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TELOS VX

THE FUTURE IS CALLING.

MEET TELOS VX: THE FUTURE OF BROADCAST

TALKSHOW SYSTEMS.
VX is the worlds first VoIP talkshow system. Its incredibly
powerful, incredibly flexible, and highly scalable -- a powerful
whole-plant broadcast phone system thats also economical

er 3 coding in the revolutionary Zephyr ISDN codec. We produced

the first hardware MP3 streaming encoder for broadcast. We
developed the worlds first whole-plant broadcast phone system. And we invented the IP-networked radio console, and then
integrated broadcast phones into that network via Ethernet.

enough for stations with just two or three studios. VX con-

Telos has grown steadily since our initial production run of 25

nects to traditional POTS and ISDN telephone lines via stan-

Telos 10 units in 1985! With tens of thousands of systems in the

dard Telco gateways. But it can also connect to VoIP-based PBX

field, it now is hard to find a broadcast facility in the world with-

systems and modern SIP Trunking services to take advantage

out at least one piece of our gear. Our organization, now called

of low-cost Internet-delivered phone services. Using standard

Ethernet as its data backbone, VX significantly eases the cost
of phone system installation, maintenance and cabling, while
making it easier than ever for talent to take control of their
phone system. VX is truly the future of broadcast phones.

The Telos Alliance, includes the Omnia Audio, Axia Audio and Linear Acoustic brands, and our R&D department the largest research team in broadcasting continues to develop innovative
audio products for radio and television broadcasting, telephony,
and the Internet.

ABOUT TELOS

ABOUT THE VX BROADCAST VoIP SYSTEM

Steve Church founded Telos Systems in 1985. As both a talk

With the VX, weve wedded the capability of modern networking

show host and radio group Technical Director, Steve was only
too familiar with the frustrations of bad phones and even less
responsive equipment manufacturers, so he set about eliminating the technical problems that plagued radio call-in segments.
In 1984, he invented the Telos 10, the first DSP-based telephoneto-broadcast interface system allowing radio stations to significantly improve the technical quality of call-in segments. The
overwhelming response to Steves economical and technically
elegant solution to a nagging problem provided the spark from
which Telos was born.

to the remarkable power of todays digital processing to bring

the benefits of the resulting synergy to broadcast facilities. With
VX, you can move and share lines between studios at the touch
of a button. VX is naturally scalable, capable of serving even the
largest of facilities while remaining surprisingly cost-effective
for even single stations with more modest needs. To make the
most of this networked environment, weve built VX around the
VoIP (Voice over IP) standard.
WHY VOIP FOR BROADCAST?
VoIP has taken the business world by storm, increasing the flex-

A lots happened since then. We pioneered the use of MPEG Lay-

ibility of office phone systems and PBXs while simultaneously

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VX: WHERE PHONES AND

IP NETWORKING INTERSECT.
lowering maintenance and equipment costs. In fact, most Fortune 500 companies have replaced their older PBX systems with
VoIP for just these reasons.

attention from the board.

The VX systems standards-based VoIP architecture helps you
save money, too, by widening your choices in telco providers.

VoIP is a natural for broadcasters, interconnecting the phone

You can connect to traditional POTS or ISDN phone lines using

system CPU with audio interfaces, phone sets, console control-

standard telco gateways or connect to pure VoIP services us-

lers, and PCs running screening software by way of efficient,

ing modern SIP Trunking, which can deliver substantial savings

low-cost Ethernet. Using VoIP, you can finally share phone lines

to stations that need a large number of lines.

among multiple studios and route caller audio anywhere in your

facility, easily and instantly. Got a hot talkshow that suddenly
needs more lines in a certain studio? Just a few keystrokes at a
computer and youre ready no delays, and no cables to pull.
VX can even connect with your business offices VoIP PBX to allow easy call transfers.

But its not just VoIP Its VoIP from Telos. Every incoming line
has its own fifth-generation Telos Adaptive Digital Hybrid, our
most advanced ever packed full of technology engineered to
extract the cleanest, clearest caller audio from just about any
phone line, even notoriously noisy cellular calls. Multiple lines
can be conferenced with superior clarity and fidelity. Smart AGC

REDUCED COST. INCREASED FLEXIBILITY.

ensures consistent caller audio levels. New Acoustic Echo Cancel-

This sophisticated networking allows rich communication be-

lation from FhG removes feedback and echo in open-speaker stu-

tween devices. For example, caller information entered by a pro-

dio situations. And should you choose to use SIP Trunking telco

ducer is displayed on the studio phones color LCD. Caller audio

services, calls from mobile handsets with SIP clients will benefit

is available on studio PCs for easy recording. Operators at mixing

from VXs native support of the G.722 HD Voice codec, instantly

consoles can directly control line switching without taking their

improving caller speech quality.

