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VX Prime

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Description

Product Description

Telos VX talk-show systems are the world’s only true VoIP-based broadcast phone systems. The VX Prime gives you incredible operational power, flexible, adaptable workflows, and superior audio quality—a powerful broadcast phone solution that’s economical enough for stations with just two or three studios. VX Prime connects to VoIP-based PBX systems and SIP carriers to take advantage of low-cost and high reliability service offerings. VX Prime makes it easier than ever for talent to have complete mastery of their callers. No other broadcast phone system delivers the power of VoIP to the broadcast studio like Telos VX. With VX Prime, the world’s leading broadcast phone system is now available to those with smaller budgets, giving you big studio sound at a small studio prices. Simply put, you’re paying for the capability you need, and nothing extra.

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Additional Information

System

Maximum number of simultaneous calls on-air, VX Prime: 8 (more with conferencing)

Maximum number of SIP numbers, VX Prime: 96

Analog Line Inputs

Input Impedance: >40 k ohms, balanced

Nominal Level Range: Selectable, +4 dBu or -10dBv

Input Headroom: 20 dB above nominal input

Analog Line Outputs

Output Source Impedance: <50 ohms balanced

Output Load Impedance: 600 ohms, minimum

Nominal Output Level: +4 dBu

Maximum Output Level: +24 dBu

Digital Audio Inputs And Outputs

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

Impedance: 110 Ohm, balanced (XLR) h Signal Format: AES-3 (AES/EBU)

AES-3 Input Compliance: 24-bit with selectable sample rate conversion, 32 kHz to 96kHz input sample rate capable.

AES-3 Output Compliance: 24-bit

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

Internal Sampling Rate: 48 kHz

Output Sample Rate: 44.1 kHz or 48 kHz

A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling

D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling

Latency <3 ms, mic in to monitor out, including network and processor loop

Frequency Response

Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz

Dynamic Range

Analog Input to Analog Output: 102 dB referenced to 0 dBFS, 105 dB “A” weighted to 0 dBFS

Analog Input to Digital Output: 105 dB referenced to 0 dBFS

Digital Input to Analog Output: 103 dB referenced to 0 dBFS, 106 dB “A” weighted

Digital Input to Digital Output: 138 dB

Total Harmonic Distortion + Noise

Analog Input to Analog Output: <0.008%, 1 kHz, +18 dBu input, +18 dBu output

Digital Input to Digital Output: <0.0003%, 1 kHz, -20 dBFS

Digital Input to Analog Output: <0.005%, 1 kHz, -6 dBFS input, +18 dBu output

Crosstalk Isolation, Stereo Separation And CMRR

Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kHz

Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz

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

IP/Ethernet Connections

One 1 Gigabit Ethernet via RJ-45 LAN connection (livewire)

One 1 Gigabit Ethernet via RJ-45 WAN Connection (SIP provider)

Processing Functions

All processing is performed at 32-bit floating-point resolution.

Send AGC/limiter

Send filter

Gated Receive AGC

Receive filter

Receive dynamic EQ (3 band)

Ducker

Sample rate converter

Power Supply AC Input

Hot-swap capable dual-redundant internal auto-ranging power supplies. 90 – 132 / 187 – 264 VAC, 50Hz/60Hz. IEC receptacle, internal fuse.

Power consumption: 100 Watts

Operating Temperatures

-10 degree C to +40 degree C, <90% humidity, no condensation

Fanless, convection-cooled

Dimensions and Weight

Rackmount, 2RU

3.5 inches x 17 inches x 15 inches

10 pounds

Studio Audio Connections

Via Livewire Ethernet. Each selectable group and fixed line has a send and receive input/output.

Each studio may be configured with its own Program-on-Hold input.

Livewire-equipped studios take audio directly from the network.

Telos Audio Interface Nodes are available for professional-level analog and AES3 connection breakouts for clients without Livewire AoIP networking.

Telco Connections

Audio: standard RTP. Codecs: G.711u-Law and A-Law, and G.722.

Control: standard SIP endpoint

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