM, FM and HD
Multicast, and studio audio processing applications. Its features include 24-bit processing,
four-band AGC plus wideband AGC, and a
four-band limiter. Omnia ONE also contains a fully distortion-controlled final limiter/clipper, an integrated digital
stereo generator with advanced peak control, and GPI remote control ports, are of which are accessible and configurable via Web browser.
Omnia VOCO 8 makes maximum use of
networked environments, packing eight separate
microphone processors into a single unit.
Although analog and AES3 I/O are provided, this
unit really shines when connected to a Livewire+
network, enabling all inputs and processed
outputs to be delivered using a single CAT-5
Ethernet connection.
Networked Codecs
Broadcast audio codecs can also become a native part of Livewire+ networks, offering the same networked control,
consolidated I/O and service density described above. Here are some examples now available from Telos:
Zephyr iPort PLUS Multi-CODEC Gateway enables
broadcasters to transport eight bi-directional
channels of stereo audio, plus eight linear PCM
channels, all with associated GPIO commands,
across any network with guaranteed QoS, such as T1
and T3 connections, MPLS networks, et cetera. Each
iPort contains eight stereo MPEG-AAC codecs, and
connects to Livewire+ networks using a single CAT-6
cable for all I/O. The Zephyr iPort makes use of
Livewire+’s advertising system in such a way that when two remote facilities are connected via its gateway, each
facility’s audio streams are visible to the users on the other end of the link. Operators can then select desired
streams by name, and consume and control them just as if those streams were local audio sources.
26 | Section 3
Z/IP ONE IP Codec is the natural product of the
work on codecs, streaming, and Audio-over-IP that
Telos has been conducting for years, making generic
IP networks as reliable and easy to use as ISDN. Z/IP
ONE features lots of proven tech and genuine FhG algorithms to deliver broadcast-quality audio over IP connections from 16kBit to 256kBit via UDP/RTP and TCP/IP via wired and Wi-Fi connections, with network edge
traversal (NAT) and built-in directory assistance. With Z/IP ONE as a shared resource of a Livewire+ network,
broadcast remotes can remain IP-based from beginning to end.
Zephyr Xstream ISDN Codec, the most widely-used broadcast codec in the world, has Livewire+ capabilities. It has
all the capabilities familiar to Zephyr users, with the additional functionality of a Livewire+ interface. With this
addition, the codec can become a truly shared resource within the broadcast facility, enabling users in-studio to
take control of an idle codec and make use of it on-demand, from anywhere inside the studio complex.
Networked Telephone Systems
Besides the consolidated I/O, one of the advantages of equipment that connects via Livewire+ is the ability to
bring control directly into the console. Phone systems can naturally benefit from this type of distributed control.
One of the most concentration-breaking operations for air talent has always been phone operation; removing the
operator’s hands and eyes from the console to operate the phone control next to it can easily disturb the flow of
on-air productions. Telos phone systems such as the Nx6, iQ6 and VX VoIP system make use of a console module
specifically designed for on-console control of the phone system, enabling operators to keep their eyes on the board
while simultaneously taking callers, leading to smoother shows and fewer errors.
Also, phone systems with Livewire+ can now be used as shared devices. The VX system, Telos’ multi-line VoIP
phone system, has a native Livewire+ connection, and has so much processing power that it can handle an entire
broadcast plant and supply a high-performance digital hybrid for every incoming phone line; this mix of features
means that VX can easily handle telephone needs for multiple stations and, thanks to its Livewire+ connectivity,
can share lines and control between multiple studios.
Even the VSet phones which are used for off-console control of certain of these systems can plug into the Livewire+
network to communicate with the systems’ phone-switching base units, and can draw power from PoE-equipped
network switches, further reducing wiring cost and infrastructure complexity.
Finally, call-screening and on-air contesting programs, such as those from Livewire+ partners Broadcast Bionics
(which come with each Telos phone system) run on computers connected to the Livewire+ network, simplifying
connection of yet another control interface.
Telos VX, the world’s first multi-studio VoIP
phone system for broadcast, is scalable to provide
advanced phone interface capabilities to facilities
of nearly any size; VX’s use of standard SIP
(Session Initiation Protocol) means that it can
work with nearly all VoIP-based SIP PBX or SIP
Telco services. A VX Engine can easily have over
100 SIP phone numbers associated with its
configuration, and each VX Engine can have 48 of
these numbers active at any time, available for
callers to ring-in. Of the 48 active lines, any 16
may be simultaneously engaged in conversation
on- air, with full audio processing. If your in-house PBX is not VoIP-based yet, commonly-available gateways from
companies like Patton allow you to connect to traditional Telco lines, including T1/E1, ISDN, and POTS. Making
use of the backfeed system that is an integrated part of Livewire+, each VX studio is supplied with its own dedicated
Program-on-Hold input and, of course, each caller receives a custom mix-minus generated on-the-fly, with no
console operator involvement.
Telos HX6 and iQ6 are cousins, six-line digital
Talkshow system with two integrated high-performance digital hybrids that can manage up to six
callers. They can connect to analog POTS or digital
ISDN phone lines, and are controlled, like the VX,
either by on-console control modules, desktop
VSet phonesets, or call-screening software. They
differ in that Hx6 includes standard Analog or
AES/EBU ins and outs along with its Livewire+
connection, while iQ6 eliminates legacy I/O in
favor of a single Livewire+ connection, which
transports all audio inputs, outputs, mix-minus and PoH audio, plus control of course, on a single CAT-6 cable. For
legacy plants migrating to AoIP, Hx6 makes sense; for a new all-Livewire+ plant, iQ6 is an excellent choice.
Networked Web-Streaming Encoders
Internet streaming of audio has been around for a long time of course (Telos first entered this field in 1999 — you
may remember the AudioActive Encoder, the first professional hardware for MP3 streaming), but as consumers
increase their use of cellular data networks and WiFi connections to take their entertainment with them, streaming
content has finally come of age. A recent industry study showed that Web audio streaming contributed 36 billion
hours to listening time in 20141, and a Cisco research paper predicts that online video will account for 79% of the
world’s IP traffic by 20182 — a staggering amount of time spent listening/viewing digital entertainment.
It makes sense, then, for generation of streaming content to be an integral part of broadcast studio networks. The
Telos Alliance has fielded a full family of products which take advantage of Livewire+ networks to select, process,
encode and deliver streaming audio to listeners, with tightly integrated workflows that require almost no additional infrastructure.
Z/IPStream R/1 is a rackmount hardware device
which combines audio processing for bit-reduction with MP3 and AAC encoding, and dual-stream encoding for simultaneous output of
high and low bitrate streams, or of different
encoding methods (such as MP3 and AAC). The
unit is fed a selected program output from your Livewire+ network; the audio processing section then tailors input
audio to match selected output bit-rates using AGC, 3-band compressor/limiter, EQ, low-pass filter and lookahead final limiter. Then the stream is encoded with your choice of MPEG Layer III, AAC-LC, HE-AAC or HE-AAC
v2 formats. Support for Adobe RTMP and RTP streams (including RTP multicast) are also included. Streams can
then be “tagged” with “now playing” metadata received from automation systems, and the encoded audio fed to
your streaming replication service via a WAN gateway.
1 Research And Markets, “Internet Music Radio Programmers 2014 – 2017: Music Plays and
Monetization Mainstays”, August 2014
2 Cisco Systems Inc., “Cisco Visual Networking Index: Forecast and Methodology, 2013–2018”,
June 10, 2014.
28 | Section 3
Z/IPStream X/2 and Z/IPStream 9X/2 Streaming Software Encoders for PC workstations have capabilities similar
to the hardware device just described, but with added support for “adaptive streaming”, a video technology now
making inroads with audio streamers. This is interesting because, using adaptive streaming, technologies such as
Microsoft’s Smooth Streaming and Apple HLS formats allow media players to automatically adapt to changing
network conditions by encoding at multiple bitrates, whereupon sample-aligned output streams enable listeners’
media players to seamlessly switch between bitrates in response to Internet conditions. Z/IPStream 9X/2 includes
multiple processing upgrades, notably “Undo” for restoring poorly mastered source material, as well as 6-band
parametric EQ and multi-band processing configurable from two to seven bands.
Taking advantage of its networked environment, Z/IPStream software has an audio replacement capability that
allows some streaming audio material – such as local spots – to be automatically replaced with content from
secondary source or audio files, all from Livewire+ networked sources of course. The program is cloud-ready as
well, and can be monitored and configured remotely, the same as all the rest of your Livewire+ network devices.
Z/IPStream.S4 is a PCIExpress card for PCs that allows offloading of streaming content origination to a local PC.
Connecting to the network via Livewire+, Z/IPStream.S4 cards can process and encode audio stored on the PC
itself, then deliver it to a streaming provider, or take Livewire+ streams from elsewhere in the facility and perform
processing and encoding before delivering them for distribution via a WAN gateway.
Networked Time Management
Broadcasters have long taken advantage of digital audio manipulation in order to bend time – make programming
shorter or longer, to fit, with associated pitch-management routines that achieve inaudible results. The devices
that perform these functions, operating in the digital domain already, are an excellent fit for the networked
Livewire+ environment. They can receive and manipulate audio by way of their network connections; those same
connections carry the GPIO commands which trigger START, STOP and DUMP commands. Further integration
is accomplished by attaching those commands to reserved keys on the operator’s console, allowing the delay unit
itself to be located in the TOC or other remote area, while the operator controls its function without removing
attention from the mixing board itself.
The 25-Seven Program Delay Manager is one of the
best-known of this type of unit, and is equipped with
Livewire+. In addition to its primary function of
blocking unwanted words from air, it has the unique
ability to capture both the censored and uncensored
audio, and send it via e-mail to the appropriate manager. Because Livewire+ networks can connect, via gateway, to
your plant’s business LAN, this is a very handy and easily-implemented feature.
Networked Intercom
In larger radio operations, and nearly all television plants, intercom communications are an essential piece of
day-to-day functioning. In the past, whole-plant intercom meant a complicated and expensive central switching
matrix, discrete wiring pairs to and from every location imaginable, and audio bandwidth so limited it sounded
more akin to a burger drive-through speaker than something employed in a broadcast plant.
With IP, all of this can be radically simplified, and QoS greatly improved as well. Intercom audio becomes no
different than any other audio source on your network. Point-to-point or point-to-multipoint Intercom communications can be sent and received using the Ethernet infrastructure already in place for your broadcast audio; GPIO
closures for attention tallies or speaker mutes may be sent with Intercom audio in the same way logic commands
are sent with audio from playout systems, or to activate recording devices, etc. With an IP-based intercom system,
substantial savings are realized because the intercom audio “rides along” with the broadcast audio for free – no
additional wiring infrastructure or central card cage required.
Also, a huge performance upgrade is realized. Since AoIP carries full-bandwidth digital audio, the sound from an
IP-based intercom system is radically upgraded from the old 5kHz systems; in fact, it can carry full-fidelity 20Hz –
20kHz audio suitable for air. No more need to hold a breaking news story until a reporter can be mic’d; you can now
simply put the intercom itself on the air.
Axia’s IP Intercom is just such a system. It’s the world’s first
broadcast intercom system that’s IP based. It connects to the
same Livewire+ network as your consoles, mix engines and
other audio devices. There are drop-in modules for Axia
consoles that enable board operators to converse directly with
any location in your plant using the board mic and preview
speaker already in place (rackmount and desktop stations are
available as well, natch).
Further, since PCs can contribute to and consume audio from
the Livewire+ network, why not turn them into intercom
stations as well? Axia SoftCom software installed on any
workstation or laptop turns that computer into an intercom
station as well, making it possible for literally every location
in your facility to be reached instantly.
Nuts & Bolts: Making Livewire+ Play
Now we move to making audio happen. Time to take the gear out of the shipping carton and make it play. This
section gives you practical information. Details about the underlying tech are reserved for later.
Livewire+’s Channel and Name system
An advantage of having a data network carrying our audio streams is that we can send identifying information on
the same cable and system. Receivers can build tables of available audio, and testers can identify specific streams on
a cable. In Livewire+, we have both a numeric and a text ID for each audio source.
Hardware Livewire+ devices are configured either using a networked PC’s web browser, or with local pushbuttons
and displays. PC Livewire+ nodes will have a configuration window that opens when you click on the application
icon. Details for each are in the manual specific to the product, but the general approach is the same for all audio
and GPIO.
Channel numbers may range from 1 to 32767. You assign these to audio sources as you wish.
New units are pre-configured from the factory to start with channel 1, thus a 4-channel xNode will come assigned
to channels 1-4. Two new units can be connected to each other with a “cross cable” (described later) for immediate
out-of-the-box testing. For your network, you should reserve channels 1-4 for testing and not assign them for
routine use. Then, if you plug a new unit into the network before you configure the channels, there will be no
problem with conflicts.
In a large system, you will probably want to have a people-friendly naming and numbering system that reflects
studio use or location and to help prevent accidental duplication of channel assignments (a big no no by the way).
You have plenty of numbers to use, so you don’t have to conserve them. For example, the channels associated with
Studio 1 could start with 100, Studio 2 with 200, etc. There is no requirement that channels be assigned in order or
contiguously from a multi-channel device.
Devices such as telephone hybrids and codecs need audio in both directions, so when appropriate, a single channel
contains a “to device” audio stream as well as the usual “from device” audio. In this case, you can think of the
channel as something like a telephone number that connects a call with audio in both directions. The advantage of
this “bundling” of the two audio directions is that the association is naturally maintained when studio mixers are in
the picture. Mixers generate the feed to devices (usually mix-minus, but not always) and automatically assign it to
the source channel number, and this association is kept regardless of which fader is being used, etc.
Text Name
The text name may be up to 24 characters and you choose it as you wish. This is what will appear on the xSelector’s
display screen, console source select lists, etc. Most devices are not able to display all 24 characters, so will truncate
to show what they can.
You may wish to include in the name the rack number or room name of where the Node is located, to help orient
yourself in the case of a future emergency.
A typical name might be ST1CD2 for Studio 1, CD Player 2.
Our studio mixing consoles (Fusion or iQ, for e.g.) automatically generate return feeds to devices that need them,
creating the text name for these in the form “To: name”. For example, if you have a source called “Hybrid 1”, the
mixer will generate an audio stream named “To: Hybrid 1” and advertise it to receivers.
There are also GPIO (General Purpose Input/Output) channels and text names. These work in a fashion very
similar to the audio source channels and names.
GPIO channels often share the same channel number as an audio source. A typical situation would be when you
have a CD player that needs start control from an audio mixing console. The mixer automatically generates the
start command and puts it on the channel number you assigned to the audio source. To cause a particular hardware
GPIO to output this command as a contact-closure pulse, you configure the GPIO device to listen to this channel.
As with back-fed audio, control follows the audio source to whichever fader is being used.
But GPIOs may also be independent of audio sources. In this case, the Livewire+ system provides a pass-through
function where outputs follow inputs – sort of like a GPIO distribution amplifier.
Sources vs. Destinations
We’ve always struggled with terminology when referring to audio input/output from devices such as codecs and
hybrids where there’s local audio I/O as well as a combined network I/O port. We will try to be consistent within
the Livewire+ realm by using the following terms:
♦♦ Source – this is an audio input to a hardware Node and therefore available on the Livewire+ network as an
audio stream that can be accessed by other Livewire+ nodes. Of course a mixing engine can generate new
audio sources and in this case there is no associated hardware audio input.
♦♦ Destination – this is an audio output from a hardware Node and therefore represents playback of some
stream from the network. Of course a mixing engine or Livewire+ capable audio device may access a
Livewire+ stream and in these cases there would be no associated hardware audio output.
So, to reiterate, sources represent the feeding end of the audio stream equation whereas destinations are just that,
one or more destinations where that stream is used.
32 | Section 4
Following Gauss’ dictum that “an example is worth two books,” let us now turn to some to show you how Livewire+’s channel and name identification work.
On the next few pages we’ll show some of the web configuration pages for the Analog xNode. The first example
is the home page that is displayed once you have logged into the node. It simply lets you navigate to the other
configuration screens.
The Sources page shown above permits you to configure locally generated sources. It allows you to assign names
and channels to the sources that will be generated by this node, and to configure the audio inputs associated with
those sources.
♦♦ The Source Name entry is where you put the text ID for the node. There are 4 Source Name entries, one
for each audio channel. This is the text name for the individual audio source. (xNodes can also split their
4 stereo inputs, turning them into 8 independent mono inputs; in this case, you would have 8 Source
Name entries.)
♦♦ Channel/Address is where you enter the channel number for each source. Note the instructions at the
bottom of the screen for entering AES67 sources; more on this in the xNode User Manual.
♦♦ Stream Mode allows you to decide whether each stream should be a Livestream, Standard Stream or
Surround Stream. Why would you use one versus the others? A satellite feed will never be in a mic-toheadphones path, so only the Standard Stream would be required. A microphone source is live audio, and
so would be a Livestream. You may also select Mono streams or Surround streams (only used if yours is a
5.1 Surround broadcast facility).
♦♦ You can trim Input Gain by specifying an amount in dB. This can also be set on the Meter screen, in case
you desire to set gain “by eye”.
The sample Destination Screen, pictured here, is used to configure the local units’ outputs, or destinations.
This is where you configure the output channels, with the menu options described below.
♦♦ As with sources, you can enter a text Name for the destination associated with each output port.
♦♦ Channel/Address is where you tell the unit which audio stream is to be output from each audio output.
As previously discussed, each Livewire+ stream is identified by both a text name and with audio channel
number. You can enter the channel number directly, or you can use the button to the right of the channel
entry field to open a page with a list of active audio channels; you can choose from among the channels
listed within it. Usually the list contains text names of audio streams, but if no text name has been
assigned, you will see the device type and IP number instead.
♦♦ Type lets you choose whether the Destination feed is one-way (“From Source”, outbound to a monitor or
headphone feed, or a record input) or two-way (“To Source”, supplying a backfeed for a phone system or
♦♦ Gain again appears in the Analog Node Destination screen to allow you tailor specific outputs to feed
High-Z (professional) or 600-Ohm (prosumer) gear.
Hot Tip!
You will only have a complete list of audio sources if you have already configured all your source
nodes and have them connected and operating so that they are advertising to the network. You can
always just enter the channel number here if you don’t yet have your source nodes working, but it
will be more convenient if you prepare all your sources before moving on to destinations.
34 | Section 4
The Meters page for our example audio node lets you monitor the levels for each input and output channel. This is
also an alternative to the source page for setting input gain.
The Synchronization and QoS page is an advanced feature page, and its lets you set some other values related to
system clocking and AES67 operation. These are described in detail in the unit’s manual, but to appreciate the context, you will also need to understand more of the Livewire+ and networking basics described later in this document.
36 | Section 4
The System page shown below permits checking the IP address and performing software updates.
Our samples so far have been from the 4x4 Analog xNode. Next let’s look at how sources and destinations are
handled by the IP-Audio Driver used on Windows™ computers.
Soundcard Emulation. The IP-Audio Driver looks like a standard
sound card to Windows™. Each of the Driver’s sources (e.g. streams
originated by this computer) shows up as a sound card. Computer
sources sent to the network are called Livewire Out XX in the
Windows™ Sounds and Multimedia Properties Control Panel; inputs
to be received or recorded from the network are called Livewire In XX.
You can define one of these Livewire+ sources as Windows’ Preferred
Sound Playback Device from the Windows Control Panel as
shown here.