VX SYSTEM COMP0NENTS

VX ENGINE
The VX Engine is the heart of the system. A 2RU rack-mount de-

what kind of line or phone the caller uses. Caller audio benefits

vice with enormous processing power, the VX Engine provides

from Smart AGC coupled with famous Telos three-band adap-

all the call control and audio processing needed for the system.

tive Digital Dynamic EQ and a three-band adaptive spectral pro-

You can have as many as 48 lines active simultaneously on-

cessor. Send audio gets its own sweetening with a frequency

hold, being screened, queued for air, etc. and up to 16 audio

shifter, AGC/limiter and FhGs Advanced Echo Cancellation tech-

connections on-the-air at once. (Think of an audio connection

as a hybrid, with each bidirectional console connection containing the audio from phone system to console and the clean feed
from console back to phone system.). Its two Gigabit Ethernet
ports provide a cost-effective interface to both telephone lines
and studio audio via proven Livewire AoIP. VX is Web-based, so

nology that literally eliminates open-mic feedback. Call ducking

and host override are part of the VX toolkit as well, and talent
can manage and customize their telephone settings and workflow using VX Show Profiles to store and recall commonly used
show configurations.

remote control and configuration are a snap engineers can

Youll notice that there are no audio I/O or telco ports on the

work with it from any place they can get online.

VX Engine. All connections to the Engine are via the two Ethernet

Call processing is sophisticated and flexible. Lines may be readily shared among studios; the Web interface allows easy assignment of lines to shows, which can then be selected by

jacks that connect to your systems Ethernet switch to support a

wide variety of peripherals: telephone lines, Livewire studio audio, VSet phones, VX Producer PC applications, console-integrat-

users on the studio controllers. Each studio can provide its own

ed controllers, etc.

Program-on-Hold as well.

For traditional phone services, you can choose standard telco

Audio processing features also have taken a leap forward. The

gateways from Patton, Cisco, Grandstream and others to con-

processing power of the VX Engine provides a hybrid for every

nect to T1/E1, ISDN, and POTS providers. And, if you have a VoIP-

line, allowing multiple calls to be conferenced and aired simul-

based PBX or SIP Trunking telco service, the VX uses standard SIP

taneously with excellent quality. The hybrids are equipped with

(Session Initiation Protocol) and RTP (Real-time Transport Proto-

a rich toolbox to make caller audio sound its best, no matter

col) to direct up to 48 simultaneous on-air phone calls.

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VX SYSTEM COMP0NENTS
VSET12
The VSet12 phone controller is an IP-based phoneset with two large, highcontrast color LCD panels that provide line status and caller information. VSet
phones can work like a traditional Telos controller, with calls being selected,
held, and dropped in the way to which operators have grown accustomed. But
because the VX system has a hybrid per line, much more functionality is unlocked: you can now spread multiple calls over a number of faders, using one
for each call so that operators can control each lines level individually. You can
hard-assign individual lines to fixed faders, such as for VIP calls. You can even
map groups of lines to a single fader.

VSET6
VSet6 is a six-line phone controller for VX. Like the VSet12, it has a bright, attractive
LCD color display with Status Symbols that feed talent instant information about
line and caller status, and controls that enable talent to step through queued calls,
busy incoming lines, lock calls on-air, start an external recording device, et cetera.

VSET1
VSet1 is a single-line phoneset thats perfect for news booths or production facilities where multiple lines are not required. As with its multi-line brethren, VSet1
has a big LCD color display that helps users intuitively navigate through available
options, and provides information such as caller ID time ringing and time on-hold,
and even screener comments from the VX Producer software application. All VSet
phones can be powered by PoE from a Telos-approved switch, a PoE port on an Axia
console engine, or by using the included power injector.

VSET CONTROLS

TRANSFER

ADDRESS BOOK

BUSY ALL

DEVICE CONFIG.

DIALED CALLS

ON AIR

AUTO ANSWER

RECORD

CHAT

MISSED CALLS

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VX SYSTEM COMPONENTS
VSET CONTROLS
Easy-to-understand VSet controls, with a friendly visual inter-

application. Shown above are a few of the attractive, instant-

face, let talent manage incoming lines, lock calls on-air, start

ly-understandable Status Symbols that help talent run tight,

an external recording device, and take a queue of calls to air

mistake-free shows.

sequentially, for precise management of multi-call interviews

or conferences. The LCD displays deliver detailed line status,

A built-in address book and call history log round out VSet12s fea-

caller information, caller ID, time ringing-in or on-hold, and

tures. As with the rest of the VX system, each VSet12 has its own

even comments entered in the VX Producer screening software

web server for easy remote configuration and software upgrades.

VX PRODUCER
VX Producer comes with every VX Engine purchase, and has

VX Producer also has a built-in audio recorder/editor with which

a site license for deployment throughout your entire facil-

a producer can record and edit a phone call without leaving the

ity. Its intuitive, user-friendly interface provides the usual call

application. A producer can readily record calls for later play,

screening functions for phone-active broadcasts (caller status,

which can be edited with a PC application running on the same

screeners comments, Caller ID, time-per-call for up to 12 lines)

machine, then sent to the on-air studio or talents PC over the

but with a number of enhancements enabled by the IP nature

network for quick and convenient airing.

of the VX system.