Driver Configuration. The Axia IP-Audio driver is configured for
sources and destinations much like the Axia audio nodes. The driver is
configured from the window shown below which is invoked either by
a Start menu shortcut or a Control Panel applet; the various settings
are described as well.
♦♦ Sources and Destinations – You can see that the node and source naming and channel idea is the same as
for the hardware nodes. Any audio channels you want to receive are entered into the Destinations boxes.
If you don’t know the ID number, you can choose from text lists instead, by clicking on the Browse button.
♦♦ Livewire Network Card – A PC running this driver may have two network cards, one for general data and
another for audio streams. The Livewire+ Network Card entry lets you associate Livewire+ audio with the
appropriate card.
♦♦ Allocation – Clicking this brings up a screen that lets you set stream characteristic values. This is covered
in greater detail in the Axia IP-Audio Driver manual.
♦♦ Statistics – This button brings up a screen with lots of information useful for debugging network problems.
♦♦ GPIO – Opens the configuration screen that allows GPIO triggers to be attached to individual Driver
audio streams; useful, for example, when you wish your automation system to turn a console fader on
when an audio stream begins to play. This is covered in more detail in the User Manual.
When you have finished configuration, the Livewire+ network looks like a sound card to any Windows application
that uses the standard wavin/out or ASIO audio interface. In Windows applications where you normally select the
soundcard you want to use, you will select a Livewire+ channel instead. In the example, an audio player that has
selected Livewire Out01 will put its audio into Livewire+ stream channel 11111 and be available to all Livewire+
devices on the network.
38 | Section 4
The GPIO xNode is a hardware box with 6 DB-15 connectors, one for each port. Each has 5 inputs and 5 outputs.
GPIO channels may be associated with audio channels or may be independent. If they are independent, they must
not use the same number as any audio channel – they share the same “channel space”.
You can monitor the status of each with the indicators on the xNode’s front panel display or configuration Web page.
xNode Configuration & Web Access
As shown in our examples above, you’ll need to configure various parameters in the hardware nodes. Some very
basic parameters such as the name and IP address can be configured from the front panel. In fact, for the basic audio
snake application we need not even access the xNodes’ web pages. However in most cases we will need to do so, but
we will need to assign an IP address first.
Front Panel xNode Configuration
A number of items can be programmed from the front panel of most of the hardware nodes. This is covered in more
detail in the individual manuals. However we will cover setting the IP address here since you’ll need to assign an
IP address to enter via the web browser, and we’ll cover that next. Here’s how to assign an IP address to a typical
audio xNode.
Configuring xNode IP address
Each Livewire+ node must have a unique IP address. The only exception is when two nodes are connected in the
point-to-point configuration.
Fast Setup
A Fast Setup routine is built into each xNode to assign an IP address and I/O channel numbers in less than 1
minute. Here’s how:
♦♦ Apply power to the xNode and wait for the boot process to complete. You will see a home screen identify-
ing the xNode’s type and software version.
♦♦ Press the top button to the right of the display twice, to reach the ID Page. You’ll see a Node ID with no
value no IP address.
♦♦ Press the lower button, pencil icon, for 10 seconds to enter into edit mode.
♦♦ With the cursor shown, use the top button to increment value and the bottom button to move the cursor
to the next position. Press “next” button twice and press the increment button once to give the ID value of
1 to the xNode. (You’ll give a different ID number to each xNode you install.)
♦♦ Pressing the move button one additional time completes edit mode and automatically configures the
xNode with IP address and channel numbers of 101 – 108.
That’s actually the end of fast setup! With IP and channel numbers configured, you can attach the xNode to your
network and begin connecting sources.
Manual Configuration
If you’d rather assign an IP address manually, you can do so from the xNode’s front panel:
♦♦ Press the top button on the xNode front panel until you see the screen that displays the assigned IP
address and netmask.
♦♦ Press and hold the bottom Pencil button to enter edit mode.
♦♦ With the cursor shown, use the top button to increment value and the bottom button to move the cursor
to the next position. Do this for each digit in the IP address field; at the end, pressing the bottom button to
highlight the netmask field.
♦♦ The entire netmask field will be highlighted; press the top button to begin editing the octets or press the
bottom button to complete the netmask editing.
♦♦ Pressing the top button one additional time completes edit mode and highlights the selection for the
xNode’s active Ethernet port. Press the top button again to complete your editing and save your new
IP address.
Accessing a Node via a web browser
To access the built in web server from a computer, the computer and node must be connected to the same LAN (or
the computer and node can be connected using a “crossover 100 Base-T” Ethernet cable).
Note that the IP range (e.g. the first three numbers of the four numbers of the IP address of the computer and the
node must match, or additional configuration will be required.
♦♦ Open your computer’s browser and type in the IP address of the xNode, such as http://
where “” is the IP address of the xNode to be configured, into the URL field.
♦♦ You’ll be asked for a username and password. Default Authentication is:
¸¸ Username: user
¸¸ Password: (leave blank)
♦♦ Once logged in, select the Simple Setup button to enter into a simple configuration screen for the xNode.
The options available will vary based on the xNode.
♦♦ Enter descriptive text in the Source Name fields which describe what devices are connected to the xNode
(e.g. CD player, Turntable, PC Out-1, Aux Input).
♦♦ Enter descriptive text into the Name field under the Destination section which documents what is
connected to the xNode’s outputs (e.g. Control Room Monitors, Headphone, CF Recorder, STL input).
Plugs & Cables
Livewire+ systems use primarily copper cables, but you can add fiber where it makes sense. We’ll start here
with copper.
An important goal in the design of Livewire+ was to simplify installations. One of the ways we do this is to let you
standardize on a single cable type, plugs, patchfields, etc. This is consistent with the modern way of thinking about
cabling in office buildings where a common type can serve different applications. You can use the same connectors
and cables for everything in your plant. And for big new installations, outside contractors can install and test the
wiring infrastructure for everything. In fact this is one reason why we suggest that Broadcast Engineers should
become familiar with the relevant standards such as EIA/TIA-568-A & B.
40 | Section 4
The 100/1000BASE-T Ethernet we need for most Livewire+ devices requires RJ-45 8-pin modular plugs and jacks.
So, we’ve standardized on RJ-45s for balanced high-level analog and AES connections as well. There are a lot of
connectors being used for analog audio these days, so why did we go this route? The reasons are cost, density,
compatibility, and convenience. RJ-45 sockets and plugs are a lot cheaper than other choices, both for us at the
manufacturing level and for you at the time of installation. Density is an important advantage: We can get only a
few XLRs across the rear panel of a 1U rack box and we need two of them for each stereo connection.
xNodes would have to be twice their size to have the same channel capacity as we have now using 1U nodes that
fit in half a rack-width. A single RJ can do both channels on one jack and we can fit lots of them on a 1U box. XLRs
and DBs need to be soldered, and shells assembled, etc. Molexes need a separate crimp for each wire and are not
standard. RJ crimping is convenient procedure compared to the others. And you will already have the plugs, cables,
and tools at hand.
The following tables summarize the cable types that could be used in a Livewire+ system and their applications.
Non-Ethernet Cabling Relevant to Livewire+ Systems
Analog Audio, balanced, high-level
Usually shielded CAT-5, but unshielded with care
Balance 110-Ohm AES3 Digital Audio
Usually shielded or unshielded CAT-5 or 5e
Ethernet Cabling Relevant to Livewire+ Systems
Max Length
Where Used
10Mbps on 2 pairs CAT-3 copper. Obsolete for
new installations.
100 m.
Not used.
100Mbps on 2-pairs CAT-5 copper (CAT-6
recommended for Livewire+ to add a safety
margin). Most common Ethernet media.
100 m.
Livewire+ nodes, PCs.
100Mbps on fiber
2 km, 20 km.
Livewire+ nodes with ext.
media converters
1Gigabit on 4 pairs CAT-5e copper (CAT-6
recommended for Livewire+ to add a safety
100 m.
Mixing engine to switch, PCs,
1 Gigabit on short wavelength fiber, multimode
220-550 m.
1 Gigabit on long wavelength fiber, single-mode
5 km.
CAT-5 for Audio?
Using CAT-5 “digital cable” for audio may seem strange at first, but it does make sense. The low capacitance and
tight twisting requirements necessary for high-speed networks are good for analog and balanced AES audio as
well. Because you have a single cable and connector type for everything in your facility, as your studios evolve, the
same cable that was once used for analog may be used for AES, Livewire+ digital audio, general Ethernet data, or
whatever else might come along.
Does this work in the real-world? Sure it does – as demonstrated in the many installations that have been done
using the Radio Systems “StudioHub+” product family. We use the StudioHub 2 format for our connectors by the
way, to allow convenient use of prefabricated connectors by Livewire+ users. More on this below.
Ethernet 100BASE-TX
Livewire+ uses 100BASE-TX copper wiring with RJ-45 style plugs and jacks for connections from audio nodes to
You must use Category 5e cable and accessories or better. If you’re interested in “future-proofing” your new
facilities , you might also consider CAT-6 in order to be ready for 1000BASE-T. CAT-6 is not much more expensive
that 5e and it does perform better, particularly when a run has lots of bends that could disturb the pair relationships
within the cable jacket, or has many punch blocks and/or patch cables.
Pin numbering, jacks, cables, and color codes
Modular wall jacks are normally installed so that the pins are at the top of the cavity. This helps to protect the
contacts from water when baseboards are mopped and from dust. When the jack is oriented in this position,
looking into the jack with the contact pins at the top, the pins are numbered 1 to 8 from left to right. Some jacks
may not have all pin positions populated, but the numbering would still begin with the first position. For instance,
the “RJ-11” style jack is a 6-position 4-pin jack. Therefore it has pins 2-3-4-5 and pins 1 & 6 are usually absent.
You should take care not to plug an RJ-11 into an RJ-45 jack. It will work to connect the pairs that are supported
in the plug, but the plastic part on both sides will push the outer pins on the jack up, and they may not make good
connection when the jack is again used for an RJ-45 plug.
Ethernet uses 8-position 8-pin modular connectors. TIA/EIA specifies two standards for wiring RJ-45 style cables.
The T568A color code is “preferred” by TIA/EIA but is not so usual in the USA for business installations.
The TIA/EIA T568B color code cable specification has the
same electrical connections, but has the green and orange
pairs swapped (as shown). This is also known as the AT&T
258A wiring sequence and has been widely used in the USA.
It is used by the Radio Systems StudioHub+ system for analog
and AES connections, so we recommend it for all new
Either sequence will work just fine if you have it on both ends.
In either case, you have a cable with 4 pairs wired straight
through, both ends wired identically.
Interesting note: 100BASE-T with the final X being dropped is oftentimes used as shorthand for 100BASE-TX. The
100BASE-T designation officially refers to both copper and fiber formats at 100Mbps rate, with TX the specific
designation for copper. The abbreviation in popular use arises from the fact that the copper formats on either side
are called 10BASE-T and 1000BASE-T. And that the -T is supposed to stand for “twisted pair” – except here for
some reason. Leave it to standards bodies to be non-standard.
42 | Section 4
Depending on the cable manufacturer, the color conductor of each pair may or may not have a white stripe. The
other half of the pair is usually white with a colored stripe, but sometimes can be only white. Both formats are
shown in table form below:
TIA/EIA-568-A T568 Wiring Standard (preferred for Livewire+)
Protective Ground
Transmit +
Transmit -
Receive +
Receive -
TIA/EIA-568-B T568 Wiring Standard (Optional and acceptable for Livewire+)
Protective Ground
Transmit +
Transmit -
Receive +
Receive -
Something to watch out for: The old telephone USOC wiring code has the pairs in the wrong place, with the wiring
in simple one-pair-after-the-other sequence. You’ll have a split-pair if you have this sequence – and a lot of crosstalk
and interference problems. You need to be sure that the pairs correspond to Ethernet’s requirements.
Why does Ethernet have such a strange wiring sequence? Because the center two pins, 4 & 5, are where telephone
voice circuits are wired. The designers of the standard thought that some people would want to use a single cable
for voice and data, so they kept Ethernet clear of the telephone pins. There is also this: if a user plugs his PC’s
network connection into the phone jack, he doesn’t blast the network card with ringing voltage.
Even though you have two unused pairs in the standard CAT-5 4-pair cable, you should not share the cable with any
other service since 100BASE-TX was not designed to withstand additional signals in the cable. The reason for the
extra pairs is that you might want to upgrade to 1000BASE-T or some other yet-to-be-introduced service later.
Finally, on this topic, something really nuts… The overall cabling specifications standard and document from
TIA/EIA was called TIA/EIA-568-A Commercial Building Telecommunications Standard. Within this were the
T568A and T568B pinout standards. Note the dashes and lack of same. Now there is a new TIA/EIA-568-B overall
standard, which has the same two pinout standards within. Couldn’t these guys have been a bit less confusing?
Crossover 100BASE-T Ethernet Cable
Sometimes you want to connect two Livewire+ nodes directly together without a switch, such as for initial setup,
testing or when you want to make a snake. Or you might want to connect an xNnode directly to a PC for initial configuration or as a sound card. In this case, the Transmit of one device must be connected to the Receive of the other.
For this, you’ll need the special crossover cable wired as shown below.
Not Used
Not Used
Not Used
Not Used
Many modern Ethernet switches have ports that automatically sense the need for a crossover function and
configure their ports appropriately. So when you are connecting ports from two switches, you probably will not
have to use a crossover cable.
1000BASE-T Gigabit Copper
We use 1000BASE-T to connect studio mixing engines to switches. If your Livewire+ network consists of multiple
switches, you will also use it to link switches to each other.
1000BASE-T works with CAT-5e, but again we recommend CAT-6, because 1000BASE-T uses the same RJ-45s as
100BASE-TX, but needs all four pairs. Either the T568A or T568B wiring sequence will work, but you will have to
be sure all four pairs are wired through and working. Again here, the advantage of choosing one scheme and using it
for everything (e.g. T568B on CAT-6) is obvious. 1000Base-T Signal Designations are as follows:
There are no separate send and receive pairs for 1000BASE-T. Each pair both sends and receives with a hybrid at
the ends to separate the two signal directions. Thus, there are effectively four paths each way. The signaling rate for
1000BASE-T is the same as for 100BASE-T – which is why it can be run over the same cable.
Nevertheless, 1000BASE-T is more sensitive to certain performance issues owing to the hybrids and twice the
number of signals in a 4-pair cable. That’s why CAT-5e or CAT-6 is recommended. And you should always use
high-quality factory-made patch cables.
You shouldn’t ever need a 1000BASE-T crossover cable, but who knows? Anyway, a universal crossover cable can be
made (or better, purchased) that works for both 100 and 1000BASE-T; the following table shows the configuration
for a Universal 1000Base-T/100Base-T Crossover Cable.
44 | Section 4
100Base-T Crossover Cable
Audio connections
xNodes are high-density devices, and they provide many inputs/outputs in a ½-width, 1RU package. To address
this, we give you two options: RJ-45 connectors for individual pairs of wires, or pigtails such as those used in
pro-audio applications which provide an entire bundle of 8 pre-wired XLR-M or XLR-F connectors while using just
a single DB25 hookup to the xNode.
AES/EBU xNode rear panel. Both RJ and DB audio connections are provided
For RJ-45 connections, we use the pin-outs established by the Radio Systems StudioHub+ wiring system, which
has become a de-facto standard. Since we follow this standard, Studio Hub wiring components may be used for
the analog and AES part of Livewire+ installations. Radio Systems, and other Livewire+ partners such as Xi Audio,
offer an extensive line of single “dongle” and multi-pair harness cables pre-wired to connect to a variety of popular
studio gear. There are also balanced-to-unbalanced, AES to S/PDIF, and AES to TOSLINK adapters, headphone
amps, etc.
For the DB25-to-XLR fantails, cables such as those made by Hosa are generally regarded as gold-standard, although
these are available from other makers, too. They come in lengths from 3 – to 7 m. to suit your application.
Pre-made breakouts from Radio Systems, Hosa, Xi and others help quicken connections.
While unbalanced connections can be used be very short runs with unshared and shielded cables, balanced
connections are essential for anything over a few feet in length. The input stage of any attached analog equipment
needs to have good CMRR (Common Mode Rejection Ratio) and high-frequency filtering in order for balanced
connections to effectively cancel crosstalk and interference. With 60dB CMRR, Axia Livewire+ xNode inputs are
designed to be no trouble in this respect.
The pinouts for the RJ-45 style audio connectors are shown below:
Axia/Radio Systems Standard for Analog and AES wiring on RJ-45
Protective Ground
White/Slate & Slate/White*
N/C (GND)**
N/C (15-)**
N/C (15+)**
** Used to power “spoke” devices such as balanced-to-unbalanced converters. xNodes do not supply this
voltage, but external supplies can be used if needed.
Off-the-shelf or homemade RJ-45 cables, together with adapter dongles from Radio Systems, Xi, Hosa and others,
connect the nodes to audio equipment.
46 | Section 4
Installing RJ-45s
It would be possible to build a sophisticated multi-studio facility without ever wiring a single RJ-45 plug – you
would use modular patch fields or jacks at each end of the long “horizontal” cable with punch-down 110-style
connections. Then factory-made patch cords would be used to get from the switch or Livewire+ node to the patch
jack. And this might not be a bad idea!
Nevertheless, you’ll probably find yourself installing your own plugs at some point, so here is some advice:
♦♦ If you are making a patch cord, use stranded-conductor cable. Solid is likely to break after some time
being plugged and unplugged. However solid cable should be used for backbone wiring as it has less loss.
♦♦ Be sure you are using plugs designed for the cable type you are using. Plugs for solid and stranded wires
are not the same.
♦♦ Plugs from different manufacturers may have slightly different forms. Be sure your crimp tool correctly
fits. In particular, the crimper made by AMP will only work with AMP plugs. Buy a high-quality crimping
tool to help prevent any problems.
♦♦ The outer jacket should be cut back to about 12 mm (.5 inch) of the wire tips. Check to be sure there are
no nicks in the wires’ insulation where you cut the jacket (an appropriate tool can be purchased to permit
you to do so rapidly without fear of damaging the inner insulation).
♦♦ Slide all of the conductors all the way into the connector so that they come to a stop at the inside front
of the connector shell. Check by looking through the connector front that all the wires are in correct
♦♦ After crimping, check that the cable strain relief block is properly clamping the outer cable jacket.
♦♦ When checking the cable either with a tester or a real device, wiggle the cable around near the plug to be
sure that connector is working reliably with stress.
If it’s your first time crimping your own, you’ll probably need a couple of tries
to get it right, but after some experience it will start getting pretty easy. Certain
RJ connectors include a small carrier that the wires can be fed into first, and
then slid into the connector itself. These are recommended as they speed
installation and improved accuracy.
Designing & Building Your Livewire+
Ethernet System
As with analog audio installations, Livewire+ set-ups range from the very simple to complex facility-wide installations with hundreds of ports. This section is aimed primarily at those who will be building large systems.
Ethernet is balanced and transformer coupled, so has very good resistance to interference and has no problem
with ground loops. However, frequencies ranging to tens of megahertz are being used, so care must nevertheless be
In the bad old days, wiring was specific to the task – and often to the vendor. Each telephone, network, and audio
had its own cable type and wiring protocols. The idea at standards bodies like the Telecommunications Industry
Association (TIA) and the Electronic Industries Association (EIA) in the USA is to define classes or categories of
cables and accessories that can be used for all applications specified for that class. With this, you have a vendor-independent way to wire buildings and facilities so that services from many vendors can be supported over time
without replacing cabling and connectors. The name for this concept is structured wiring.
The long cables that go from equipment rooms to node locations are called horizontal cables. They usually
terminate in RJ-45s, either in patchfields or on wall jacks. Patchcords with RJ-45s at each end complete the system,
connecting the nodes and central equipment to the jacks.
(Charles Spurgeon in Ethernet: The Definitive Guide says that you should consider wiring to be the essential
skeleton for your network installation. He goes on to point out that network cabling skeletons are often hidden in
the time-honored place for skeletons: a closet. Rim shot.)
Twisted-pair Cable Categories
Cable categories are key to the structured wiring concept. The cabling specifications for the various categories
are in the TIA/EIA-568-A (and B) Commercial Building Telecommunications Cabling Standard. The following
categories are defined:
♦♦ Category 3: These are used only for telephone and Ethernet 10BASE-T, so are not useful for Livewire+
♦♦ Category 5: This designation applies to 100 Ohm unshielded twisted pair cables and associated connect-
ing hardware whose transmission characteristics are specified up to 100MHz. CAT-5 cables are today’s
most common because they support Ethernet 100BASE-TX.
♦♦ Category 5e: This is enhanced Category 5 cable. The main application is for Gigabit 1000BASE-T. While
CAT-5 is acceptable for 1000BASE-T, CAT-5e is officially preferred.
48 | Section 5
♦♦ Category 6: While not strictly necessary except for 1000BASE-T links, we recommend CAT-6 for all new
Livewire+ installations. CAT-6 provides significantly higher performance than that of CAT-5e. The main
difference is that this cable has a plastic pair separator inside that holds the wires in correct relation to
each other. This is what makes CAT-6 larger in diameter than CAT-5 cables.
Belden has a CAT-6 cable called Mediatwist that looks very interesting. This cable has a half-moon shape
and the pairs are tightly held in molded channels. This product also has the two wires in each pair glued
together so that the twist characteristic is fixed and stable regardless of manufacturing tolerances, cable
flexing, etc.
The most significant difference between cables from each category is the number of twists per foot and the
tightness with which the twists and the spacing of the pairs to each other are controlled. The wire pairs in a voicegrade Category 3 cable usually have two twists per foot, and you may not even notice the twists unless you peel back
quite a lot of the outer insulation. Category 5 is tightly twisted, something like 20 per foot. This results in superior
crosstalk performance at higher frequencies.
Another characteristic of twisted-pair cables is the type of insulation used on the wires and the cable jacket.
“Plenum rated” cables are more stable with changing temperatures due to their using Teflon rather than PVC insulation. Plenum rated cables are required in air handling spaces in order to meet fire regulations. Teflon produces
less smoke and heat in the case of a fire than PVC and does not support the spread of flames.
Special Care for Ethernet Audio
“Normal” data over Ethernet is usually TCP/IP protocol. As discussed later, TCP has a re-transmission mechanism
that detects errors and fixes them by requesting and obtaining replacement packets when one has been received
with a defect. This mechanism is not used for audio – it can’t be when you need low delay and multiple receivers.
So it could be possible that a network could apparently be OK with computer data, yet exhibit errors with audio
because TCP is covering-up underlying problems.
A particular concern is to prevent impedance reflections at cable termination points and to not disturb too much
the position of the wires inside the cable. Here are some specific recommendations:
♦♦ Use the minimum number of terminations and patches that will support your application.
♦♦ Use patch cables, connectors, and other accessories rated at the same or higher category level as the cable
you are using. Generally, your best bet is to buy pre-made patch cables to both save money and time as
well as assure reliability.
♦♦ Keep a wire pair’s twist intact to as close as possible to any termination point. For Category 5, this should
be to within 1.3 cm (.5 inch).
♦♦ Maintain the required minimum bending radius. For a 4-pair 0.5 cm (.2 inches) diameter cable, the
minimum bend radius is 4 times the diameter, or about 2 cm (.8 inches).
♦♦ Minimize jacket twisting and compression. Install cable ties loosely and use Velcro fasteners that leave a
little space for the cable bundle to move around.
♦♦ Do not staple the cable to backboards. If you tightly compress the jacket, you will disturb the twists inside
and the relationship of one pair to another, which could cause crosstalk.
♦♦ Do not overfill conduits.
♦♦ Avoid stretching the cable. The official recommendation is to use less than 25 lbs. pulling pressure.
♦♦ Avoid close proximity to power cables and equipment that generate significant magnetic fields. The
official recommendation is minimum 6.4 cm (2.5 inches) from power cables when the Cat 5 is either
inside a conduit or shielded. Care should be taken also with fluorescent lighting fixtures, motors and
♦♦ The pins on RJ-45 plugs are gold plated. But not all connectors are. For maximum reliability, use connec-
tors with 50m gold plating.
To Shield or Not to Shield
Unless you are in a high RF environment or you intend to run your network cables close to audio cables with
equipment that has poor balancing on the inputs, you should be able to use unshielded twisted pair for your
Ethernet connections. If you decide to shield, the usual procedure to attach it only at one end applies in order to
prevent ground loops.
Unbalanced Connections
Livewire+ nodes have very good common mode rejection. This, coupled
with the highly twisted CAT-5 or -6 cable works extremely well in the
balanced pro-audio environment. Unbalanced interconnections are
problematic however and should be avoided for the usual reasons. If you
need to interconnect a Livewire+ node to unbalanced gear we strongly
recommend that you use a balanced to unbalanced buffer amplifier or
transformer located as close as practical to the unbalanced equipment.
There are a number of commercial off-the shelf options to accomplish
this. In particular the Radio Systems StudioHub+ Matchjack series offer plug and play compatibility between the
RJ-45 balanced and consumer unbalanced worlds.
More than Four Pairs in a Cable
Back in the 10BASE-T days, it was usual to have phone-type 25-pair cables carrying data signals. But the standards
for Cat 5 and better call for individual cables for each connection due to the possibility of multiple disturber near
end crosstalk – or many signals adding up to create combined crosstalk at too high a level.
On the other hand, Belden has some papers on their website proposing that their finest cable, Mediatwist, would
support even 100BASE-T and analog audio inside a shared sheath. Nevertheless, they offer the cable in only 4-pair
versions at this time.
Patch Panels
Patch panels come in versions for rack or wall mount and with varying numbers
of jacks. CAT-5/5e cables are punched down at the rear into 110-style insulation
displacement connectors using a tool very similar to the one that is used with
traditional “66 blocks.”
50 | Section 5
Wall Jacks
Again 110-style IDC connectors terminate the cable. Then these wired-up “Keystone” RJ-45 jacks are pushed
into a hole in the wall plate to complete the job. The diagram presented here shows the simple steps involved in
terminating these.
CAT-6 Jacks
CAT-6 cables and their accessories need more care to maintain the twists as
close as possible to the end. At left is a high-end CAT-6 jack assembly ready
for installation into either a rack-mounted patch-field or a wall jack. This is a
shielded version, so the shell is made from metal to maintain the shield all the
way to the edge of the jack.
Next, you can see the components that make up the jack disassembled. This is
now the non-shielded version, so the shell is plastic, the blue piece in the photo.
At the lower left is a closer look at the part that holds the wires. Assembling
one of these can be done in a minute or two. First the wires are put into the
slots and the ends are trimmed. Then this piece and the front part of the jack
are pushed together. The shell is then placed over these pieces and pushed over
them, which draws the wires into the insulation displacement forks and locks
everything together.
Architecture Options
There are a lot of ways to build a Livewire+ network. For many people a simple one-switch layout will be perfectly
sufficient. Others will want to build sophisticated networks to support multiple studios and perhaps hundreds of
audio channels. Fortunately, Ethernet scales easily – and therefore so does your Livewire+ installation.
Here are some examples and ideas to get you started.
Simple One-Switch Network
Common 1U switches can have as many as 48 ports. That is a lot of audio! Most modern switchgear has also
migrated to all-Gigabit with 2- or 4-port SFP (Small Form-factor Pluggable) ports for fiber or copper connections.
Here’s a setup that supports an on-air studio and a production studio.
There is the microphone version of the xNode in the on-air studio and the 4x4 analog line version in the rack. The
production studio connects with the xSelector node, which has one send channel and a selectable receive channel.
The Element power supply (not shown in the diagram) includes plenty of GPIOs for starting CD players, lighting
on-air lamps, remote mic on-off, etc.
The StudioEngine connects with a 1000BASE-T copper link to one of the two 1000BASE-T SFP ports with a
standard copper transceiver module.
The delivery PC connects directly to the audio network with the Axia IP-Audio Driver software. Control for it may
be directly over the network or could be with a hardware parallel connection. Servers and additional PCs can be
connected to the switch for file storage and delivery systems.
Peripherals such as codecs, telephone systems, and satellite receivers may be connected into the network wherever
it is convenient. In the diagram, the Z/IP ONE IP codec and Telos phone system are shown attached directly to
the switch; many Telos Alliance devices, as well as those of other manufacturers, can be attached directly with
Livewire+ connection ports. Equipment without Livewire+ can be attached directly to an xNode, as is the satellite
receiver in the drawing.
You could expand this to two Elements and Engines to support two studios since the switch has two 1000BASE-T
ports. An all-Gigabit 100/1000BASE-T switch can support nearly as many studios as you want.
52 | Section 5
In another variation as seen here, the standalone switch can be replaced entirely with a PowerStation mixing
engine, which also reduces separate I/O device count by aggregating audio and logic IO with the network switch
into a combined mixing engine environment. The switch contained within PowerStation and QOR mixing engines
supports daisy-chaining of up to 4 devices without the need for a core switch, so smallish studio networks can
dispense with additional switches nearly altogether.
Daisy Chained Multiple-Switch Network
While one switch can support multiple studios, you would have a single point of failure. Here’s another approach
that gives each studio its own switch. This example uses three switches, one for each studio group. This layout style
could easily be expanded to any number of switches and studios.
The switches are connected together so that audio sources are shared. A 1000BASE-T link between the switches
allows hundreds of audio channels to flow from one group to another. With more than two switches you could have
a “circular backbone” with redundant Spanning Tree links (described below) between the switches.
Peripherals that are used in common could be plugged to any of the studio switches, or there could be a separate
switch to pick up such feeds. In the example here, one studio takes in a satellite feed for sharing among studios;
another hosts a Telos VX multi-studio phone system base unit controlled in other studios by VSet phones. An
Omnia VOCO 8 multi-mic microphone processor is likewise shared.
54 | Section 5
Hierarchical Multiple-Switch Network
This is a layout that could support a very large facility. Dual-redundant Gigabit switches are at the center, and
100/1000 switches are used at the edge with one for each studio or logical group.
Note that the studios themselves are using a mix of standalone switches and Axia PowerStation and QOR mixing
Note that the studios themselves are using a mix of standalone switches and Axia PowerStation and QOR mixing
engines with built-in switches, each connected back to the central switching core. In this way, with each studio
coupled to an individual studio switch, there is no single point of failure for any studio since each operates as a
separate, independent facility.
Gigabit links are used between the edge switches and the center. These could be copper or fiber with a suitable switch.
The physical location of the switches is a matter of taste and trade-off. Putting the edge switches near the studios
saves cable runs, but locating all the gear in a central room simplifies engineering activities. So this is a quite
reasonable-cost option that provides a lot of power, flexibility, and expandability. Dozens of studios and thousands
of audio channels are possible.
Options for Redundancy
Ethernet switching has a built-in scheme for redundancy, called Spanning Tree and standardized as 802.1D. A newer variant is called Fast Spanning Tree. Switches with spanning-tree enabled exchange information with each other
about the topology of the overall network. You can have redundant backup links that are automatically activated in
the case that a main link has failed. Depending on the switch and layout, it could take as little as a second or as much
as a half-minute for a redundant link to be connected.
Link aggregation (sometimes called port trunking) is another method. With Spanning Tree, even if you have
two links between two switches, only one of them at a time will be active. But, it’s often better to have both active
simultaneously because you get twice the bandwidth during normal operation and instantaneous backup should
one fail. The link aggregation standard is 802.3ad. To use it, you usually have to specifically enable it on your
switch. Incidentally, this is supported on some PC network interface cards intended for servers, so it’s not only for
switch-to-switch links.
Most Ethernet switches offer a redundant power supply option.
We’ve been talking here about automatic on-line redundancy, but there is also manual swap-out as a reasonable
option. Because RJ-45s are so easy to unplug and re-plug and because switches and other Livewire+ components are
much cheaper than traditional alternatives, you can have spare units on the shelf for fast substitution.
Fiber optic links can extend the range of Ethernet. Because they are not subject to crosstalk and magnetic interference, they also can solve problems that might crop up in difficult locations with copper cables.
External media converters can be very simply plugged to Livewire+ nodes and switch 100BASE-T ports to convert
copper connections to fiber.
When choosing fiber connectors, you probably expect something with “multi” in the name to have more capability
than the same thing designated “single”. But this is not the case with fiber optics: single-mode cables are better
and more expensive than multi-mode. These names refer to how light is contained within the fiber. Single-mode
strands are smaller and more carefully control the light so that it doesn’t bounce around so much inside, thus are
more efficient and permit longer ranges.
This Gigabit unit from Allied Telesis uses 1000Mbps SC
single-mode fiber for up to 75km range. Units supporting
ST multimode fiber, for 100BASE-T links, are also available
for runs up to 2km. Many are available with PoE capability
to power nearby peripherals.
Modern Ethernet switches often have the option to plug a
media converter directly into a special socket so that fiber
may easily be connected from switch to switch. This is useful to make high capacity backbone links without any
external boxes.
56 | Section 5
Here is a typical case. The Axia xSwitch shown above at left has two “uplink” ports for use with 1000BASE-T SFP
(Small Form-factor Pluggable) fiber or copper transceiver modules. The device to the right of it is a typical modern
media adapter in the “SFP/mini-GBIC” size – about the same in width and height as an RJ-45 jack. They come in
different flavors, for 1000BASE-SX, 1000BASE-LX, etc. Generally, SX cables have a range to 500 meters, LX to 5km,
and LH to 70km.
Radio Links
There are Ethernet IP radios with surprisingly high bandwidth – and at surprisingly low cost. Not all units are
capable of achieving true Ethernet performance in terms of error rates, so some caution is in order. Most of these
operate in the unlicensed ISM bands, but with modern spread-spectrum technology and elevated directional
antennas, interference doesn’t look to present much problem. Licensed radios following the IEEE 802.16 “Wimax”
standard are easily available.
Bitrates range to 150 Mbps and distance to 25 miles depending on power level,
antenna, and terrain.
For studio-to-transmitter link, remote pick-up, and studio-to-studio applications,
these offer multiple channels of uncompressed audio, two-way transmission, and
the ability to multiplex VoIP telephone, remote control, and general data. When
audio and general data are mixed, the Ethernet switch provides the prioritization
function. As with all Livewire+ elements, you can check them with a web browser on
a network-attached PC.
We have studied many of these Ethernet IP radios, testing for Livewire+ compatibility
and general performance. You should consider these like the Ethernet switch – please
let us advise you on the best choice and help with your application.
Designing For Security
You will have 100% security if you keep the Livewire+ system completely isolated from any other network, local or
wide area. Those very concerned with protecting the studio system may well want to take this approach.
But there are advantages to sharing with or linking to an office network. You can configure and monitor the system
from any connected PC and audio can be monitored on any desktop with access. In this case, separate switches or
VLANs (described later) can be used to provide isolation. An IP router passes only the correct packets from one to
the other and thus provides a firewall function.
The next step up in connectivity would be to have a network linking co-owned or otherwise affiliated stations. In
this case, a network engineer is probably in the picture and he can take the necessary steps to protect your audio.
Connection to the public Internet brings the advantage that you can monitor and configure from a remote site, but
you now have much risk from unwanted intruders, viruses, etc. A qualified network engineer should be consulted
to be sure you have an appropriate firewall and other protections in place.
In Livewire+ nodes, web and Telnet access are password protected to provide some measure of security. But we do
not use exotic techniques like SSL (Secure Sockets Layer), so please understand that our devices were not designed
to be exposed to the public Internet without external protection.
The Ethernet Switch
This is what makes it all possible. Here are some details on requirements for a capable Livewire+ switch.
Livewire+ streams are “routed” at Layer 3 of the Ethernet model. We recommend a “Layer 3 switch” that includes
the required IGMP Querier and snooping functions. Switches range from very simple to amazingly elaborate.
Livewire+ Ethernet Switch Requirements
♦♦ Sufficient backplane bandwidth. It is required to be fully “non-blocking” to handle all ports at full capacity.
♦♦ Sufficient frame forwarding rate. Livewire+ Livestreams, as well as AES67 streams, have small packets at
a fast rate. The switch needs to handle this.
♦♦ Correct handling of IEEE 802.1p/Q frame prioritization. Livewire+ audio frames must be given priority
without too much delay or jitter. The IEEE standard specifies 8 levels of priority, but few switches support
all the levels. Many support only 2 or 4, lumping some of the incoming levels together. We recommend 4
as the minimum for a Livewire+ system.
♦♦ Support for multicast, with sufficient filter entry capacity to cover the total number of audio streams you
need. This latter is important, because when the filter capacity is exhausted, switches forward multicast
packets to all ports, subscribed or not. This would cause serious problems. You will probably want 256
♦♦ IGMP control for multicast. Traffic must be under IGMP control – strictly no flooding of ports with
multicasts under any circumstances.
♦♦ Support for both port-based and tagged-frame-based VLAN. This latter is the IEEE 802.1Q standard and
is what allows the switch to determine priority on a frame-by-frame basis. Port-based VLAN can also be
useful: it lets you “hardwire” a particular port for a single VLAN, useful to be 100% sure an office PC can’t
get onto the Livewire+ audio VLAN.
♦♦ If you will use a separate VLAN for Livewire+, the switch needs to have an “IGMP querier” on each
one, which also means that you can assign an individual IP number to each VLAN. This is an advanced
capability and its absence disqualifies many switches.
♦♦ Management. This is how you get remote monitoring.
The practical bottom line is that you should use a switch that has been selected and tested by the Telos Alliance.
When we check a switch, we use a laboratory setup that lets us send frames on a number of ports at a high rate,
while switching channels on/off with IGMP, etc. We have a lot of experience with different switches and know
what to look for. Using a recommended switch will also help you when you need customer support because we will
be familiar with it. You are encouraged to contact Telos Alliance Customer Support or visit www.TelosAlliance.
com/switches for our full list of currently supported switches.
58 | Section 6
Some Switches We Like
There are new switches introduced everyday it seems, with ever increasing performance and falling prices. Please
check our website for our latest recommendations.