Like the VSet12, VX Producer includes an address book and call

For instance, VX Producers integrated softphone uncompli-

history log. Theres also a Chat function that lets producers and

cates the producers life, since the PC interface is used for all

talent communicate quickly via off-air instant messaging. And

operations, including answering and making calls, assigning

VX Producer is included at no extra cost with every VX Engine

priority, writing notes, etc. It also reduces cost, as no hardware

purchase.

phone is needed.

CONSOLE CONTROLLERS

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VX SYSTEM COMPONENTS
CONSOLE CONTROLLERS
Live calls or pre-recorded, interviews or audience participation,

screen, and drop calls without ever diverting their attention from

one things certain: phone segments are an integral part of to-

the console. Information about line and caller status can be dis-

days fast-paced radio. But up to now, the phone system was

played right on the console as well. And soon, VX console control-

separate from the on-air console; audio was shared, but little

lers will be available for use with other console brands, too.

else. Wouldnt it be great if talent could take control of phones

without ever having to divert their attention from the board?
They can: IP-Audio networking technology provides the ideal
way to integrate broadcast phones into the on-air console the
control center of every studio.

There are plenty of other advantages to melding phones with

consoles. Like ease of installation: IP-Audio consoles with builtin phone controllers dont need any additional wires or connections. Their control signaling, caller audio and backfeeds ride
on the network connection thats already there. Bringing caller

VX connects directly to Axia Element 2.0, iQ and Radius mixing

audio into the IP-Audio domain makes it routable like any other

consoles using Livewire IP-Audio to eliminate the cost and com-

audio source. With the Virtual Mixers built into Axia consoles, you

plexity of old-style inputs, outputs, and mix-minuses. Multiple

could even choose to dynamically conference multiple lines and

phone lines each with a dedicated hybrid can automatically

control their gain with a single fader. And since the console now

map to individual console faders for complete control of caller

communicates directly with the phone hybrid, mundane tasks

audio. And users enjoy seamless console integration, with phone

such as mix-minus generation, starting recording devices, and

controls right on the board so that talent can dial, answer,

playback of recorded off-air conversations can all be automated.

VX INTERFACES
VX Audio and Logic Interfaces let you connect VX to any nonnetworked radio console or other broadcast equipment, using
standard analog or AES/EBU interfaces. A GPIO Logic interface
provides control logic where needed.

VX ANALOG AUDIO INTERFACE

The VX Analog audio interface has eight balanced stereo inputs
and eight balanced stereo outputs, presented on easy-to-install
RJ-45 connectors. The inputs are switchable to accommodate
consumer level -10dBv or professional level +4dBu. Outputs are
short-circuit protected and capable of delivering up to +24dBu
before clipping. Superior performance specs include 102dB of dynamic range, <0.005% THD.

VX AES/EBU AUDIO INTERFACE

The VX AES/EBU audio interface provides eight digital AES3 inputs and outputs, each on a separate RJ-45 connector. Studiograde performance specs, like 138dB of dynamic range and
<0.0003% THD.

VX GPIO LOGIC INTERFACE

Each VX GPIO logic interface has eight assignable logic ports.
Each port contains 5 opto-isolated inputs and 5 opto-isolated
outputs, which can be associated with audio input peripherals
and/or output destination devices to provide machine start/stop
pulses, lamp drives, and transport controls. Once a port is configured to be associated with a particular device, it automatically
activates with that device.

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THE POWER OF IP REALIZED

With VX, theres no need for the maze of discrete cables once required by a multi-line
talkshow system. All VX components are linked with standard Ethernet, so a single
CAT-5 cable provides:
Connection to the telco interface,
Line switching commands,
Data communication between the VX Engine and VSet12 phones,
Transport of caller audio to mixing consoles,
Return of mix-minus and program-on-hold audio to the caller,
Data messages (such as call notes and IM) between producer and talent,
Livewire audio for the recording of calls,
Transfer of recorded call files from the producer to the studio.
NOW HOW MANY DISCRETE CABLES DOES THAT SAVE YOU
FROM HAVING TO WIRE UP?

HOOK IT UP YOUR WAY - NON-AXIA INSTALLATION DIAGRAM

WANPORT

LANPORT

VXENGINE

ETHERNETSWITCH
VXANALOGINTERFACE

GPIOVXINTERFACE

TELCOGATEWAYORPBX

VSET12

DELAYUNIT

VSET12

CONSOLE
CONSOLE
ROUTER
ORORROUTER
LINERINGINGLIGHT
VXPRODUCER

NON-AXIA INSTALLATION DIAGRAM

Got an Axia Livewire AoIP studio network? Telos VX will plug

The Telos VX is a facility wide on-air telephone system. Facility

right in. Its the seamless integration of studio phones, mixing

wide means multiple studios, multiple stations, multiple shows

consoles and routing network youve dreamed about!

and with minimal hardware requirements. Telco is delivered via IP

Dont have IP-Audio networking yet? Not to worry VX will

work with all console brands, networked or not, via VX Audio and
Logic interfaces compact 1RU breakouts that put multiple I/O
channels right where you need them.

through a POTS, ISDN or T1 gateway device, a SIP PBX, or a dedicated IP circuit using SIP Trunking. All line and audio connectivity
is via Ethernet. This diagram shows a typical studio with an analog mixer, using VX Analog and GPIO logic interfaces to connect
to the console and other broadcast equipment.