Cisco Switches
Having said that, the Cisco Catalyst 6500, 4900, 4500, 3560X,
and 2960 series switches are some of the units that are fully
compatible and qualified for use with Axia. There are literally
hundreds of models that can be used with your Axia networked
audio system. 24 10/100BASE-T ports, 48 10/100BASE-T ports,
with or without 1000Mbps SFP (copper/fiber) transceiver ports
— connect them together and have virtually unlimited flexibility
and redundancy. All of these units include built-in simple
router and IGMP Queriers on every VLAN. They also come in a
powered-port version supplying IEEE 802.1af PoE (Power over
Ethernet) voltage that can be used with VoIP phones such as the
Telos VSet, or even to power Axia xNodes (which are capable
of running on either AC mains or PoE, cutting down on power
Axia xSwitch
The Axia xSwitch is a half-rack 1RU unit with the same form factor as our xNodes. Because xSwitch is designed and
built by Axia, it is optimized for Livewire+ IP-Audio applications, comes pre-configured and needs virtually no
setup. Fast setup requires only IP address assignment, accomplished in much the same manner as that of an xNode,
via the front-panel display or Web management interface. It is an advanced switch designed specifically for AoIP
network deployment:
♦♦ Eight 10/100MBit Ethernet ports — 4 with PoE .
♦♦ Two Gigabit for trunking, both with RJ-45 (copper) and SFP (fiber) connections.
♦♦ Supports redundant copper/SFP Gigabit connections with auto-switching.
♦♦ Has 2,000 Multicast groups and 2,000 ARP table entries (8x more than other small-form Ethernet
xSwitch is an excellent choice for either main switches in small studio installations, or edge switches in
larger facilities.
Switch Configuration
Most switches offer three connection options: an RS-232 console port, Telnet over Ethernet, and web over Ethernet. For Axia-supported switches, we offer a configuration “cheat-sheet” that gives you the basics. We also will be
happy to pre-configure your switch and test it at the Telos Alliance before shipping it to you. There are two different
software images that Cisco switches can be configured with. They are available with either the standard multilayer
software image (SMI) which includes advanced QoS, rate-limiting, ACLs, and basic or the enhanced multilayer
software image (EMI) which includes a richer set of enterprise-class features and advanced hardware-based IP
unicast and IP Multicast routing as well as policy-based routing (PBR). We do not expect or require anyone to be
Cisco Certified Network Engineers so we provide you everything you need to know about configuring your network
switch based on the software image of your unit. Configuration documents can be found on our website.
Testing, 1-2-3…
There are tens of thousands of people installing Ethernet networks every day, and many millions of working
installations. So there are a lot of tools to help you. Livewire+ equipment have a lot of diagnostic functions built-in
as well.
General Ethernet Troubleshooting
Ethernet is a mature technology, with years of proven reliable service. You are not very likely to see problems in the
fundamental technology if you follow the network wiring and layout recommendations.
The best way to avoid downtime is to build the network well in the first place. Use high-grade cables, good quality
factory-made patch cords, etc. And be careful with the punchdown and plug installation.
More on the topic of patch cords: If you really must make your own, they should be built with stranded wire cables.
Solid conductors are likely to crack when flexed a lot, usually right at the RJ-45 plug. From this you can get intermittents and bit errors. Also, as mentioned in the cabling section, be sure you have the right plug for the cable you are
using. An RJ-45 plug designed for stranded wire will cut through a solid conductor.
But you know all that. So let’s get on to troubleshooting, when despite all due care something still goes wrong.
The Basics
Link Test
A layer 2 test, this checks the connection between the switch and a designated network device on the same LAN.
During the link test, IEEE 802.2 test packets are sent to the designated network device in the same VLAN or
broadcast domain. The remote device must be able to respond with an 802.2 Test Response Packet. Most switches
support this test via a web or command line interface.
A layer 3 test, a simple and effective way to check basic “reachability” of an IP-enabled device. Ping sends a test
packet to a device and waits for an echo response. A Windows PC can do this within the command prompt window.
Just enter ping x, where x is the IP number or the domain name (if a DNS server is available) and see the result.
If you get the echo, the basic connectivity (including Layers 1, 2, and 3) is OK. Most switches and almost all
IP-enabled devices support this test.
TESTING, 1-2-3… | 61
Switch Diagnostics
Ethernet switches have many diagnostic tools, ranging from front panel LEDs to sophisticated software monitoring functions. See the switch manual and software description for details for your unit.
Simple Network Management Protocol (SNMP) and Remote MONitoring (RMON) are part of the TCP/IP
internet suite. (RMON is built on SNMP so they are closely related.) They offer a way to probe and monitor
network equipment operation in a vendor-independent way. For example, an Ethernet port has a standard way of
communicating its status that is supposed to be used by all products with these ports.
Almost all sophisticated Ethernet switches offer these, and they are useful tools to monitor traffic, check operation,
etc. You can do a lot of this with web and Telnet based communication but SNMP usually offers a deeper look.
You will encounter the acronym MIB, for Management Information Base. This is how information is organized
within SNMP. Cisco MIB files are available as free downloads from their website.
To use SNMP and RMON, you will need a software application that presents the information. There are many
freeware, standalone programs that can be used for this purpose; Cisco themselves offer an online tool, SNMP
Object Navigator, available at .
A full discussion would be too much for this document, but there is a lot of info that comes with Ethernet switches,
and a lot more in bookstores and on the web.
Some Things to Check
♦♦ Switch configuration must be correct. IGMP must be switched on, VLAN parameters set if you are using
them, etc. In our experience to date, this is the most common cause of problems. (With the exception of
cables, of course.)
♦♦ Ethernet links can be 10, 100, or 1000 Mbps, and full or half-duplex. We always want the maximum rate
and full duplex. You can configure the Ethernet ports on some devices for specific modes – but you should
not do this. The Auto mode is the correct setting, which will cause the device and node to automatically
negotiate to the appropriate condition. If you manually set the mode to full-duplex, the switch – in
compliance with a flawed IEEE standard – will set itself to half-duplex (!), leading to many problems.
Livewire+ hardware nodes are always set to the auto mode, so this problem will not arise with them, but
with other equipment such as PCs.
♦♦ If you want and have multiple redundant links using port trunking or spanning-tree, you have to set up
the switch to support them. Taking the default will usually not work.
♦♦ The “activity” LEDs (usually amber or green) on many network cards and switches will be on continu-
ously when any Livewire+ audio streams are present on the link. That is because the logic that drives the
LED extends the on time so that you can see it with normal traffic. Livewire+ packets are traversing the
network at such a fast rate that the LED never has a chance to turn off.
♦♦ Most Axia Livewire+ hardware nodes have status displays. They provide useful information and should
be checked. This is covered later in this section.
62 | Section 7
Cable Testers
“It’s the cable – it’s always the **@@ cable!” said my first boss. About half the time, he was right. That percentage
is probably a bit higher in Ethernet systems. Indeed, a number of surveys have put the “network medium” to blame
70-80% of the time. This being the cables, connectors, and hardware components that make up the signal-carrying
portion of the installation.
Wiggling and unplug-plug operations are legitimate and effective troubleshooting methods. But there are plenty
of cable testers to help you perform more elaborate checks. These range from simple conductivity testers to
sophisticated units that test cables for adherence to the TIA/EIA standards, detect breaks with a Time Domain
Reflectometer, and more. Contact info for the main manufacturers of these are listed in the Resources section.
The testers shown here represent something of the range available.
The Fluke DSX CableAnalyzer™ Series can certify that your cable meets the appropriate category
requirements with regard to crosstalk, attenuation, etc. and perform a number of sophisticated
tests. An adapter can be changed to allow the unit to work with both copper and fiber cable types.
The Fluke MicroScanner™ Cable Verifier is a much simpler and cheaper variant from Fluke that
checks for conductivity and correct wiring. It can also tell you the distance to a break with a TDM
function and can do tone trace with an optional remote unit.
Finally, the Byte Brothers LAN Tester TVR10/100/1000™ 2-piece set is a basic wiring tester and
tone line-finder.
These are software applications that run on PCs and can listen-in on the packets flowing
on an Ethernet link. Usually used in conjunction with an Ethernet switch’s port-mirroring function. This lets a designated monitoring port mirror that traffic on any other port
you select. Livewire+ audio packets are small in length and very frequent compared to
general data traffic so are quite challenging for a sniffer. To be useful, you will need a
good one and a fast computer to run it. Very useful, but expensive. Perhaps best
borrowed from your company’s network guys’ kit.
TESTING, 1-2-3… | 63
Diagnosing Problems using Livewire+ Components
All Livewire+ components have built-in diagnostic tools. For example, audio nodes have a loop-back testing
procedure that measures audio noise and distortion. The web interface lets you check a number of internal values.
The xSelector is a useful device for displaying available audio streams and listening to them. It has one channel of
send, so is useful as an audio source injector as well.
The Livewire+ IP-Audio Driver has a diagnostic window that tells you a number of things about the system clock
and audio streams.
xNode Indicator Displays
xNode displays have indicators of the status of Livewire+ and Ethernet connections, as well as system synchronization as follows:
LINK and ACTIVITY LEDs on each Ethernet and Livewire+ port. A solid amber LINK light indicates the presence
of a live Ethernet link to another Ethernet 100 Base-T device. A flashing green ACTIVITY LED indicates that the
connected Ethernet segment has Livewire+ traffic present. If the LINK LED is illuminated, and the ACTIVITY LED
fails to illuminate, it’s likely that the Ethernet switch has not been programmed to pass such traffic through to the
port to which this node is connected.
SYNC & MASTER indications in the upper right corner of all xNode
front-panel readouts. The Livewire+ system employs a sophisticated master/
slave clocking system over the Ethernet network. By default any device may
become the clock master, however this can be changed if desired. The system
has the ability to automatically change to a different clock master should the
current master become disconnected, or otherwise inoperable. This happens
transparently without any audio glitches.
♦♦ SYNC indicates the receipt of clock information from another (Master) Livewire+ node, and that the
local xNode’s PLL is locked.
♦♦ MASTER indicates that an xNode is acting as the master clock source for the Livewire+ network.
♦♦ If neither SYNC nor MASTER are displayed, something is not correct.
Network Engineering For
Audio Engineers
You don’t need to know most of what’s in this section to use Livewire+. Just as a beginner can plug analog XLRs successfully together without knowing anything about op-amps, you can connect and use Livewire+ without knowing
details about packets. But just as fixing tricky problems in the analog world calls for higher-level understanding, so
does an awareness of the internal technology of Livewire+ help you to solve problems and build complex systems.
This section introduces basic concepts – enough for you to get a feel for how data networks work and to understand
the lingo so you are ready to ask intelligent questions of network guys and vendors. It also explains a lot of points
specific to Livewire+.
Livewire+ is built upon standard components, so if you understand data networking generally, you’ll be ready for
the specifics of Livewire+ audio networking. Network engineering is a rich topic, abounding with information and
nuance, and in constant flux. Fortunately, Livewire+ uses only a small subset that is easy to learn and understand.
That is mainly because most of the complexity comes with IP routing and wide-area networks such as the Internet
– and we don’t use much of that, staying only with the much simpler Ethernet LAN level. Even if you don’t know
anything yet, you’ll get pretty much what you need in the next few pages. If you want to know more, bookstores
have shelves loaded with networking advice and information. We offer a few starting points in the Resources
As always, Telos Alliance support is at your side to help with any specific practical issues that may come your way.
If you are developing for Livewire+, this will offer only a brief introduction, and you’ll want to know more. Please
contact us for any of your needs, such as software API documents. We welcome partners!
Ethernet / IP Networks
Layering Model
You need to know layers to know networks. The notion of layers and
the open systems they support are central to network engineering.
Because layering is a key to enabling multiple vendors for each
function, this design has also been a major factor in the growth and
operation of the Internet. It’s also one of the keys to Livewire+,
allowing us to build our professional audio transport application on
existing standard lower layers.
For many years, the Open Systems International (OSI) model was the reference paradigm for data networking.
For example, the ISDN D-channel communication between nodes and the telephone network is loosely based on
this model:
Data Link
But this proved to be too complex for most practical applications, and an architecture has evolved that is simpler
than the OSI model. The next chart shows how that simpler model applies to the IP-over-Ethernet combination we
are using.
HTTP (Web), POP (mail), etc.
IP Routing
Ethernet Addressing
Ethernet Physical
Layer 1, Physical Interface: This layer is responsible for hardware connectivity, which is provided by Ethernet.
Layer 2, Ethernet and Switching: This layer is Ethernet’s end station addressing and everything related to it. An
Ethernet switch is working at Layer 2 because it forwards packets based on Ethernet Media Access Control (MAC)
addresses which are unique ID numbers assigned by the Ethernet-capable equipment manufacturer.
Layer 2 does not ordinarily extend beyond the corporate boundary. To connect to the Internet requires a router. In
other words, scaling a Layer 2 network means adding Layer 3 capabilities.
Officially, the transmission units comprising header and data are called frames at this layer. At Layer 3, the correct
designation is packets. But, since Ethernet frames are almost always carrying IP packets, the word used to describe
the combination most often depends upon the context or the author’s preference. Unless we are referring to layer 2
functions, we usually use “packets” because Livewire+ audio has the IP header – and because “packets” has become
the usual way to describe this sort of thing generally.
Layer 3, IP Routing: In addition to Ethernet addresses, each IP packet on a LAN also contains source and destination IP addresses. These were intended to be used by routers to forward packets along the most efficient route
and link LANs of different types. When the Internet was invented, there were dozens of LAN technologies in use
and this was an important capability. Now, IP addressing is used both within LANs as a way to access servers from
clients, etc, and to connect to Internet resources offsite.
IP in itself is not a particularly complex protocol, but there are numerous capabilities supplied by the other components of the IP suite. The Domain Name System (DNS) removes the burden of remembering IP addresses by
associating them with real names. The Dynamic Host Configuration Protocol (DHCP) eases the administration of
IP. Routing protocols such as Open Shortest Path First (OSPF), Routing Information Protocol (RIP), and Border
Gateway Protocol (BGP) provide information for Layer 3 devices to direct data traffic to the intended destination.
Layer 4, Transport: This layer is the communication path between user applications and the network infrastructure
and defines the method of communicating. Transmission Control Protocol (TCP) and User Datagram Protocol
(UDP) are well-known examples of elements at the transport layer. TCP is a “connection-oriented” protocol,
requiring the establishment of parameters for transmission prior to the exchange of data and providing error
recovery and rate control services. UDP leaves these functions to the application.
66 | Section 8
Layer 5, Application: Web browsers, audio editors, and email clients, for example. And our Livewire+ audio.
Applications developers decide on the type of Layer 4 transport necessary. For example, database or Web access
require error-free access and use TCP, while live streaming media use Real-Time Protocol layered on top of UDP/IP.
Making Packets
Livewire+ Standard Streams use all of the recommended Internet protocols and are constructed in the usual layered
fashion. Here is one representation of the packet structure:
In the next graphic, below, you can see this structure in more detail. This is the way network engineers usually
visualize a packet. It’s not important to know what each of the fields means; the idea is for you to see how a packet
is constructed generally. Each of the horizontal bars are 4 bytes. At each layer, devices are operating only with the
information contained within the associated header. An Ethernet switch only cares about the layer 2 headers and
everything else is just payload.
An IP router only “sees” the layer 3 header and doesn’t care about the lower-level transport. Applications don’t
care about headers at all – they just deliver their data to the network and expect to get it back at the other end
(There are, however, exceptions, such as sophisticated Ethernet switches that can inspect layer 3 headers for some
advanced functions).
IP and Ethernet Addresses
As with everything connected to IP/Ethernet networks, Livewire+ devices require both IP addresses and Ethernet
MAC (Media Access Control) addresses.
IP Address
IP addresses are four bytes long and are written in “dotted decimal” form, with each byte represented decimally and
separated by a period. For example, in the IP address, the 193 is the value for the first byte, 32 for the
second, etc. Since a byte can hold values from 0 to 255, this is the range for each decimal value. Host IP addresses
are assigned to your organization by your internet service provider and parceled out to individual host computers
by your network administrator. He may give you this number to be entered manually, or could opt for DHCP
(Dynamic Host Configuration Protocol) to let your computer get the address automatically from a pool. Because
Livewire+ devices are permanently attached and because it is more convenient to know the IP address attached
to a particular node and perhaps assign them in some kind of logical pattern, we do not support DHCP for our
hardware nodes. Therefore, you will need to enter an IP address into each node.
In addition to the address, there are a few more numbers to enter into an IP configuration:
Subnet mask
Subnets allow a network to be split into different parts internally but still act like a single network to the outside
world. There are three logical parts to any Internet address: the main network address, the subnet address, and
the particular device address. The mask marks the dividing point in the address between the subnet part and the
device (host) part. What is meant here by “network” and “subnet” depends on your internet provider. A network
in this context could mean all of the address space allocated to the provider, and the subnets could delineate the
individual customers. Or the network could be all the addresses allocated to a university or major corporation and
subnets could divide the address space to correspond to departments. Network addresses are assigned by IANA,
the internet names and numbers authority, while subnets may be changed without any official approval.
32-bit (4-byte) IP address space
The mask is written in the same dotted-decimal form as IP addresses. In the example a very large network supporting 64k hosts is divided into 64 subnets, each with 1k hosts. The subnet mask would be, which is just
another way of writing the binary ones and zeros value shown above.
As a practical matter, you usually just take the number given to you by you network administrator or service
provider and enter it.
Gateway address: This is the IP address of the device that passes traffic out of your local network to the internet.
This is usually a router.
DNS server address: This is the address of the computer that provides name look-up service, translating text
domain names like to IP address numbers.
68 | Section 8
In careful language, devices that attach to the Internet and have IP addresses are called hosts, a name that probably
made sense in the early days. (They “host” the IP stack and interface.) And Ethernet-connected devices are
officially called stations to keep the radio/ether analogy going. But what do you call something that is both host and
station, as almost everything being “Host” doesn’t sound very natural for our audio devices and “station” would
be very confusing, indeed. As you’ve noticed, we usually just say “Livewire+ node” in the context of our audio
equipment, which should be clear enough. They won’t be nodes, will they? Unless something better comes along,
we’ll probably say Livewire+ device. As to “host” and “station” for other devices, we’ll just use connected PC or
some variant, thank-you very much.
Ethernet Addresses and Address Resolution Protocol (ARP): Machines that use IP and are connected to an
Ethernet have two addresses, IP and Ethernet MAC. While the IP address is user-determined, the Ethernet MAC
address is usually programmed into the network card or interface by the manufacturer.
You will probably never have to deal with them directly, but who knows? Ethernet addresses are 6 bytes long
and are written in “dashed hexadecimal” form like this: 5C-66-AB-90-75-B1. (Sometimes colons are used as the
separators.) Hex notation is just another way to write binary values. Single digits range from 0 to 9, A, B, C, D, E,
F and byte values from 00 to FF. The value FF means all the bits in a byte are 1s and is equivalent to decimal 255.
While this notation may seem strange at first sight, it is very useful to programmers, who need to think in powers of
There is a unique Ethernet MAC address for each and every network adapter ever made in the world. IEEE handles
the allocation among manufacturers and each manufacturer is responsible to ensure that they make no two alike
within their assigned range. I (Steve) used to feel bad about all those wasted addresses from obsolete and thrownaway network cards – guess that’s the Protestant USA mid-westerner in me – but supposedly 6 bytes is enough that
each of Earth’s grains of sand could have its own address, so not to worry.
There is a need to translate between IP and Ethernet addresses. Consider a server sending data to a machine it
knows only by IP address. To communicate, it has to generate an Ethernet frame including the Ethernet destination
address corresponding to the desired IP address. To do this, every IP-based device has an ARP module, which takes
an IP address as input and delivers the corresponding Ethernet address as output. It maintains a local table with
the associations. When it encounters one it doesn’t yet know, it broadcasts an ARP query packet to every device on
the LAN and the device that owns the specified IP address responds with its Ethernet address. If there is no owner,
the packet is presumably intended for an off-site device and is sent to the gateway address of a router. How does the
transmitting device find the router’s Ethernet address?
With ARP, of course. Entering arp -a into Windows’ command prompt will give you the current list of IP addresses
and associated Ethernet addresses – the ARP table for that machine.
Multicast Addresses: All of the above discussion was only relevant to the usual Unicast situation that is used for
web surfing, emails, file transfers, etc. We also use it in Livewire+ for configuration and control, such as when a
web browser is connected to a hardware node. But audio is multicast because we want it to be available to multiple
destinations. The principle is simple: rather than specifying a specific destination, a special “virtual” multicast
address is used that is not assigned to any particular device. Audio nodes can listen-in in a party-line fashion by
receiving any packets at this address.
Our audio streams are multicast at both Layer 2 and Layer 3, using the set-aside multicast addresses at each layer.
The Livewire+ channel number is automatically translated to the appropriate addresses at both layers internally.
Livewire+ uses the IP address range starting from This choice is based on the assigned numbers from
the IANA (Internet Assigned Numbers Authority) allocation of this range for use within organizational and site
specific scopes. These addresses are to be used for multicast applications that are not used across the global Internet. Since our application will be used within a single facility on a single switched LAN, this range is appropriate.
Over 8 million unique IP multicast addresses are available with each address mapping to a globally unique Ethernet
multicast address.
Even so, IP is relatively stingy with its multicast space. Ethernet has set aside half of all destination addresses for
multicast - 140,737,488,355,328 addresses, which should be enough for even the very largest broadcast facility!
The designers clearly had big plans for multicast that have not yet been realized.
The distinction is made in the first transmitted bit of the 48-bit address that divides the total available address
space in two: a 1 in this position signifies a multicast.
Ethernet Switching
Ethernet switching has caused a revolution in data networking. With switching, each device owns all the bandwidth on its link. No sharing and no collisions. Incoming frames are forwarded only to the nodes that need them.
Despite their amazing power, the invention of switching was more akin to falling off a log than sawing one in two.
The switch builds up a table of what addresses are attached to what ports, which it does by merely by examining
the source addresses of sent packets. When frames come in, the switch looks into the table, discovers what port
owns the destination and forwards the data only to that port. In the rare case that no entry exists for an address, the
frames are “flooded” or broadcast to all ports to be sure the intended recipient gets it. If a connection is unplugged
or there is no data for a long time, the entry is removed. Pretty simple, eh?
The operation described above is for the common Unicast, or point-to-point, communication that you have for
typical traffic such as web, email, etc. But Ethernet supports three communication types:
♦♦ Unicast means point-to-point. The usual mode for traffic.
♦♦ Multicast means that multiple receivers may “tune in” to the transmission from a source so that a selected
subset of nodes is served.
♦♦ Broadcast means that packets are sent to all receivers, which is quite common on Ethernets. Microsoft
file sharing, for example, advertises the PCs on a network this way. ARP uses this to get a query to all
machines on the network.
We use multicast for Livewire+ audio streams because we want to emulate distribution amps and audio routers,
with multiple receivers being simultaneously able to listen in to a source. The automatic procedure described above
does not work for multicasts because they are not associated with a particular output port and node. Fortunately,
switches offer a way to control these one-to-many streams. A multicast Ethernet frame has a “virtual” destination
address that is just stopped inside the switch if there are no interested receivers. When receivers want to tune-in,
they send a message to the switch telling it to turn on the stream to their port.
The switch knows what frames are multicasts because the destination address belongs to the set-aside multicast pool.
Livewire+ uses one Ethernet/IP multicast address for each audio stream. These are derived automatically from
the channel numbers you assign. Streams are multicast at both Ethernet and IP layers using the assigned multicast
addresses at each.
IGMP (Internet Group Management Protocol): We need some way to tell the switch what streams go to what ports
– that is, a way to control multicast switching. IGMP was designed for just this purpose.
IGMP is part of the IP suite and is a Layer 3 function that was designed to communicate with IP routers to control
multicasts. But switch manufacturers started to implement “IGMP snooping” on the messages between hosts
(computers) and routers as a way to control multicasts at Layer 2. In recent switch implementations of IGMP, this
is taken further and a router is not necessary as long as a switch is configured to support IGMP with the “Querier”
feature enabled. We want this because there is often no router in the system. Even were there to be one, better to
have this capability in the switch as a back-up.
70 | Section 8
IGMP uses three types of messages to communicate:
♦♦ Query: A message sent from the querier (multicast router or switch) asking for a response from each
host belonging to the multicast group. If a multicast router supporting IGMP is not present, then the
switch must assume this function in order to elicit group membership information from the hosts on
the network.
♦♦ Report (Join): A message sent by a host to the querier to indicate that the host wants to be or is a member
of a given group indicated in the report message.
♦♦ Leave Group: A message sent by a host to the querier to indicate that the host has ceased to be a member
of a specific multicast group.
An IP multicast packet includes the multicast group (address) to which the packet belongs. When an IGMP client
connected to a switch port needs to receive multicast traffic from a specific group, it joins the group by sending an
IGMP report (join request) to the network. (The multicast group specified in the join request is determined by the
requesting application running on the IGMP client.) When a networking device with IGMP enabled receives the
join request for a specific group, it forwards any IP multicast traffic it receives for that group through the port on
which the join request was received. When the client is ready to leave the multicast group, it sends a Leave Group
message to the network and ceases to be a group member. When the leave request is detected, the appropriate
IGMP device will cease transmitting traffic for the designated multicast group through the port on which the leave
request was received (as long as there are no other current members of that group on the affected port).
Thus, IGMP identifies members of a multicast group and allows IGMP-configured hosts (and routers) to join or
leave multicast groups.
The function of the IGMP Querier is to poll other IGMP-enabled devices to elicit group membership information.
The switch performs this function if there is no other device, such as a multicast router, to act as Querier. The
switch automatically ceases Querier operation if it detects another Querier. A switch with IGMP querier capability
will become a Querier in the absence of any other Querier on the network. If you disable the Querier function on a
switch, ensure that there is an IGMP Querier (and, preferably, a backup Querier) available. If the switch becomes
the Querier, then subsequently detects queries transmitted from another device on the same VLAN, the switch
ceases to operate as the Querier for that VLAN. In the above scenario, if the other device ceases to operate as a
Querier, then the switch detects this change and can become the Querier as long as it is not pre-empted by some
other IGMP Querier.
In a Livewire+ system, it is the responsibility of the audio nodes to generate the IGMP messages.
Within a link, we sometimes want to have audio mixed with general data. This happens, for example, when a
delivery PC is playing audio and downloading a file at the same time, or when our mixing engine is sending and
receiving audio and control messages simultaneously. To be sure audio always flows reliably, we take advantage of
the priority functions that are part of the switched Ethernet system.
Compared to the original, modern Ethernet has an additional 4 bytes of data inserted into the frame’s header. One
field provides a 3-bit priority flag, which allows designation of eight possible values.
Priority Level
IEEE Recommendation
Livewire+ Assignment
Network Control
Livewire+ Audio
Telephone Audio
Video Conferencing
Call Signaling
High Priority Data
Medium Priority Data
Best Effort Data
Livewire+ Control & Advertising
Highest-priority packets have first call on the link’s bandwidth. If high-priority packets are in the queue and ready
to go, the lower-priority ones wait. If there is not enough bandwidth for both, low-priority packets will be dropped
– but this is not a problem, as you will soon see.
The adjacent graphic shows only two queues, but the idea is the same for four or eight. Switches used for Livewire+
must support a minimum of four queue and priority levels. Some low-end switches have no support or may have
only two queue levels.
If you have multiple switches in a hierarchical configuration, the priority information is carried automatically to all
the switches in a system.
This prioritization scheme works only within a facility’s local area network. Because it is at the Ethernet layer, it has
no effect past the router boundary into the internet. However, we also set the priority bits in the IP header to match
the Ethernet priority so that as LAN switching evolves to use more Layer 3 intelligence, our packets will be ready.
72 | Section 8
The Role of TCP
TCP is a key to sharing high-priority audio with best-efforts data on a single network link. Because the acronym
TCP/IP is so often written, many people think that they are necessarily and always joined. This is certainly not so.
IP is independent from TCP and may well be used without it. For example, RTP/IP is specified for streaming media,
and UDP/IP is used for a variety of transmissions, such as DNS, the Internet’s name look-up service.
TCP has two functions: Ensuring reliable transmission and controlling transmission rate.
Routers and switches may drop packets when there is not enough bandwidth to transmit them or when they are
They also do not guarantee to deliver packets in the same order as they were sent. And there is no protection for
bit errors from signal corruption. None of this is a mistake or oversight in the design of the Internet. The inventors
knew what they were doing: they wanted control of any needed correction process to be as close as possible to the
endpoints, consistent with the general internet idea to move as much as possible from the center to the edges.
Certainly we need 100% reliable transmission for most data files – even a missed bit could have bad consequences.
TCP gets this done by using a checking and re-transmission approach. Whenever TCP detects any corrupted or
missing data, it requests another copy to be sent and holds any data it might already have in its queue until the
replacement has arrived. Packets are numbered by the sender so that they can be delivered to the application in
correct order. The application always gets good data – but it could be after significant delay.
Transmission rate control is essential for most internet applications because the bandwidth of the many transmission “pipes” from sender to receiver are almost always different. And the available bandwidth to a particular user is
constantly changing as the demands from the many users sharing the net ebb and flow. Think of the common case
that you are working at your laptop with a cellular data modem, connected by VLAN to your office server. The server and its local network can certainly send data faster than your modem can take it. And the available bandwidth
on the public part of the net is varying. So something needs to slow the sending rate to match both the network and
your modem’s ability to receive. That function is performed by TCP. This is called flow-control. While the details
are complicated, the principle is simple: a TCP sender monitors the condition of the buffer at the receiver so it
knows how fast the data is arriving and can adjust its transmission rate to maintain the correct average buffer fill.
TCP also has a function called congestion-control. While it also controls rate, it does it with a different mechanism
and for a different reason. The re-transmission procedure we discussed earlier addresses a symptom of network
congestion, but not its cause – too many sources trying to send at too high a rate. To treat the cause of congestion,
we need to have some way to throttle senders when needed.
TCP’s congestion control is unusual in that it is a service to the network at large rather than to the individual user. It
was conceived as a way to fairly ration network bandwidth to all users. To do this, TCP monitors dropped packets,
assuming that lost packets signal congestion. When a new connection is established, a slow-start function causes
the rate to start low and ramp up until a lost packet is detected. Then the rate is cut in half and the ramp up begins
again. In this way TCP is always probing for the maximum available bandwidth and always adjusting its transmission rate to match. Its really a very slick technique, one that is very well suited to getting the fastest transmission of
bursty data over a shared link.
We’ve gone into a lot of detail on TCP because it is one of the keys to being able to share Livewire+ audio over a
network link with other general data. The Ethernet switch handles congestion in a similar way to the routers in the
internet – when there is too much traffic, it drops packets. But we have something very important: Priority. Audio
packets are assigned higher priority than general data. So they are never dropped before all TCP packets are. The
usual condition is that some percentage of the link is filled with constant audio streams and the remaining capacity
is left for data. For example, an 8-audio channel Livewire+ node with all channels active will take about 40% of its
100BASE-T link, leaving 60% for data. But, we could have one or we could have a dozen audio streams active on a
link – and this number could well change over time. TCP automatically finds how much bandwidth it can use and
adjusts its rate naturally to match.
You might be thinking, “All well and good, but what if we put too many high-priority packets into the link? Won’t
we have drop-outs then?” Yes, we would. But we never allow this to happen. Remember that each Livewire+ node
knows about the link attached to it because it “owns” it. The link from a node to a switch is full-duplex point-topoint with no sharing. The node knows how many streams can fit and never is allowed to send more into or request
for reception more than can be supported by the link.
All of the above applies to a shared link, such as for a delivery PC that needs both audio and data. It is the Ethernet
switching function that allows the overall network to be shared, since general data never even gets to a port
connected to a Livewire+ node.
Virtual LANs (VLANs)
This is a technology that came to Ethernet along with switching. It is a way to have “virtual” isolated LANs, while
using common hardware.
Remember those Broadcast packets? They go to all devices, even with an Ethernet switch in the picture. If there are
a lot of computers on the network, there could be a lot of traffic generated by these transmissions. VLANs can be
used to contain broadcast packets, since they are not propagated outside of their assigned VLAN.
VLANs can also be used for security. If the Livewire+ network is on a different VLAN than the internet, a hacker
would not be able to gain access to your audio streams or send traffic on the audio network.
In a Livewire+ network that is shared with general data, VLANs offer protection against a computer that could have
a problem with its network software or interface card. The Ethernet switch can be configured so that the ports to
which general computers are connected are not able to forward packets outside of the assigned VLAN, so can never
reach Livewire+ audio ports.
Finally, VLANS protect against the rare case that an Ethernet switch has not yet learned an address and has to flood
all ports until it knows the specific destination.
All Livewire+ devices allow choice of VLAN. We recommend:
♦♦ If you have a separate network for Livewire+ audio, you can just stay with the default VLAN 1 and pay no
more attention to this topic.
♦♦ If you have your Livewire+ network connected to the Internet, or shared with a large group of office
computers, use the default VLAN 1 for general data and VLAN 2 for Livewire+ audio and control.
A router must be used to bridge the traffic between VLANs, while providing a “firewall” function. So if you have
PCs on the Livewire+ network that will be used for audio and web surfing, etc, you will need to provide this bridge.
You will also need this to access Livewire+ devices on VLAN 2 with PCs connected to VLAN 1 for configuration,
monitoring, etc.
A router that bridges VLANs is sometimes called a “one-armed” router because it has only one Ethernet port, rather
than the usual two. But you can use the same router that you have for your Internet link to provide this function.
Or maybe better: Some sophisticated Ethernet switches provide an internal routing capability that can be used to
bridge VLANs. Simpler, and saves boxes.
74 | Section 8
Tagged vs. Port-Based VLAN Operation
When the VLAN information embedded in the Ethernet frame is used to direct the switch, this is called tagged
VLAN operation. With Livewire+ devices, when you configure a VLAN value, the device will transmit Ethernet
frames with the embedded value you specify. But some devices are not able to do this. That means the switch itself
has to insert the tag – a procedure called port-based VLAN. In this case, all frames that enter from a particular port
are tagged with a certain value, defined by switch configuration. To enable this, you must configure the switch
In Windows 7, you can install Advanced Network Services to get VLAN support; it should also be available, when
using compatible NICs, in Windows 8 and Windows Server 2012. Even with this capability, however, it is still more
efficient to handle VLAN via the switch. Because the PC uses an Access port, we simply assign ‘switchport access
vlan 100’ (for example) and then the PC participates in VLAN 100 with no settings needed on the PC.
In Windows 7, you can install Advanced Network Services to get VLAN support; it should also be available, when
using compatible NICs, in Windows 8 and Windows Server 2012. Even with this capability, however, it is still more
efficient to handle VLAN via the switch. Because the PC uses an Access port, we simply assign ‘switchport access
vlan 100’ (for example) and then the PC participates in VLAN 100 with no settings needed on the PC.
There is one special case: Frames tagged with VLAN=0 are called priority frames in 802.1p standard. They carry
priority information, but not the VLAN ID. The switch will translate to whatever VLAN is default for that port.
This is useful if you want to use port-based VLAN assignment at the switch, rather than tagging from the Livewire+
Many switches allow a combination of port and tagged VLAN. In this case you assign a default value to the port and
frames either with no tag or with tag=0 go to this default VLAN, while tagged frames override the default.
It would be possible to use port and tagged VLAN in combination. For example, you use Livewire+ node configuration to put all your audio devices onto VLAN 2. Since the debut of Windows 7, there has been VLAN support inside
Windows, but an easier method is to use your switch’s port-based assignment — simply set a port to be always
VLAN 2, then plug your PC into this port.
Some switches have other options for assigning VLANs. Assignment could be “hard-coded” to MAC addresses with
a configuration set-up. Or layer 3 protocols (TCP, RTP, etc) could be detected and used as a way to make VLAN
assignments. These may have their place, but since Livewire+ devices provide the tagging, it doesn’t seem that these
methods make much sense. The less you have to configure the switch, the better.
Ethernet Switching versus Routing
Both switches and routers examine packet addresses and send them appropriately on their way. So what is the
relationship between these technologies? Why and where would you use one versus the other? Routing works at
Layer 3, where IP information resides, while Ethernet switching works at Layer 2. Routing is a much more complex
operation than switching, where multiple paths from one site to another are the norm, and it is the job of the router
to find the optimum route (get it?), which may well be changing from minute-to-minute. A comparison of the two
Ethernet Switch vs IP Router Comparison
Layer 2/Ethernet
Layer 3/IP
Determines to which port the addressed node is
connected and switches incoming frame to it
Finds the best route from among many and
forwards packet to next router along the path
Simple table look-up in hardware
Complex dynamic best-route determination in
Many, connecting mostly to end nodes
A few, connecting to networks and Telco lines
As do switches, routers also support multicast and prioritization. But routing is a much more precise and efficient
way of controlling destinations for our mission-critical audio. This is why Livewire+ networks user Layer 3 routing.
Traditionally, routers did their work with software, while switches had dedicated hardware chips. Now there is
something called Layer 3 Switch, a hybrid of traditional routers and Ethernet switches. Layer 3 switches perform
their forwarding – whether Layer 2, Layer 3, unicast, multicast, or broadcast – in hardware. Software handles
network administration, table management, and exception conditions.
Since we first introduced studio AoIP in 2003, the cost of such devices has fallen dramatically, making it even more
economically feasible to base studio networks on Layer 3.
Livewire+ Networks
So now we are ready to consider all that has gone before in the context of Livewire+. And to begin the discussion
technologies specific to Livewire+.
Quality of Service (QoS)
An important concept in a converged network is Quality of Service. When general data is the only traffic on a
network, we only care that the available bandwidth is fairly shared among users and that the data eventually gets
through. But when our studio audio and general data are sharing the same network, we need to take all the required
steps to be sure audio flows reliably.
Our method for achieving QoS is system-wide, with the following components each contributing a part of the whole:
♦♦ Ethernet switch. Allows an entire link to be owned by each node. Isolates traffic by port.
♦♦ Full-duplex links. Together with switching, eliminates the need for Ethernet’s collision mechanisms and
permits full bandwidth in each direction.
♦♦ Ethernet Priority assignment. Audio is always given priority on a link, even when there is other high-vol-
ume non-audio traffic.
76 | Section 8
♦♦ IGMP. Ensures that multicasts – our audio streams – are only propagated to Ethernet switch ports that
are subscribed.
♦♦ Limiting the number of streams on a link. Audio nodes have control over both the audio they send and
the audio they receive, so they can keep count and limit the number of streams to what a link can safely
The result is rock-solid QoS, combined with the ability to share audio and data on the same or interconnected
Source Advertising
Audio source nodes advertise their streams on a special multicast address. Receive nodes listen to these advertisements and maintain a local directory of available streams. The advertisements are sent when the streams
first become available and at 10-second intervals after that. (Actually, only the data version number is sent every
10 seconds. The full data is advertised only upon entering the system, on any change, and on explicit requests
from those having detected the data version number increase.) If a node’s advertisements are not received for 3
consecutive periods, it will be assumed to be removed from service. There is also an explicit “stream unavailable”
Receive nodes maintain a local table of available streams and their characteristics, updated as any new information
arrives. Sources are cleared from local tables when an explicit message is received announcing that a stream is no
longer available, or when three consecutive advertisements have been missed.