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HOOK IT UP YOUR WAY - AXIA INSTALLATION DIAGRAM

LANPORT

WANPORT

VXENGINE

POWERSTATION

TELCOGATEWAYORPBX
LIVEWIRE-CAPABLEDELAY

VSET12

LINERINGINGLIGHT

VSET12
AXIACONSOLE

VXPRODUCER

AXIA INSTALLATION DIAGRAM

Installing VX in facilities with an Axia network requires even less

your existing Axia system just like the rest of your audio. Axia

time and hardware. The audio inputs and outputs used and pro-

console GPIO ports can be used for phone ringing tallies or re-

duced by VX are Livewire real-time audio channels and travel over

mote control of the profanity delay units.

VX GIVES YOU OPTIONS.

BROADCAST BIONICS
Broadcast Bionics offers PhoneBOX VX, a tailored-for-VX version of their original PhoneBOX
software. PhoneBOX VX gives VX users amazing amounts of information and a high level of
control over the VX system. Theres prize management, call editing and recording, sophisticated visual talkback, including a drag-and-drop database your shows calls, plus a rich phonebook and visual warnings, tied to Caller ID, for persistent or nuisance callers.
FIND OUT MORE FROM WWW.PHONEBOXVX.COM .

NEO SOFT
NeoSoft offers NeoWinners, a full-featured contest-management software that integrates
with Telos VX to add sophisticated screening and caller demographic features to your phone
segments. NeoWinners powerful suite of software tools can automatically schedule contests,
record winner information, automatically track prize inventory, and easily generate winner lists,
confirmation e-mails and even mailing labels. Contest and winner data is stored in a central,
networked database for instant access by air talent, Promotions personnel and reception staff.
VISIT WWW.NEOGROUPE.COM TO LEARN MORE.

ECHO? ITS OUTTA HERE.

The Fraunhofer AEC in the Telos VX is a remarkable new development. Its quite frankly the
most stunning echo cancellation that weve ever heard. Its performance is shockingly impressive, permitting very high loudspeaker volume with no noticeable feedback or return echo,
and it completely solves the longstanding problem of feedback and echo when a loudspeakerto-microphone acoustic path is required in the studio, such as when DJs prefer to record calls
without using headphones or when guests need to hear calls without headphones.

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FEATURES ATA GLANCE

The first VoIP telephone system designed and built specifically for broadcasting.
Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost-savings.
Standards-based SIP/IP interface integrates with most
VoIP-based PBX systems to allow transfers, line sharing
and common telco services for business and studio phones.
Standard Ethernet backbone provides a common transport
path for both studio audio and telecom needs, resulting in
cost savings and a simplified studio infrastructure. Connection of up to 100 control devices (software or hardware) is
possible.
Modular, scalable system can easily manage a whole facilitys studios, with support for up to 100 control interfaces
(VSet phones, console controllers, screening software connections).
System capacity of up to 48 standard phone lines; supports
up to 250 SIP numbers.
Up to 48 calls may be placed on-air concurrently truly a
whole-plant solution for on-air phones.
Each call receives a dedicated hybrid for unmatched clarity
and superior conferencing.
Native Livewire integration: one connection integrates
caller audio, program-on-hold, mix-minus and logic directly
into Axia AoIP consoles and networks.

Connect VX to any radio console or other broadcast equipment using available Analog, AES/EBU and GPIO interfaces.
Audio interfaces feature 48 kHz sampling rate and studiograde 24-bit A/D converters with 256x oversampling.
Powerful dynamic line management enables instant reallocation of call-in lines to studios requiring increased capacity.
VSet phone controllers with full-color LCD displays and
Telos Status Symbols present producers and talent with
a rich graphical information display. Each VSet features its
own address book and call log.
Drop-in modules can integrate VX phone control directly
into your mixing consoles.
Included VX Producer screening software with built-in softphone allows a phone connection on any networked PC.
Integrated recorder/editor simplifies recording of off-air
conversations.
Clear, clean caller audio from fifth-generation Telos Adaptive Hybrid technology, including Digital Dynamic EQ, AGC,
adjustable caller ducking, and send- and receive-audio dynamics processing by Omnia.
Wideband acoustic echo cancellation from Fraunhofer IIS
completely eliminates open-speaker feedback.
Support for G.722 HD Voice codec enables high-fidelity
phone calls from SIP clients.

FAQs
The Telos VX uses VoIP. What does that mean to me?

What about ISDN BRI and PRI lines?

Lets define what uses VoIP means. The VX uses it in two dis-

If you have ISDN now and want to keep it, again there are gate-

tinct ways: One: it can connect to Telco services using standard

ways available. However, it is often cheaper to port these num-

SIP VoIP. You benefit from having options connecting to PBXs

bers to a VoIP dial tone provider. All of these options are worth

digitally, to ISDN and analog lines via gateways, etc. With VX you

considering.

can finally integrate your on-air phones with office phone systems from a variety of vendors. Getting Telco service from VoIP

But you must have old-fashioned analog and ISDN circuit

dial tone providers means that your audio quality and hybrid null

switched connections covered somehow, eh?

will be much better as VoIP dial tone is delivered 4 wire without

Yes, VX works with these by means of gateways made by many

hum, noise, and loop loss. Building on ubiquitous VoIP standards

third-party providers such as Grandstream, Patton, and others.

means a variety of third party hardware can offer flexibility. And

See the Gateway Products and Suppliers section of the VX User

you might save a lot of money getting service this way. Two: VX

Manual for some suggestions.

system components connect to each other over standard IP/Ethernet networks with all the advantages that brings. For example,

How do I know if my existing PBX is VoIP capable?

in Livewire-equipped facilities, one RJ-45 jack connects dozens of

Since VoIP depends on SIP, and SIP is a defined standard, the only

audio channels and rich control to phones-like controllers, PC ap-

question is if your PBX can provide a trunk between itself and

plications, integrated console controllers, etc.

your VX system. If so, your PBX will work with VX but if you have
any doubts, call Telos Support and well help you figure it out.

So I can use the VX with my regular POTS lines? How? Can I use
ground start lines for incoming calls only?

My PBX is not VoIP capable. What else do I need to order, and

You can do that, and its not difficult. You only need a POTS gate-

from whom?

way device. However, we encourage using digital delivery for the

If your existing PBX cannot provide a SIP trunk line, it may be too

best sound quality.

old to offer such service, or might need upgrading. Contact your

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FAQs

PBX provider and ask if your PBX can be upgraded for use with

Codecs, you probably have noticed the same terminology issue

VoIP systems.

there. IP-based STLs over an IP T1 are just as reliable as traditional

STLs over traditional TDM T1 circuits. You are completely right to

If you do not need to connect VX to your office PBX, however, you

be concerned about a VoIP trunk (or an STL) over the Internet, as

might opt for pure SIP Trunking service. We have experience with

that is not at all the same thing, and performance in that case

several SIP Trunking providers, such as EndStream Communica-

could be variable. Inside the facility, on a LAN, all problems disap-

tions and Bandwidth.com, but you can obtain service from any

pear, since you have plenty of bandwidth and full control over

VoIP provider that will give you a my own device plan. Be aware,

the network.

though, that a simple Vonage account will not work they limit
the number of calls you can take based on the number of trunks

So shared bandwidth is the only problem with VoIP?

you have (usually, in this case, just one). This type service is a

Well, theres audio quality. In the early days VoIP used a lot of

home phone replacement not a true SIP provider.

compression, with bit rates being as low as 6kbps. Needless

to say, the resulting audio was not impressive. These low-rate

So I can experiment with SIP VoIP trunks? Why would I want

codecs have mostly fallen by the wayside. The lowest-grade co-

to do that?

dec the VX supports is g.711, the standard for digital audio in the

Actually, we recommend it! We think that youll find that they

telephone network pre-IP. And you will eventually benefit from

work better than you may have expected, as many of the VoIP

higher-fidelity codecs as these proliferate in the VoIP world.

problems we have seen are caused by limitations of analog terminal adaptors (IP to POTs gateway devices)! Since these are not

How does AoIP relate to VoIP?

needed with a VoIP-based system such as the VX, that class of

Despite the similar names and underlying technologies, they

problem is eliminated. There are a number of inexpensive ways

are very different with regard to performance and application.

to try VoIP without risk. You can also get VoIP-delivered numbers

An analog phone line and a balanced 600_ studio audio circuit

from distant area codes and exchanges. If youre paying mileage

are pretty much the same tech, but the applications and perfor-

for foreign exchange lines or have national toll free numbers,

mance are very different. AoIP has come to mean professional

youll definitely want to consider this option.

studio-grade audio networking - full-fidelity and usually with no

compression. Low-delay and synchronized channels are other

Lets say I decide to use SIP trunking. What sort of line provi-

distinguishing characteristics.

sioning do I need to order from my VoIP provider?

Unlike ISDN, SIP trunks are actually pretty simple. Your VoIP pro-

Another way the two differ is that AoIP uses an advertising/

vider will give you the IP address and registration information

discovery protocol for receivers to find sources instead of the

you will need. Once you have this, enter it into your VX Engine

Session Initiation Protocol (SIP) that VoIP employs. AoIP uses a

using the SIP Configuration web page in the VX Control Center.

system-wide clock mechanism to support low-delay and tightly

synchronized channels.