A receive node may be configured to be permanently connected to particular multicast streams, or users may select
audio sources from a list. The list may display all available sources, or a filtered subset.
You may ignore this matter completely – and your Livewire+ system will work automatically “out of the box”. But
there are times when you might want to modify the default behavior of the clock sync system, so here is some detail
on how the system works.
Livewire+ (and AES67) streams need careful system-wide synchronization in order to have small buffers for
low-latency streams. If we did not have a distributed way to derive a bit clock, we would eventually have buffer over
or under-flow, resulting from the input and output node clocks being not exactly the same frequency.
A PLL (Phase Lock Loop) in each Livewire+ node recovers the system clock from multicast clock packet that is
being transmitted at a regular interval. At any given time, one Livewire+ hardware device is the active system clock
master. In the event the master develops a fault or is removed from service, the local PLLs in the nodes are able to
“ride out” the brief interruption and there will be no problem with operation.
All nodes are capable of being a clock source, and an arbitration scheme ensures that only the unit with the highest
clock master priority is active. Clock mastership priority may be set by the user, or left to the default case of all
being equal priority.
When the clock goes away for 3 consecutive periods, all capable units begin transmitting clock packets, after a delay
skewed by their clock mastership priority.
When a unit sees clock packets from a unit with a higher mastership priority on the network, it stops its own
transmit of clock packets.
You can specify the clock mastership priority behavior. The clock mastership can be made predictable, rather than
end up being any node in the plant – maybe the one down in an out of the way equipment closet.
Each node has a clock mastership configuration setting that can range from 0 to 7.
♦♦ ‘0’ means never - slave only
♦♦ ‘7’ means always - forced master (don’t use multiple forced masters in a system)
♦♦ Factory default is 3. So all units have equal priority out of the box, and the following is used to break
ties (in descending order): lowest Livewire+ audio transmit base channel, then lowest IP address, then
lowest Ethernet address.
Livewire+ xNodes have a status display on their front panel that shows “MASTER” when that unit is the clock
master; else the unit displays “SYNC” to indicate that it is a slave synchronized to the master clock.
Jitter in the timing and PLL functions ultimately set a lower bound on output buffers and therefore audio delay.
And any drift in the time calculation produces buffer pointer drift. Further, jitter in the derived A-to-D and D-to-A
bitclocks causes sampling uncertainty that can generate unwanted noise in the audio.
The LAN network is a “noisy” environment with packets of various kinds and lengths being numerous and
unpredictable. Thus, the PLL system needs to be quite smart so as to generate a reliable, consistent, low-jitter
output, while not being confused by dropped or jittered time packets.
Our method for handling this PLL problem is the subject of a patent, to give you some idea of the novelty and
complexity involved in solving it.
Synchronizing to AES3 Systems
To avoid passing audio through sample-rate-converters, we recommend that Livewire+ be synchronized to your
AES master clock, if you have one. Our Livewire+ AES node provides this function, recovering the clock from an
attached AES input and creating a Livewire+ sync packet. The AES node must be active clock master.
Here’s an interesting sync application for Axia AES/EBU xNodes: Two AES xNodes can be used as a way to
synchronize two AES systems located apart, but with an available IP path between them. One becomes the master,
connecting to a Livewire+ AES input. The slave attaches to a Livewire+ AES output and is configured to recover
clock from it. Ingeniously simple, no?
Synchronizing to AES67 Systems
AES67 standard requires device to use the IEEE-1588 network time protocol standard. Synchronization related
options are available on the xNodes’ QoS web page, accessible with your PC’s browser.
♦♦ Clock Mode selects the protocol required by the standard. xNodes offer a few options depending on other
system requirements:
¸¸ PTP/IEEE 1588 ARB clock class 248 mode allows the node to be a master when no better clock is
available on the network. Since xNodes are not locked to GPS or any other high precision source, only
Arbitrary (ARB) Clock class is provided.
¸¸ PTP/IEEE 1588 slave only is recommended for networks with high precision grand masters present.
78 | Section 8
♦♦ PTP Domain Number: “0” should be selected unless there is a need for multiple clock domains on a
network segment. Choosing a different domain is a useful option for doing tests without affecting other
components present on the network.
♦♦ PTP delay mechanism specifies how the xNode determines network round trip latency. This setting
MUST match the master clock setting.
¸¸ In practice, End-to-End (E2E) is always used by ordinary clocks and boundary clocks.
¸¸ Peer-to-Peer (P2P) mechanism was defined to limit amount of traffic reaching the master clock,
but is rarely used.
♦♦ PTP clock priority1/priority2 numbers provide control over the master clock selection. Choosing a
lower priority number makes the clock a preferred choice over devices with a higher number. The range is
0-255, and 128 is the default and middle value.
♦♦ PTP clock sync interval determines how often clock packets are sent. The faster those packets are sent,
the faster and more accurate synchronization can be achieved. Some IEEE-1588 devices do not support
lower clock sync interval than 0.5s, so this setting may need to be adjusted depending on what other
networking equipment is used. AES67 recommends an interval of 1/8 (8 packets per second). This
setting only affects behavior of the master clock, and does nothing when the xNode is a slave.
Other xNode AES67 settings are discussed in detail in the xNode User Manual.
Network Standards and Resources
We use standards throughout. Ethernet, as well as the AES67-2013 specification which defines the way that AoIP
devices interchange audio between manufacturers, is standardized by the IEEE and information is available on
their website at Internet Protocol and associated technologies are standardized by the Internet
Engineering Taskforce (IETF) and much can be learned from their website at Documents are a free
download. Bookshops are full of books on Ethernet, IP, and networking and we offer a list of suggested reading.
Livewire+ operates at both Ethernet and IP network layers, taking advantage of appropriate standards-based
resources at each layer.
Here are the resources we are using at the various layers:
♦♦ Layer 1: IEEE Ethernet Physical
♦♦ Layer 2: IEEE Ethernet switching, IEEE 802.1p/Q prioritization, IEEE 802.1p multicast management
♦♦ Layer 3: IETF IP (Internet Protocol)
♦♦ Layer 4: IETF RTP (Real-Time Protocol), IETF UDP (User Datagram Protocol), IETF TCP (Transport
Control Protocol), IETF IGMP (Internet Group Management Protocol)
♦♦ Layer 5: IETF NTP (Network Time Protocol), IETF DNS (Domain Name Service), IETF HTTP/WebI-
ETF ICMP Ping, IETF SAP/SDP (Session Announcement Protocol/Session Description Protocol, in the
Windows PC Livewire+ Suite application)
Network Time Protocol (NTP)
This is the Internet’s standard for conveying time. There are a number of servers on the net that users can connect
to in order to retrieve accurate time. There are also boxes from manufacturers such as EXE that receive radio time
signals and translate them to NTP packets. Livewire+ does not need NTP, but some peripherals do. For example,
Axia studio mixing surfaces and Omnia processors use NTP to automatically synchronize to the correct time.
A Note about Protocol Design
There is no question that among network protocols, the Internet has been an impressive success. One of the
reasons for this was the approach its designers took – and still use – when inventing its protocols. These are
outlined in the IETF RFC 1958 document. We’ve taken the principles to heart in the design of Livewire+. Here they
are, in priority order, and with our comments in parenthesis:
1. Make sure it works. Make prototypes early and test them in the real world before writing a 1000-page standard, finding flaws, then writing version 1.1 of the standard. (The Telos Alliance is a practical commercial
oufit, not an academic or governmental organization. We had two years of extensive lab tests of prototypes
in two locations, and then real-world field tests at radio stations before locking the core tech down. Our
prudence has now been proven out around the world in thousands of broadcast facilities.)
2. Keep it simple. When in doubt, use the simplest solution. William of Occam stated this principle (Occam’s
razor) in the 14th century. In modern terms, this means: fight feature creep. If a feature is not absolutely
essential, leave it out – especially if the same effect can be achieved by combining other features. (We
believe firmly in this principle. We tried very carefully to add nothing unnecessary.)
3. Make clear choices. If there are several ways of doing the same thing, choose one. Having multiple ways to
do something is asking for trouble. Standards often have multiple options or modes or parameters because
several powerful parties insist their way is best. Designers should resist this tendency. Just say no. (It was
just us – and we did say no. No committees or politics to cause bloating.)
4. Exploit modularity. This principle leads directly to the idea of having protocol stacks, each of whose layers is
independent of all the other ones. In this way, if circumstances require one module to be changed, the other
ones will not be affected. (We built Livewire+ on all of the available off-the-shelf lower layers.)
5. Expect heterogeneity. Different types of hardware, transmission facilities, and applications will occur on
any large network. To handle them, the network design must be simple, general, and flexible. (We had to
accommodate both dedicated hardware audio nodes and general-purpose PCs being used as audio nodes.)
6. Avoid static options and parameters. If parameters are unavoidable, it is best to have the sender and receiver
negotiate a value than defining fixed values. (These were avoidable – we don’t have any such negotiated
parameters. We do have the receiver selection of stream types, but this is simple one-ended selection.)
7. Look for a good design, not a perfect one. Often designers have a good design but it cannot handle some
weird special case. Rather than messing up the design, the designers should go with the good design and
put the burden of working around it on the people with the strange requirements. (Steve, Mike, and Greg’s
mantra! Make it work, make it solid, build enough flexibility to get the job done – and no more.)
8. Be strict when sending and tolerant when receiving. In other words, send only packets that rigorously
comply with the standards, but expect incoming packets that may not be fully conformant and try to deal
with them. (We told the s/w guys to do this. Hope they did!)
9. Think about scalability. No centralized databases are tolerable. Functions must be distributed as close to the
end-point as possible and load must be spread evenly over the possible resources. (We kept very close to
this idea – which is the main spirit of the Internet. We don’t have any central databases or other pieces along
these lines. We have a fully distributed system. If one part fails, the others keep going.)
10. Consider performance and cost. If a network has high costs and there are cheaper variants that get the job
done, why gold-plate? (Compare the power and cost of our solution with others. Using simple off-the-shelf
commodity parts was the guiding principle for our work.)
Frequently Asked Questions
We know there will be questions. Here are some we’ve already heard, and some we imagine.
General Questions
How do I know that Audio over IP will be reliable?
Axia gear with Livewire+ uses the same technology that underlies VoIP telephony. Did you know that nearly
75% of the Fortune 100 companies now use VoIP? Or that VoIP PBX systems now outsell the old kind by a wide
margin? With these systems, telephones plug into a standard Ethernet/IP network. Contrast this with traditional
PBX phone gear — proprietary devices which required you to purchase phone sets and parts exclusively from
the company that built the mainframe. You were locked into a single vendor, because the technology that ran the
mainframe was owned by the company that made the gear. (Kind of like the TDM router companies.)
IP is now accepted as a universal transport for almost any kind of signal. You see it in television studios, business
teleconferencing, government communications, banking, etc. And it’s hardly unproven, especially for applications
specific to radio studio infrastructure. As of April 2015, over 5,500 studios around the world - many in mission-critical, 24/7 broadcast applications in major markets like New York City, Chicago, Paris, Rome and Bangkok
- have been built using Axia IP-Audio infrastructure.
That’s a lot of customers. Are they happy?
Very! Axia systems are faster to install than traditional routing setups, work reliably and are easy to reconfigure.
Why not talk to the people actually using it and see what they have to say? We’ll be happy to provide you with a list
of references upon request.
Can the network be used for general data functions as well as audio?
Most certainly, should you choose to do so. The Ethernet switch naturally isolates traffic. You may even use one
link for both audio and data, since the audio is prioritized. This will probably be the case when a PC is connected to
the network – you will sometimes want to download files, receive email, etc. in addition to the audio stuff. Switch
selection is important, though, and you must use one tested and recommended by us. You could have two networks
and link them as described below.
Of course, we would never mix on-air audio and business functions or open ourselves up to hacking.
Can I make this a completely separate network?
Yes, we understand and agree. You have a few choices:
♦♦ Have a completely separate and isolated network for Livewire+. Take advantage of Ethernet, but don’t
combine any internet or business functions with studio audio.
♦♦ Have two physical networks and link them with an IP router. Correctly configured, the router provides a
security barrier.
♦♦ Share the network hardware for audio and general functions but isolate Livewire+ to its own VLAN.
Again, an IP router could be used to link the two networks.
I’ve heard that Livewire+ networks cost a lot less than traditional studio systems.
What did you leave out?
Nothing. Our cost savings compared to traditional routers are achieved by using standard, off-the-shelf switching
hardware rather than custom-built solutions. It’s a lot less expensive to use standard Ethernet for signal switching
and transport than it is to construct a customized cross-point routing switcher, with its cards, frame and peripherals. This is the same principle that has driven almost all stations to use PCs for audio playout and editing – they are a
lot cheaper and more powerful than any broadcast-industry specific machine could be.
Another way Livewire+ saves money lies in the way PC audio is handled. With a traditional router, PC audio must
be brought in through a router input card or console module; bringing multiple channels of audio into the system
in this manner (from workstations or digital delivery systems) can significantly increase the overall cost of the
Instead, we wrote an IP-Audio Driver for Windows PCs that looks just like a sound card to the OS, but streams
audio in and out of the computer’s network card instead. Or, if you need the realtime MPEG compression or time
compression features of a high-end sound card, our partner AudioScience makes an audio card with a Livewire
output that plugs directly into the Axia network. Either of these approaches eliminates the cost of the I/O needed
to get audio into the switching network. So Axia clients usually realize several thousand dollars worth of savings
over and above the cost of the sound cards themselves.
Do Livewire+ networks have any single points of failure?
Is there a central ‘brain’ I can lose that will take the system down?
Livewire+ networks are distributed, with no central box. Ethernet networks can be designed any number of ways,
including those that are fully-redundant and self-healing. Normally, our clients build larger facilities with “edge
switches” serving each studio, connected to a redundant core. Each studio is able to operate stand-alone.
I’ve heard that with AoIP, latency increases whenever you add inputs.
The more sources you add, the higher the delay. Is this true?
No, Livewire+ latency remains fixed at the same low value regardless of the channel count. You can run a system
with a thousand channels and the latency will be the same as for a single stereo stream. Indeed, the delay is so
consistent that channel-to-channel phase shift is less than 1/4 sample. The total latency of an analog input to
analog output using the Axia Livestream format is about 2.75 milliseconds:
♦♦ The time through the A/D and D/A converters is about 1.5 ms.
♦♦ The network transit time is 1.25 ms.
To put this into perspective, modern, high end audio processors typically clock in with around 10 ms. delay (and
talent regularly monitors those on-air).
82 | Section 9
Does Livewire+ route logic with audio, too?
Of course! IP is great for data, no? Logic commands from external devices enter the network using GPIO ports on
xNodes and mixing engines. The logic data is then “bound” to the audio stream, and is routed with it to whatever
console the source is loaded on.
Devices equipped with Livewire+ interfaces (like Telos Zephyrs and phone hybrids, Omnia audio processors,
25-Seven profanity delays and IDC satellite receivers, for example) supply audio and control logic directly from
the device to the Ethernet switch over a single CAT-5e connection, further simplifying in-studio wiring and making
Livewire’s audio+logic routing even more convenient.
Are there any problems with delay of control commands over the network? I’ve heard of
other systems using TCP/IP that have problems in this respect.
No, Livewire+ control latency is very small – no more than 50ms for hardware GPIO closures from Surface button
pushes. We are using a special network protocol we invented called R/UDP (Reliable UDP) rather than TCP/IP, in
part to be sure control delay is low.
What about Program Associated Data? Is Livewire+ compatible?
Yes. Devices that generate PAD plug into the Axia network; the information they supply is sent along with its
associated audio, and any devices that need it can also plug into the network and retrieve it. This means that you
can send audio and PAD together, without incurring extra costs for separate audio and data networks.
Mix-minuses are always tricky and I’ve heard that Axia consoles are good at mix-minus.
What’s so different about the way you do it and how are they generated?
As a part of Telos, you might imagine we’ve studied mix-minus for a long time. And we’ve always been amazed at
how complicated and confusing it is to set up correctly. With today’s radio shows relying heavily on phones and
remotes, something needed to be done.
Our consoles generate mix-minuses automatically, on-the-fly, without any intervention needed from talent. The
way it works is simple: when a caller is on the air, he hears the main Program feed, minus himself. When off the air,
he hears a special “off-line” phone mix that can contain audio sources: pre-fader audio from the host mic, other
phone callers, etc.
Best of all, mix-minus settings for audio sources such as phone hybrids and remote codecs are assigned to the
source itself, not the console fader — so when a source that needs a mix-minus is loaded onto any fader, on any
console, the mix-minus settings are automatically loaded too.
At the physical level, mix-minus is easy, too. Livewire+ carries audio in both directions, so one RJ-45 covers
How many mix-minuses can your consoles have?
One for every fader! There’s a lot of processing power in our mixing engines, enough that Axia consoles can
provide automatic mix-minuses simultaneously for every fader on the console. You’ll probably never need to have
40 mix-minuses running at once, but isn’t it nice to know that you could?
Can I use Livewire+ without Axia consoles?
Yes, of course. You could just use it as a snake or router system and connect whatever consoles and other equipment
you like.
How does Livewire+ compare to other audio networking systems?
The original Livewire protocol debuted in 2003 as the first audio networking system to allow real-time uncompressed digital audio to be conveyed over standard Ethernet hardware, so it’s accurate to say that all other AoIP
networking systems sprang from our work.
AES67-compliant Livewire+ is extremely low latency, which is especially important for broadcast facility operation, where live monitoring and cascaded links are common. Livewire+ includes all the technology you need for
practical studio application: Switches are controlled, sources are ID-ed and advertised to receivers, GPIO over the
network is covered, etc. Finally, Livewire+ connects directly to PCs – no soundcard or other hardware is required.
Livewire+ is a not just a technology, but rather a get-the-job-done solution. We offer you all the pieces you need to
build a modern broadcast studio. Interfaces, mixing engines, consoles, PC drivers. We are experienced broadcasters, so we know how to support radio studio applications.
As you compare Livewire+ to other systems, be sure to note the number of devices that work with those systems,
and the number of technology partners in-hand. Livewire+ is a mature system, with dozens of products and partners that contribute to a robust, connected studio equipment ecosystem. Remember, the more devices connected
via Ethernet, the less infrastructure cost and the easier systems are to deploy, configure and manage.
What is the best audio format to use with Livewire+ systems?
Our networks don’t care what format your audio is in. During the playout process, your playout software may
uncompress any compressed-format files (MP3, MP2, apt-x, etc.) and present them to the Axia IP-Audio Driver as
Similarly, compressed audio, such as Dolby Digital or Dolby E, can be carried across a Livewire+ network without
being decoded, just like it is handled in an AES or SDI stream.
What this means is that all audio that moves within the Livewire+ system can be the same — 48 kHz, 24-bit
studio-grade audio as either linear audio or compressed audio data, and the switches we recommend have enough
bandwidth to carry 10,000+ channels of uncompressed, real-time stereo audio simultaneously.
Is it possible to run out of bandwidth?