Will a SIP trunk ordered from, say, AT&T work directly with VX
as long as it has QoS?

Finally, AoIP often takes advantage of IPs multicast capability

Sure! AT&T (or any SIP Trunking provider) will provide you with

to permit multiple receivers to listen to an audio source effi-

the authentication info necessary for the connection between VX

ciently. AoIP is intended for managed, guaranteed-bandwidth

and your VoIP provider.

networks, such as with an Ethernet switch as the core of a local

area network.

I cant put a flasher across a VoIP line, so how can I flash a light
when the hotline rings?

And what about IP codecs?

This same issue arises with ISDN, so beginning with our TWOx12

Now you are very close to VoIP! These use SIP for call setup

we included a GPIO output for this function. Not to worry The

and various codecs for compression, so are similar to VoIP tele-

VX has this capability. In fact, it has multiple outputs which can

phones. In fact, they actually are VoIP telephones and can some-

be assigned to any of your VoIP lines.

times interoperate with them. They have better codecs than VoIP
phones, though. AACELD, in particular. Advanced IP codecs, such

I have been watching VoIP with interest. But reports I have

as our Z/IP, employ sophisticated technologies to overcome the

heard about services such as Vonage are that sometimes they

Internets deficiencies. Dynamic buffering, error concealment

work well, but other times not. Frankly, I am surprised to see

and more clever stuff.

Telos advocating it.

We get this concern often and understand why you ask. The term

I have read that some VoIP PBXs use IAX trunking. Can I use

Voice over Internet Protocol (VoIP) does not distinguish between

the VX with these?

VoIP over the Internet versus Voice over other (managed) IP

IAX is a protocol invented by the Asterisk people. It provides

networks. If you have been keeping up with the transition to IP

functions similar to SIP, but with more bandwidth efficiency. The

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FAQs
VX doesnt support IAX trunking at this time. But you can connect

Steve Churchs paper on the adaptive IP codec, Advanced Tech

the VX to an Asterisk with SIP trunking or as multiple SIP exten-

for IP Remotes, can be downloaded from Telos-Systems.com/

sions. Theres plenty of bandwidth on a LAN, so this works fine,

techtalk/ .

while staying with a standards-based approach. We like Asterisk as a VX adjunct. It can add voice mail, automated attendant,

Does Livewire technology come in to the VX picture?

blocking callers from caller ID, off-premise SIP extensions, and

Yup. The VX takes VoIP on the Telco side and Livewire AoIP on

more, to a VX installation. Asterisk is free Linux-based PBX soft-

the studio side. This makes integration with Axia consoles and

ware that runs on a PC. The VX and Asterisk PBX are an attrac-

networks easy and efficient. If you dont already have a Livewire

tive combo we expect will become popular within the broadcast

network, you would use Axia analog or AES audio nodes to pro-

industry.

vide I/O in either format. Each node provides eight stereo inputs
and eight stereo outputs to and from the system. Each Axia GPIO

I would like to use an Asterisk PBX with my VX system. Does

node provides 8 groups of 5 inputs and 5 outputs, covering the

Telos offer a pre-configured Asterisk installation?

needs of 8 studios. Telos Support is always available to help you

A pre-configured Asterisk box just isnt practical due to the va-

specify exactly what you need. If you are new to Livewire Technol-

riety of configuration options needed to tailor such a system

ogy you may wish to skim through our Primer at AxiaAudio.com/

to your stations specific equipment and Telco service. Well be

manuals/files/IntroToLivewire2.1.pdf .

happy to provide resources that can help you get up and running, though.

So the two can live together side-by-side

on the same LAN?

I see that Asterisk can host POTS and ISDN interface cards.

Yes they can.

Should I use those to interface my old-tech lines, or should I

use a stand-alone gateway?

So lets talk caller audio quality. What does VX offer compared

This is a matter of taste, but a general guideline is that analog

to the Nx series and your legacy products?

lines are best interfaced with an external gateway to optimize

Advanced audio processing, and the fact that you never have to

audio quality, while ISDN BRI and PRI can use Asterisk cards to

overcome Telco loop losses or extra two- to four-wire conver-

have a lower-cost solution.

sions means that the voice quality is as good as it can be. Calls
from mobile phone calls will be less than perfect at times, but VX

How can I get reliable VoIP trunk lines? What is involved? Can

extracts the best possible from them. Caller audio is maintained

you recommend a vendor?

digital and four-wire end-to-end, and VX does a clean sample-

Yes, we can assist. We have been working closely with the VX

rate conversion from the Telco rate to Livewires 48khz. There is

beta sites and other early adopters of VoIP, so we have plenty of

also a sophisticated dynamics processing section and automatic

experience to share. There are several types of VoIP dialtone pro-

EQ designed with help from our colleagues at Omnia.

viders. Youll want to consider how the service will be delivered

to you; via the Internet (like Vonage), or via a dedicated IP circuit

How are callers on the VoIP trunks going to sound? What

from the provider that includes a Service Level Agreement and

about echo? Wont cell phones sound even worse than usual?