No. The backplane of a modern Ethernet switch can handle full duplex traffic on all ports simultaneously without
any packet loss. And since our component links are designed so that they never exceed any port’s capacity, we never
exceed the switch capacity. The way we prevent port overload is simple: we “own” each port. Every xNode audio
interface is plugged into an unshared 100Base-T port on the switch. Even when all of an xNode’s inputs and outputs
are active, we are still well under the bandwidth of the ports, and the switch is completely under control. Because
the switch has the backplane capacity to handle all ports fully loaded, the system performance doesn’t change from
one to thousands of audio channels.
Let’s explore the issue of switch capacity a little further. We know how much capacity is required per port for each
node, and we know that a node will never produce or consume more than 16 stereo streams total. But what about
the studio mixing engine? To support a large console with a lot of buses, inputs, mix-minus outputs, etc., you may
have 40 or 50 simultaneous signals (or more). Because this could exceed the port capacity of a 100Base-T port,
the mix engine is connected via Gigabit Ethernet only. Using Gigabit for the engine, we could support a 200 fader
console with 200 outputs and still have room to spare!
84 | Section 9
Are you sure this is robust enough for 24/7 operation? My Windows networks always
have downtime.
Livewire+ equipment is based on tight, embedded hardware and software. The Ethernet switches we recommend
are fully professional devices with high reliability and options for redundancy.
Can I connect two studios across town?
Yes, but not the way you’re probably thinking. Remember that Livewire+ audio is uncompressed 24-bit, 48kHz, so
each stereo stream is 2 Mbps. To get this done over a reasonable phone line, you’d use use the Telos Zephyr iPort
PLUS MPEG Gateway. This unit can be used to transport multiple channels of stereo audio across any network
with guaranteed QoS, such as T1 and T3 connections, or MPLS networks. It contains eight bi-directional stereo
MPEG-AAC codecs in one box and connects to your Livewire+ network using a single CAT-6 cable for all I/O.
GPIO and metadata “ride along” with the audio as well, so Livewire+ audio streams transmitted across town via a
Zephyr iPort link look like “local” audio sources to the receiving end, and can be assigned to console faders and used
just as if they were originating across the hall, with no visible difference in operational behavior.
If my favorite delivery system isn’t on your list of partners, can I still use it with a
Livewire+ network?
Yes, the same way you do it now: just plug the delivery system’s outputs into our inputs, and send the contact
closures into our GPIO ports. (Then ask your delivery system provider when they’re going to become an Axia
Our facility requires mono audio, not stereo. Are Livewire+ systems capable of
mono streams?
Yes. With Livewire+, all audio is data, so Livewire+ can route mono, stereo and even 5.1 Surround streams, too.
Our xNode audio interfaces can be configured for true mono, stereo, or surround I/O. This is built-in to all of our
Analog and AES/EBU xNodes.
Is a Livewire+ network more difficult to install than traditional routing systems?
In fact, Livewire+ is much easier to install and commission, because everything connects using off-the-shelf
Ethernet cables, which carry multiple uncompressed channels of stereo audio. 100Base-T links can carry 25 stereo
or 50 mono audio channels simultaneously; Gigabit links can handle 250 stereo or 500 mono channels each. This
labor savings from this link aggregation and resulting elimination of expensive multi-pair cable for studio interconnects can be significant.
Even our audio connectors are designed to promote fast, inexpensive installation. Livewire+ devices use the Radio
Systems StudioHub+ RJ-45 standard for I/O jacks; a huge variety of adapters are available for all kinds of devices.
Tally up the savings in labor realized from not having to purchase and hand-solder hundreds of XLR and RCA
connectors, and reduction in complexity becomes even more impressive.
In fact, due to the reduction of cabling and the quick connection of devices, clients tell us that installation goes 30%
to 50% quicker than wiring studios the traditional way.
Livewire+, Standards and Other Vendors
I hear you’ve renamed Livewire to Livewire+™. What does this mean?
You probably know that the Telos Alliance has been involved in designing the architecture of AoIP interoperability
since the very beginning (when some folks claimed it wouldn’t even work!).
First, we promoted interoperability using the standard IP networking technologies built into Livewire, beginning
in 2003. Then we became sustaining members of the X.192 project committee, which wrote the AES67 standard
ratified by the AESSC in September, 2013. By December 2013, we had already implemented AES67 in our Axia
xNode audio interfaces.
But we didn’t just stop there. We’ve continued to fully integrate AES67 into Livewire+, in order to give Axia clients
the best possible experience. Not only that: we’ve structured Livewire+ so that it can be extended to comply with
future standards.
This represents a huge leap forward — so much so that Livewire technology has evolved to become Livewire+.
I heard that Livewire+ is actually the basis for AES67. Is that true?
As part of the X.192 working group, Axia and the Telos Alliance worked with other companies to help define the
standard. During this process, we made available, at no charge, parts of our own patented technology to help speed
development of the standard.
Does that mean that Livewire+ actually is AES67?
Not exactly! What it does mean is that much of Livewire+ is the basis for AES67. You might say that AES67 has
Livewire+ in its DNA.
So what does the “plus” in Livewire+ mean, exactly?
It means three things.
1. Livewire+ is completely AES67-compliant. Compliant means that Livewire+ fully complies with all parts of
the AES67 standard.
2. Livewire+ is future-proof, because it’s extensible. Future standards (such as might come from the AES
X.210 group) can be included once ratified; Livewire+ can never become obsolete.
3. Choosing Livewire+ means you don’t have to wait for interoperability. Livewire+ represents the world’s
largest grouping of AoIP technology partners – over 80 software and hardware manufacturers and integrators whose products work together, with integrated audio, source discovery, logic control and data — right
now, without the need to wait for additional standards.
You say Livewire+ is “AES67-compliant”. Other companies say their gear is “AES-67 compatible.”
What’s the difference?
The difference is subtle, but important — compliant is not the same as compatible! “Compliant” means that
Livewire+ fully complies with all parts of the AES67 standard. It’s baked-in, if you will.
“Compatible”, on the other hand, means that someone’s proprietary tech can co-exist with the AES67 standard. In
essence, compatibility is like two different people sharing one body: the proprietary tech is patched with add-on
code to get by, but doesn’t fully comply with all aspects of the standard.
86 | Section 9
What does the AES67 standard contain?
According to Mark Yonge, AES Standards Manager:
“This standard defines an interoperability mode for transport of high-performance audio over networks based on the
Internet Protocol. For the purposes of the standard, high-performance audio refers to audio with full bandwidth and low
noise. These requirements imply linear PCM coding with a sampling frequency of 44,1 kHz and higher and resolution of
16 bits and higher. High performance also implies a low-latency capability compatible with live sound applications. The
standard considers latency performance of 10 milliseconds or less. This standard provides comprehensive interoperability
recommendations in the areas of synchronization, media clock identification, network transport, encoding and streaming,
session description and connection management.”
If AES67 does all this, why does Livewire+ exist?
Good question! The AES67 specification is a good start toward interoperability, but is actually a subset of the
many functions that Livewire+ performs today. When we developed Livewire back in the early 2000s, we had to
synthesize the critical links between networking technologies, because a full standard didn’t exist yet — and earlier
networked audio systems lacked critical functionality. We developed a way to make GPIO logic “ride along” with
the audio streams, and a way for available sources to “advertise” their availability to all the networked devices that
operators might want to use.
We also recruited partner companies whose products are respected and widely-used in the radio industry. Then
we shared our technology with them, so that station engineers could connect as many audio devices as possible
directly to their audio network. Not only is native connectivity an elegant solution, it reduces total system costs by
removing the need for those extra I/O devices.
Ten years later, the industry finally adopted a standard for AoIP audio transport in AES67. The goal is for every
studio audio device to eventually click together with CAT-5 and share audio. But along with that shared audio,
there’s a whole world of other functionality that broadcasters expect — like device start/stop functions, monitor
mutes, on-air tallies, the ability to control peripherals from the console, the ability to know when an audio source is
live and ready for air, the ability for playout systems to control fader on/off functions and more. Those are functions
that AES67 alone doesn’t provide for, but Livewire+ does. And now that the first AES67 devices are appearing in
the marketplace, broadcasters have quickly found that they also need to support those additional capabilities in
order to provide an integrated control experience for the user — otherwise they’re no better than AES3 streams,
with serial GPI cables running alongside.
With Livewire+, you can have your cake and eat it, too. Over 5,500 Livewire+ powered consoles and 60,000 devices
are in use around the world daily. Livewire+ provides audio with integrated control and discovery right now. And,
because Livewire+ is extensible, broadcasters using it can enjoy its rich control architecture and integration with
over 80 partner products, exchange audio with AES67 devices, and be assured that when an AES standard for
control is ratified, it will be part of Livewire+ as well. Livewire, the original AoIP technology for broadcast is still on
the leading edge, and is the most successful by a wide margin.
Does Livewire+ still work with RAVENNA?
Yes. Axia and RAVENNA have been partners for several years, both working together to define the AES67 standard.
Going forward, standards-based Livewire+ will continue to be backward-compatible with the RAVENNA protocol.
The Telos Alliance and ALCNetworX were a big part of the X.192 project. Isn’t AES67 just a
synthesis of those two systems?
Not at all. The X.192 Task Group consisted of over 100 members from a wide variety of equipment manufacturers
— including some who are direct competitors. It wasn’t a “rubber-stamp” project! AES67 is the result of 3 years of
collaboration by these many contributors, ensuring use of the best ideas, no matter where they originated.
You say that Axia introduced AoIP to broadcast in 2002, and you’ve got 5,500 studios on-air.
So why is AES67 even needed?
When Steve Church, Greg Shay and their team were developing Livewire, they had to invent tech that didn’t exist
before. One critical piece of tech was network clock sync. Problem was, the Ethernet standard in place at the time
had no criteria for high-precision time-synched audio.
Why is this so critical? As Telos Alliance Chief Science Officer Greg Shay explained in his excellent paper “Taking
the ‘Sting’ Out of Evolving Digital Audio Networks” (presented at NAB 2013 and downloadable from “Accurate timebase recovery is directly related to, and essential for, low latency (low
delay) of the audio going over the network.”
In other words, if you want networked, real-time, broadcast quality audio without jitter and delay, you’ve got to
have all network devices synchronized to a network master clock. So we invented the first distributed high-precision clocking system for Ethernet, and debuted it in Livewire.
Although Axia has always been happy to share our tech with software and hardware manufacturers, other, open
methods of Ethernet clock sync emerged, one of which became the IEEE-1588 synchronization standard — which
is an integral part of AES67.
I need new studios. Should I wait to purchase equipment until all the manufacturers
adopt AES67?
Not unless you’re prepared to wait quite a while. Implementing a new standard always takes time, as manufacturers
adapt existing products, design new ones, and release software updates.
In the meantime, Livewire+, with over 5,500 studios on-the-air worldwide, is working today, and has a roadmap
for future compatibility of existing hardware. This assurance means that the studio you build with Livewire+ today
will be compatible with future AES standards for AoIP — your studio gear can’t become obsolete.
What changes, if any, will be required for my existing Livewire network to support AES67?
Compliance with AES67 is a part of the latest software present in Axia analog, AES/EBU, microphone and
mixed-signal xNode AoIP interfaces, and Linear Acoustic SDI xNode devices. If you have these products already,
simply download the update packages and apply them using a standard Web browser and PC, or use Axia iProbe
software to “push” the update to your devices en masse. After this update, you’ll be able to generate and consume
native AES67 streams.
And of course, the Telos Alliance plans to implement AES67 capability within more products going forward — your
assurance that studios built with Livewire+ are future-proof.
Will I need to change my Ethernet switches to work with Livewire+ and AES67?
No! Livewire+ and AES67 coexist on the same switch fabric.
88 | Section 9
What are the advantages of specifying Partner equipment that works with Livewire+,
versus gear that supports AES67 alone?
The whole idea of AES67 is to open up more equipment choices when you buy. The AES67 standard enables equipment from a variety of manufacturers to exchange “agnostic” audio streams without fuss. However, the AES67
specification provides interoperability standards for audio only — not for control or data exchange. Nor does
AES67 equipment have the ability to automatically discover available audio, and present it to the operator for use.
Most engineers agree that these abilities are a big part of the AoIP advantage. So, if control and source advertising is
important to you, choosing equipment with Livewire+ capability will definitely be to your advantage.
Are you planning to share information so that other vendors can make gear that directly
plugs to Livewire+?
Yes, and in fact we have done this for several years. Software vendors for PCs can use our driver to easily make their
applications compatible; many delivery system providers already have built in Livewire+ compatibility. Through
our Livewire Limitless License program, we provide technical data and schematics that allow hardware vendors to
build Livewire+ interfaces into their products as well; this is why Livewire+ networks have the largest ecosystem of
devices that connect directly — nearing 100 partners as of this writing.
Of course, you can use whatever equipment you want via the analog and AES xNodes.
PCs and Livewire+
Tell me about your “sound card” driver for workstations.
The official name is “Axia IP-Audio Driver for Windows”. It makes the Livewire+ network look like a sound card to
a PC Windows application. Most audio applications should work unmodified.
We’re a Linux facility. Will the Axia Driver work with Linux?
Yes, we’ve partnered with Paravel Systems, developers of the Rivendell Radio Automation System for Linux. They
can compile Axia IP-Audio Drivers to work with your Linux PCs. To find out what distros are currently supported,
contact them directly at .
Building Livewire+ Facilities
I’ve got a large facility. How many studios can I interconnect?
There is no practical limit. Livewire+ networks can handle as many as 10,000 audio streams at a time; you may have
as many studios and audio channels as your Ethernet switch fabric can support. Switches come in all sizes, some
with hundreds of ports. And multiple switches may be cascaded to expand ports. For large networks, we recommend that you use a switch per studio to isolate any problems to a defined area. These are then interconnected
with a backbone. Switches may be physically associated witch each studio or may be all in a central location, as you
What about for smaller stations? This all sounds pretty sophisticated for a simple set-up.
Look at Ethernet for data applications. You have everything from a single PC connected to a printer to a few PCs in
a small office tied to the Internet and a couple of printers to huge campus networks with thousands of nodes. This is
one of the reasons we went with Ethernet – you can use it for big and small facilities. The technology and economics
naturally scale to suit the application size. We figure, in fact, that small stations may benefit the most as they gain
routing capability at a very modest cost.
This seems like a lot of IP to keep track of. What administration tools does Livewire+ have?
All Livewire+ devices have a web browser control and monitoring capability. Keep the IP numbers in a “favorites
list” and you can easily check them. Or make your own web page with all the links.
An off-the-shelf tool is the Axia iProbe Network Management Console. iProbe is an intelligent network maintenance and diagnostics tool that makes managing, updating, and remote control of your Livewire+ system very easy
and intuitive.
What Ethernet switches will I need for my Livewire+ network?
You won’t need any bleeding-edge products from the skunkworks, if that’s what you mean. In fact, you may not
need a third-party switch at all! Every Axia console has a custom-built, zero-configuration Ethernet switch built
in – perfect for standalone studio deployment, but these switches may also be daisy-chained to create larger studio
networks. If you just need a few local switch ports on the edge of your network, or if you’re building a smaller
studio, our standalone xSwitch might be what you’re looking for – it’s pre-configured to install in under a minute.
Of course, facilities with multiple rooms and large routing networks require core switches with larger capacity. For
these systems, we’ve qualified a number of switches from Cisco, the industry leader in switchware. You can see the
selection at .
You built your own standalone network switch? Why?
Yes, it’s called xSwitch, and it’s the first of its kind: an Ethernet switch custom-built for the needs of broadcasters
using Livewire+ IP-Audio networks. Here’s why we developed it.
Up until now, configuring the network switch has been the most tiresome part of setting up an IP-Audio network.
To eliminate this, we built an Ethernet switch (using industry-standard chipsets found in high-performance,
brand-name switches) and pre-configured it for use with Livewire systems. So, no configuration, no programming,
and virtually no setup. Just assign an IP address, plug it into your network, and it’s ready to go.
Do switches need periodic replacement as new features are introduced?
No. Ethernet is a standard, IEEE 802.3. Livewire+ gear works with any switch or router that supports the standard.
You might want to upgrade to a larger or more powerful switch for some reason in the future; for example, if you
were to add more studios to a cluster. Or maybe you would like to change from copper to fiber for some kind of
remote uplink connection. Or you might want to replace an older switch at some point. Thanks to IEEE 802.3, the
replacement switch will simply plug into the network and function, but there is certainly no requirement to do this
to support new product features.
Why can’t I just use my favorite switch with Livewire+?
People ask us this question every so often. When clients require a third-party switch, we recommend Cisco because
their reliability, feature sets and performance are the best we’ve found. They also offer a wide range of switches at
all price points to meet individual users’ needs.
How come other manufacturers’ switches don’t measure up? This is mostly due to individual manufacturers’
differing implementation of the same “standards”. For file transfers and e-mail, these differences are immaterial. But
for VoIP and, most especially, AoIP, these implementations become more important. It’s quite possible for a given
switch to “work” with just a few xNodes attached, but when a more robust test is applied, that same switch can fail.
90 | Section 9
For example, we found a nice, inexpensive switch from a well-known manufacturer which, on paper, met all specs
and worked with small systems in the lab. However, it turned out to not actually meet its own published specs when
deployed for rigorous service in a large Livewire+ system.
For this reason, we recommend and qualify only selected Cisco switches for clients whose network designs require
a large switch fabric. And for more moderately-sized networks, there are Telos Alliance-designed switches built
into our PowerStation, QOR and xSwitch products — which employ some of the very same high-performance
chipsets found in Cisco switches.
Do I have to be an IT expert to run a Livewire+ system?
We’re engineers, and we like to talk tech. Sometimes, we talk about the tech more than we need to! But in this
respect, an IP-Audio network is like a car: you don’t have to understand how the engine works in order to drive it.
Just connect two pieces of gear together with CAT-5e and they will talk to each other — like plugging a mic into
a mixer. We’ve even created an Ethernet switch just for Livewire+ – xSwitch – that requires zero programming.
Plug it in, assign an IP address, connect audio devices and start passing audio, with no other setup required. The
Livewire+ protocol takes care of routing the audio without any need for intervention from you. And the equipment
interface is all web-based with GUI control. It works intuitively, and you don’t have to know anything about the
tech inside to make it work.
That being said, another of the advantages of Ethernet and IP is that bookshop shelves are full of well-written books
that can explain any aspect of standards-based networking at any level of detail you might want.
What levels of redundancy are available in Livewire+ systems?
We let you choose the amount of power and network redundancy your plant demands. If you like you can design a
system with power and network connection redundancy from the core, all the way to the edge – you decide what’s
best for you.
For instance, our StudioEngine mixing engine includes standard redundant, auto-switching power supplies that
are field-replaceable in seconds. PowerStation and QOR.32 mixing engines feature auto-switching backup power
as an option. xNode audio interfaces can be connected to both AC and PoE-enabled network switches, for automatic failover. xNodes and xSwitch components feature twin network interfaces, to protect the integrity of network
links. And we’re glad to help you optimize your network switch fabric for maximum redundancy, too; just ask us
and our Support experts will happily assist in your network planning process.
How do analog sources become part of the network?
With Livewire+ xNodes. These come in variants for line and microphone applications.
What about AES?
There is also an xNode that interfaces your AES audio to the network. This is a direct bit-to-bit procedure with no
conversion of any kind. You can also sync a Livewire+ system to an AES master clock.