guaranteed Quality of Service (as offered by a number of vendors

The VX has enough processing horsepower to deal with even ex-

including most of the traditional Telcos). For discussion of this

treme echo situations, and four-wire Telco delivery means that

and other matters, check the Telos web site on a regular basis as

the only external echo path is from the callers line, when its

we continue post material on this and related topics.

analog, and the callers handset. The VX uses Telos latest hybrid
technology (5th generation), enhanced with the latest state of

I know that SIP is supported by the new IP codecs. Will the VX

the art acoustic echo cancellation. Even when using open speak-

be able to connect to my Zephyr/IP in the field? Other codecs?

ers, and changing levels during a call, the new algorithm makes

As we hinted above, Yes! The VX supports g.722 (7khz, wideband

feedback nearly impossible.

audio and g.711 (3.4khz, phone quality).

I heard that amazing Acoustic Echo Canceller on your Axia inWhat about SIP, SDP, RTP, ENUM and UDP?

tercom system. No feedback at all with the speakers blasting

We know that engineers are lifetime learners and encourage that.

and the mic a few inches away. Why dont you put that in the

However, just as you probably dont know much about SS7 or

VX so that DJs can answer calls on the cue speaker without

IUP in the telephone network, understanding these details is

feedback?

optional. We do have White Papers on our web site to educate

Done! The VX has it.

you on these, and other, terms. Start with the one below. You
could also read Steve and Skips AoIP book, too, for a fun and
comprehensive coverage of this stuff.

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FAQs
Steve Church once told me that IP cell phones can sound

The user interface lets operators assign selectable lines to mul-

better than usual 3 kHz circuit-switched phone technology -

tiple faders.

something about G.722 dot something. Is this true?

Right. Current Cisco VoIP phones, for example, support the

If I use an Axia console, it gets even better?

g.722 codec. The VX supports this, as well. However, Steve was

Yes thats the ultimate. You start with the most flexible con-

probably referring to AMR Wide Band also known as G.722.2.,

sole/audio-platform and then add smoothly integrated phones

sometimes called HD Audio. Its 7khz and doesnt sound at all

with the IP network powering it all. Sweet! The network delivers

like phone audio - in fact, it sounds better than regular G.722!

any of your Telco lines to any of your studios, in any combination.

AMR-WB is part of the new ITU standard for mobiles, so should

Any line is available in any studio at any time.

grow over time. Meanwhile, some IP-based apps for mobiles are
starting to use wideband codecs, such as MPEG-ELD in Apples

I notice the VX engine has both a LAN and WAN connector;

facetime app.

why is that?
Its a built-in firewall, isolating the VoIP connection from your

OK, I am starting to see the light. Cool stuff, but wheres the

studio network. We use this same approach in the iPort Livewire-

catch? Is VX hard to install and configure?

WAN MPEG gateway.

Setup is via web. It may be little different than what youre used
to (or not) but its not difficult, and some customers never crack

What about call screener and database functions?

open the book to set it up. Power and flexibility do come with

A basic Call Screening app, VX Producer, comes with the sys-

a little complexity, but well always be at your side should you

tem. Other networked, PC-based apps, such as Broadcast Bion-

need us.

ics PhoneBOX VX or NeoGroupes applications put information

about your callers in front of your producers without the need for

There is no such thing as a free lunch - it must be hard to use

caller ID boxes, serial cables or other hassles.

then. I know theres a catch... I really dont have time to explain a new system to the air staff.

What about SMS messages and chat - Can they be integrated

We know! Rest assured its easier for your talent, not harder! We

into my phone system?

recognize that any time you change anything in a studio, there

Using Broadcast Bionics Phone Box VX, yes! Telos has always built

can be some transition time. While there are a lot of new features

open systems to allow others to create their own visions around

in the VX, your staff can use the basic stuff immediately because

our gear.

it works just like familiar and comfortable Telos gear. The color,
hi-rez LCDs and seamless console integration (to Axia Element

OK, so the catch has to be the price?

and iQ) enhance the user experience. As you read this, systems

The VX lets you leverage cheap networking to serve your entire

around the world are screening calls and putting them on the air

facility. Since you dont need hardware boxes for each studio,

without drama. Jocks and Talk hosts alike praise the VX! Opera-

cost is surprisingly reasonable. Youll use the VX in your on-air

tors familiar with our longstanding two-column line selection will

studios to replace older multi-line systems, and youll use it to

be right at home.

replace hybrids in newsrooms and production studios. You might

also decide to eliminate walls full of couplers for pre-delay IFB

Is there any support for VIP lines?

dial-up lines, and transitioning your Telco service might save you

Yes. We call these fixed lines, and you can have as many as you

a lot of scratch. Weve seen stations saving thousands of dollars

want. They are used for callers who stay on-air while other callers

a month (no kidding) by eliminating POTS lines, with their taxes

are coming and going.

and fees.

What about conferencing?

Anything else cool about the VX?

This is one of the strengths of the VX. Since VoIP calls are four-

Did we mention the color LCD user interface on the new VSet

wire, multiple lines can be conferenced with very high quality.

phone/control surface? Producers and talent love it!

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TELOS VX SPECIFICATIONS

SYSTEM
hh +Maximum number of phone lines: 48, when used

with aLaw or uLaw codecs for VoIP lines. (Higher

quailty codecs, such as G.722, consume more system
resources and result in a decreased number of total
lines available.)

hh Maximum number of SIP numbers: 250

hh Maximum active on-air calls: 48
hh Maximum number of simultaneous audio

connections (Livewire I/O channels): 16 systemwide

hh Maximum on-air calls on one fader: 4

AUDIO PERFORMANCE
ANALOG LINE INPUTS

hh Input Impedance: >40 k ohms, balanced

hh Nominal Level Range: Selectable, +4 dBu or -10dBv
hh Input Headroom: 20 dB above nominal input
ANALOG LINE OUTPUTS

hh Output Source Impedance: <50 ohms balanced

hh Output Load Impedance: 600 ohms, minimum
hh Nominal Output Level: +4 dBu
hh Maximum Output Level: +24 dBu
DIGITAL AUDIO INPUTS AND OUTPUTS

hh Reference Level: +4 dBu (-20 dB FSD)

hh Impedance: 110 Ohm, balanced (XLR)
hh Signal Format: AES-3 (AES/EBU)
hh AES-3 Input Compliance: 24-bit with selectable
sample rate conversion, 32 kHz to 96kHz input
sample rate capable.

hh AES-3 Output Compliance: 24-bit

hh Digital Reference: Internal (network timebase)
or external reference 48 kHz, +/- 2 ppm

hh Internal Sampling Rate: 48 kHz

hh Output Sample Rate: 44.1 kHz or 48 kHz
hh A/D Conversions: 24-bit, Delta-Sigma,
256x oversampling

FREQUENCY RESPONSE

hh Any input to any output: +0.5 / -0.5 dB,

20 Hz to 20 kHz

DYNAMIC RANGE

hh Analog Input to Analog Output: 102 dB referenced to

0 dBFS, 105 dB A weighted to 0 dBFS

hh Analog Input to Digital Output: 105 dB referenced to

0 dBFS

hh Digital Input to Analog Output: 103 dB referenced to

0 dBFS, 106 dB A weighted

hh Digital Input to Digital Output: 138 dB

TOTAL HARMONIC DISTORTION + NOISE

hh Analog Input to Analog Output: <0.008%,

1 kHz, +18 dBu input, +18 dBu output

hh Digital Input to Digital Output: <0.0003%,

1 kHz, -20 dBFS

hh Digital Input to Analog Output: <0.005%,

1 kHz, -6 dBFS input, +18 dBu output

CROSSTALK ISOLATION AND STEREO SEPARATION AND CMRR

hh Analog Line channel to channel isolation: 90 dB isolation

minimum, 20 Hz to 20 kHz

hh Analog Line Stereo separation: 85 dB isolation minimum,

20Hz to 20 kHz

hh Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz

hh D/A Conversions: 24-bit, Delta-Sigma,

256x oversampling

hh Latency <3 ms, mic in to monitor out,

including network and processor loop

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TELOS VX SPECIFICATIONS

VX ENGINE
IP/ETHERNET CONNECTIONS

hh One 100BaseT/gigabit Ethernet via RJ-45

LAN connection

hh One 100BaseT/gigabit Ethernet via RJ-45

WAN connection

hh Power consumption: 100 Watts

OPERATING TEMPERATURES

hh -10 degree C to +40 degree C, <90% humidity,

no condensation

PROCESSING FUNCTIONS

hh All processing is performed at 32-bit

floating-point resolution.

hh Send AGC/limiter

DIMENSIONS AND WEIGHT

hh 3.5 inches x 17 inches x 15 inches, 10 pounds

STUDIO AUDIO CONNECTIONS

hh Send filter

hh Via Livewire IP/Ethernet. Each selectable group and

fixed line has a send and receive input/output.

hh Gated Receive AGC

hh Each studio has a Program-on-Hold input.

hh Receive filter

hh Each Acoustic Echo Canceller has two inputs

hh Receive dynamic EQ

(signal and reference) and one output.

hh Ducker

hh LW-equipped studios may take the audio directly from

the network. Interface Nodes are available for pro
analog and AES3. (uses standard Livewire Nodes)

hh Sample rate converter

hh Line Echo Canceller (hybrid)
hh Acoustic Echo Canceller (wideband)

TELCO CONNECTIONS

hh Audio: standard RTP. Codecs: g.711u-Law and A-Law,

and g.722.

POWER SUPPLY AC INPUT

hh Auto-sensing supply, 90VAC to 240VAC,

hh Control: standard SIP trunking

50 Hz to 60 Hz, IEC receptacle, internal fuse

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