Is there a way to import audio from SDI links as well?
We have an xNnode that interfaces the audio from SDI streams, both HD- and SD-, to the network. Livewire+
systems can be locked to the reference clock of an SDI input stream, which allows for handling coded audio without
destroying the audio. Video compensating delay is included to ensure that the video and audio are synchronized at
the outputs.
Do I have to use CAT-6 to connect everything? That could get pricey.
CAT-6 is used only for heavy-traffic network segments, like connecting one studio to another, or connecting
switches to each other. All other equipment is connected with common, inexpensive CAT-5e cable.
You said I can get RS-232 data through the system. How is that done?
Using 3rd party devices, such as from Lantronics, your serial data can go anywhere across the network and be used
where it’s needed.
Most companies recommend that I bring them on-site to help install and configure their
systems. Do I need your help to install a Livewire+ system?
With those other guys, you’d better hire their systems engineers. With us, it’s much easier! If you know how to use
a Web browser and plug a telephone into the wall, you’ve got all the skills needed to install and configure your new
network. And Telos Alliance 24/7 Technical Support is there to help if you need it, too, around the clock, every
single day of the year.
If you still decide you’d like on-site installation services, we have those available as well and will be happy to talk
with you about it.
I’ve been hearing about IPv6 a lot. Will this affect how Livewire+ networks operate?
Internet Protocol version 4 (IPv4) is the IP addressing structure that powers the Internet. The Internet Engineering Task Force (IETF) has designated IPv6 as the successor to version 4, due to its larger address space, which
offers more flexibility in routing traffic and allocating addresses, providing hundreds of billions more possible IP
addresses than IPv4.
There are no compatibility issues between products using IPv4 or IPv6 addressing. Both addressing schemes can
co-exist on the same network and interoperate smoothly. Axia systems employ Multicast streaming for audio
routing. This is fully developed within IPv4, but it is not widely used under IPv6. For this reason, Livewire+ devices
will continue to use IPv4 until such time as IPv6 will provide the same consistent performance. We monitor the
technology standards very closely and we plan to move to IPv6 when Multicasting is widely implemented.
There is one case in which IPv4 could be a limitation: If you plan to have several hundred billion devices on your
network. If you have plans to build such a large network, call us. We’ll gladly implement IPv6 for you right away!
Ethernet Media
Are optical audio links supported?
Livewire+ is fully compatible with copper and fiber connection types. A common configuration is for switches
dedicated to studios to be connected with copper connecting nodes, engines, surfaces, etc. A fiber backbone can
then connect the switches in order to share audio among the studios.
What Ethernet rates do you support?
Switch-to-switch links may be any supported Ethernet media, Gigabit 1000BASE-T recommended. Our processing
engines use 1000BASE-T. PCs may use Gigabit 1000Mbps or 100Mbps, copper or fiber. xNodes and other devices
connect with copper 100BASE-T links. Media converters allow the use of fiber on nodes, such as for extended-range snake applications.
92 | Section 9
What happens if someone accidentally unplugs a cable?
Livewire+ nodes “advertise” the presence of their audio streams to the entire Livewire+ network. So if someone
unplugs a node, the sources attached to it will be offline. But all you have to do is plug that node back in, and the
node will “advertise” that the audio streams are available again. Within 10 seconds, all destinations that need those
sources will be back up and running.
Additionally, among the many features of our PathfinderPC routing control software is a silence detect function
that can be programmed to switch to another feed should one stop working for any reason.
The Internet and Livewire+
What about hooking up over the internet? With my studio audio in IP form,
can I just plug a port from the switch into an Internet router?
As Internet bandwidth becomes ever more plentiful, arguments for using it for audio transmission become more
convincing. A gateway device, like the Telos Zephyr iPort PLUS, can perform compression from Livewire+ PCM to
a lower-rate bitstream using an MPEG AAC codec.
The main problem to be overcome is the Internet’s lack of any Quality of Service guarantees; a “net storm” that
starves bandwidth and drops audio might not be a big deal to a kid at home with his computer, but it sure wouldn’t
be good for an important on-air feed. Private networks with reserved capacity are one answer. Another could be the
“resource reservation” and “differentiated services” technologies that has passed out of the laboratory and might
eventually be implemented by Internet Service Providers – at a cost, of course.
For now, “nailed up” IP links which support QoS offer us the most reliable audio transmission. Our Zephyr, Z/IP
ONE and Zephyr iPort PLUS codecs all support direct Livewire+ connection, so you can use them to get a remote
link into your Livewire+ network that way and effectively have the same result.
Can a Livewire+ network can catch viruses?
There are no general purpose operating systems in Livewire+ devices, so the answer is “No.” You can keep computers attached to your Livewire+ audio network safe by keeping it isolated from data networks.
Networked Consoles
How many mixing consoles can a single mixing engine handle?
Each mixing engine provides a huge amount of power and flexibility to each control surface, so it’s a one-to-one
The exception is our DESQ and RAQ utility mixers, which are designed for less demanding work in locations such
as VO booths, road kits, and personal studios. Our QOR mixing engines can each support two DESQ or RAQ
consoles, or one of each.
Just how powerful are your mixing engines?
The amount of processing power in modern CPUs is staggering. With optimized software design, a single Core
chipset can outperform the largest multi-DSP consoles and routers.
Additionally, we’re running a pared-down and highly-optimized version of the Linux operating system and our engine
processing application code is carefully “written to the metal”, making our mixing engines very powerful indeed.
For instance, voice dynamics are a standard feature of all Axia consoles, a feature embedded in each mixing
engine — and it was developed by Omnia Audio’s Frank Foti, who knows a little something about audio dynamics
processing. Other standard features include per-source EQ and panning, headphone EQ and a one-touch off-air recording function, all of which can be set and saved to instantly recall each jock’s favorite settings. Fusion, Element,
iQ and Radius consoles all feature four stereo program buses, and four stereo aux sends with two stereo aux returns
for off-air production, and all provide automatic mix-minus for any and all phone callers or codec inputs. That’s a
lot of processing, and Axia mixing engines are powerful enough to handle it all.
I like Axia consoles’ features and design, but I’m not ready to commit to Livewire+ for
my full facility. Can I just use one surface and a mixing engine as a drop-in console
Sure, you can. Take a Fusion console and PowerStation mixing engine, for example, plug in your audio I/O and you
have a stand-alone console that interfaces via analog or AES to your other equipment.
Analog Audio & AES On RJs And CAT-5
You recommend an outer shield for analog audio. Why?
As a precaution. Shielded cable protects against RF and eliminates any possible crosstalk between cables in
multi-cable bundles.
Is there any crosstalk between the pairs within the CAT-5 cable?
As long as your circuits are balanced, there is almost no left/right crosstalk inside the cable. With a balanced input
circuit that has 50 dB CMRR (Common Mode Rejection Ratio), separation will be greater than 90 dB.
So, must all the audio and digital signals be balanced?
Generally, yes, or crosstalk will degrade. Unbalanced connections can be used for short runs only and preferably
with separate cables for left and right if you care very much about stereo crosstalk. Radio Systems makes small
devices that adapt unbalanced RCAs to balanced RJs for their StudioHub system that could be used to convert any
unbalanced sources you have. AES3 digital audio signals are always balanced and require no conditioning.
Is CAT-5 OK for AES3 digital audio?
A 1997 report, Review of Cables for AES/EBU Digital Audio Signals, conducted by the BBC Research and Development Department, concluded that CAT-5 shielded twisted audio pair cable “offered the highest performance of all
the cables tested here.” Their tests included coaxial cables and special cables specifically designed for digital audio.
They preferred CAT-5 cables for their consistent performance and because they have the flexibility to support other
signal formats.
CAT-5 cables are engineered for data rates up to 100 Mbps to support networks such as 100BASE-T. Since AES3
signals have a bandwidth of about 3 Mb/sec (depending on sample rate), AES3’s requirements are well within
the CAT-5’s guaranteed performance parameters. Dependable error-free transmission is possible at lengths up to
920 meters (over ½ mile). CAT-5 cables perform well for AES3 because they are engineered to have characteristic
impedance of 110 ohms and extremely low capacitance – in the 12 pF/ft range. This yields low signal reflection and
excellent high frequency response, thus lowest error rates.
The information presented above refers to 100-ohm balanced AES connections used in radio plants. In TV-land,
most AES connections will be unbalanced, 75-ohm signals, and use the same type of cable as SDI audio.
94 | Section 9
Is CAT-5 OK for analog audio?
Sure, it is! The low capacitance, needed for data’s high velocity and wide bandwidths, yield exceptionally flat
analog audio frequency response, even over very long cable lengths. The tight, controlled twists are good for hum
and crosstalk rejection. Steve Lampen, a senior audio video specialist for Belden Wire & Cable writes, “Digital
cables make the absolute best analog cables. You can go farther with flatter frequency response than with any cable
designed for analog”. Steve Lampen of Belden graciously wrote us a paper on the subject entitled “The Axia Guide
to Choosing Category Cable”; you can get it from the Telos Alliance website.
Also see Belden’s web site for more interesting and revealing papers on the subject of using CAT-5 and CAT-6 cables
for analog signals.
Networking is a field well covered by books and web sites. There’s plenty of information out there. Here is a
selection of some resources we’ve found useful.
The Telos Alliance –
Contact us at: [email protected] or by phone at +1 216 241.7225
Steve Church & Skip Pizzi, Audio Over IP: Building Pro AoIP Systems with Livewire™, Focal Press, 2010
The Bible of Livewire, written by its co-creator and Telos founder Steve Church, with NAB Senior Director, New
Media Technologies Skip Pizzi. Pretty much the definitive resource for Livewire+, with chapters covering everything from basic network topology to advanced control protocols.
The standards body for Ethernet. The documents are now a free download, but will cost you a lot of paper and toner
– the basic Ethernet standard is 1,268 pages!
Charles E. Spurgeon, Ethernet: The Definitive Guide, 2nd Edition; O’Reilly & Associates, 2014
Living up to its title, it is pretty definitive on basic Ethernet topics.
General Networking and Interest
IETF (Internet Engineering Task Force) -
The Internet’s main standards organization. Look for the RFC (Requests For Comment) documents to see in detail
how the Internet is built.
Andrew Tannenbaum, Computer Networks; Pearson Education/Prentice Hall, 2003
Our favorite general networking book. Popular college textbook covers it all, including multimedia, with a breezy
style and at just the right level of detail: enough to be useful, but not so much as to be overwhelming.
J. Naughton, A Brief History of the Future; Overlook Press, 2000
Not really so interesting for audio and Ethernet, but still worth reading for perspective. This history of the Internet
tells how it happened in a friendly – even charming – way. Lots of stories and anecdotes. We particularly love
AT&T’s repeatedly making clear that digital communication had no future. (Something a lot like what we heard
from certain quarters regarding the future of computer networks for studio audio.)
96 | Section 10
Cabling Information & Standards
Cabling Installation & Maintenance -
This magazine, targeted to cabling contractors, is a good way to keep abreast of the latest TIA/EIA cabling
specs. It is also a great source for innovative cabling accessories, testers, and installation techniques.
Jim Abruzzino; Technician’s Handbook to Communications Wiring; CNC Press, Chantilly VT, 1999.
This book is concise yet contains a lot of great information including proper technique for working with CAT-5
cable and connectors. Small enough to keep with your toolbox.
Cabling Design - — Cabling tutorials
TIA - — Standards organization for cables
Global Engineering - — Sells the TIA/EIA cabling standards
Cable and Contractor Supplies
AMP - — RJ plugs and tools
Belden Cable - — Leading cable supplier
Hubbell Premise Wiring - — Devices for CAT-5, etc.
Panduit - — Marking and installation products
Corning - — Fiber optic cabling and components
Siemon - — Punch blocks
Cable Testers
Fluke - — Full range of testers
Keysight – — Top-end tester formerly made by Agilent
Triplett / ByteBrothers - — Low-end testers
JDSU – — Cable testers, Ethernet assurance and fancy sniffers, too
Ethernet Switch Vendors
Cisco Systems -
Network “Sniffers”
Wireshark -
Ethernet Radio Equipment
Here’s a list of some units we or our clients have successfully used
Cambium Networks PTP Series -
Carlson Wireless - (various products)
Dragonwave Horizon Quantum -
Exalt Communications - (EXr Series, and others)
Harris - – RF7800w
Routerboard - (Formerly Mikrotik, SXT Series, for 6 channels or less)
RAD - (Airmux-400)
SAF Tehnika - (various products)
Ubiquiti - (airMAX NanoBridge M series)
Appendix A: Livewire+ Tech Details
You don’t need to read any of this unless you want to know about the internal details.
Livewire+ Packet Structures
The speed of the link, the bit requirements of the header and payload, and the number of samples that are combined into a packet determine link capacity. The facts are the more samples that are combined, the less the header
overhead per packet, and the higher the efficiency and capacity.
There is a fundamental tradeoff: When we have more samples per packet, we have more capacity – but at the
expense of more delay. Good design means finding the best compromise.
The sampling rate and the number of samples that are combined into a packet determine delay:
♦♦ Packet-time = 1/sampling-rate * samples-per-packet.
There is one packet send buffering; three packets receive buffering, and the switch latency, therefore:
♦♦ Link-delay = packet-time *4 + switch latency
Standard Streams
Standard Streams are compatible with Internet standards. They use large packets so as to be very efficient with both
computer resources and network bandwidth.
Standard Stereo Stream Packet Format
Interpacket Delay
This is not actually transmitted, but must be taken into account for network
bandwidth calculations
Ethernet Header
Includes the VLAN/priority fields
IP Header
UDP Header
RTP Header
240 samples @ 48kHz, 24-bit, stereo
Audio (variant)
120 samples @ 48kHz, 24-bit, stereo
Total bytes per packet = 1440. Core delay = 5ms. (720 and 2.5ms with the variant format)
An Ethernet frame’s maximum data length is 1500 bytes, so you can see that we have chosen to pack the Ethernet
frame to nearly the maximum possible. There are two reasons for this: 1) the frame rate is lowest possible to put
the least burden on PC receivers, 2) the header overhead is applied to the most data so the proportion of capacity
devoted to audio vs. overhead is highest.
Livestreams are specialized for low delay, so we can pack only a few audio samples into each packet. Because they
are smaller, less buffering is needed and that means the time delay is lower.
Livestream IP Packet Format
Interpacket Delay
This is not actually transmitted, but must be taken into account for
network bandwidth calculations
Ethernet Header
Includes the VLAN/priority fields
IP Header
UDP Header
RTP Header
12 samples @ 48kHz, 24-bit, stereo
Total bytes per packet = 72. Core delay = .25ms.
The header load for RTP/UDP/IP is 40 bytes per packet, which is a significant piece of the network bandwidth
given that our audio is only 72 bytes. Most of the time this is of no consequence, since we have plenty of bandwidth.
However, there are some situations where a “lean and mean” approach makes sense.
Surround Streams
By know, you realize that Livewire+ inherently carries multiple audio streams and surround mixing is a built in
feature in the StudioEngine mixing engine for Fusion and Element consoles. This makes Livewire+ and Axia fully
compatible with surround formats for television audio. It would also make for a very nice surface to control your
personal surround sound system in your living room.
Surround Stream IP Packet Format
Interpacket Delay
This is not actually transmitted, but must be taken into account for
network bandwidth calculations
Ethernet Header
Includes the VLAN/priority fields
IP Header
UDP Header
RTP Header
60 samples @ 48kHz, 24-bit, stereo
Total bytes per packet = 1440. Core delay = 1.25ms.
Surround streams are actually multiple stereo streams carrying the 5.1-channel audio plus a 2-channel downmix
simultaneously. Surround streams carry 8 speaker channels in the following order: Left front, Right front, Center,
LFE (low frequency enhancement), Left Surround, Right Surround, stereo left, and stereo right.
100 | Appendix A
Network Link Capacity
Each Standard Stereo Stream has a bitrate of 2.304Mbps. A 100Mbps link can therefore carry 43 such channels at
full capacity and a 1000Mbps link can carry 430 channels.
Each Livestream has a bitrate of 3.776Mbps. A 100Mbps link can therefore carry 26 such channels at full capacity
and a 1000Mbps link can carry 260 channels.
In practice, links to hardware nodes will have a mix of Standard Stereo Streams, Livestreams, and control data. Our
biggest xNnode has 4 stereo (8 mono) channels in and 4 stereo (8 mono) channels out, so there is plenty of link
capacity. PCs use the more efficient Standard Stereo Streams, so again there is plenty of capacity to handle both
audio and simultaneous file transfers, etc. Our Studio Mix Engines connect with 1000Mbps links, so the sky is the
limit there.
All of the above has been concerned with per-link bandwidth. The system bandwidth is effectively unlimited with
appropriate switches.
Multicast Address Translation
Livewire+ streams are multicast at both layer 2 and layer 3. The Livewire+ channel number is automatically
translated to the appropriate addresses at both layers internally. You might want to know the translation algorithm
because maybe you or your network engineer might need to check packets with a “sniffer” or Ethernet switch
diagnostics. So here are the details.
Livewire+ channels range from 0 to 32767. Audio streams are mapped into IP and Ethernet multicast addresses
using the channel numbers for the lower 15 bits as follows:
IP Address
Livestream and Standard Stereo Streams
4 addresses are our system defaults, the others not used (left for expansion)
Back Standard and Livestream Stereo Streams
Not used (left for expansion)
Not used (Livewire v1.0 used for Ethernet Livestreams)
Not used (left for expansion)
Not used (left for expansion)
Surround Streams
Not used (left for expansion)
Not used (left for expansion)
The following special addresses are assigned:
IP Address
Livestream Clock
Standard Stream Clock
Advertisement Channel
GPIO (UDP port 2060)
These all are within the range specified for “Organization-Local Scope” use by IANA – the Internet Assigned Names
and numbers Authority. Routers do not propagate traffic on these addresses to the Internet; they stay contained
within LANs. (We also set the “link local” bit and TTL=1 in the IP header to further ensure that streams stay local.)
The range supports our 32k channels, with up to 120 stream types per channel. We are only using four types now,
but there is plenty of room for growth.
Our motivation for mapping each type to a contiguous block rather than having the type in the lower-order bits
is to allow configuration of switches and routers on a per-type basis by specifying an address range. This direct
mapping of channels to addresses also makes sniffing easier: it is simple to know where to look for a particular
audio stream.
IP addresses are mapped into an Ethernet MAC layer multicast, according to a de-facto standard process for this
procedure. This process is as follows:
♦♦ Using the Class D address, identify the low order 23 bits of the class D address.
♦♦ Map those 23 bits into the low order 23 bits of a MAC address with the fixed high order 25 bits of the IEEE
multicast addressing space prefixed by 01-00-5E.
♦♦ Assume: Channel = 80
♦♦ Assume: stream type is Standard Stereo Stream
♦♦ Then: IP address = (dotted decimal)
♦♦ And then: Ethernet MAC Address = 01-00-5e-00-00-50 (dashed hex)
Ethernet addresses are transmitted most-significant byte first, but least-significant bit first within the byte, so in
our example it is the 1 in the leftmost MAC address byte 01 that signifies a multicast address.
The Telos Alliance • 1241 Superior Ave. • Cleveland, Ohio, 44114, USA • + •
© 2015 TLS Corp. The Telos Alliance.® All Rights Reserved. TA/15 19013
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