VX Prime+ Broadcast VoIP Phone System

List: $5,395.00

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SKU: TEL-200100510 Category:
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Description

VX PRIME+ Broadcast VOIP Phone System

 

Telos VX® talk-show systems are the world’s first true VoIP-based broadcast phone systems and have been proven to deliver the power of VoIP to the broadcast studio like no other. The Telos VX Prime+, with built-in support for AES67, is the next evolution of Telos VX VoIP phone systems in a powerful new 1RU hardware unit. Additionally, support for the G.722 voice codec ensures the highest quality calls from supported mobile devices. With capacity of 8 fixed hybrids/faders, VX Prime+ is ideal for facilities with 2 to 4 studios. (For larger facilities, check out VX Enterprise with up to 120-hybrid capacity.)

AES67 support brings a new level of compatibility and flexibility to VX phone systems. Support for AES67 gives broadcasters the flexibility of integrating VX Prime+ into any AES67 environment, in addition to our own Axia® Livewire® network. With plug-and-play connectivity, you can network multiple channels of audio with any manufacturer’s AES67-compliant hardware. Beyond AES67, Livewire users have the added convenience and power of networking control (GPIO), advertising/discovery, and program associated data throughout the network.

Using VoIP, VX Prime+ gives you remarkable-sounding on-air phone calls with no ‘gotchas’. It uses standard SIP protocol that works with many VoIP PBX systems and SIP Telco to take advantage of low-cost and high-reliability service offerings. VX Prime+ can also connect to traditional telco lines via Asterisk PBX systems, which can be customized for specific facility requirements.

VX Prime+ gives you incredible operational power, flexible, adaptable workflows, and superior audio quality, while making it easier than ever for talent to have complete mastery of their callers. With VX Prime+, the world’s leading broadcast phone system is now available to those with smaller budgets, offering Big Performance for Small Facilities.

 

 

Features

 

A true VoIP telephone system designed and built specifically for broadcasting; VX Prime+ is ideal for small to medium studios with 2 to 4 studios.

Includes support for AES67, giving broadcasters added flexibility of integrating VX Prime+ into any AES67 network, in addition to our own Axia Livewire network.

SIP call-handling throughout—no internal conversion to analog call handling like some other so-called “VoIP” systems.

Standards-based SIP interface integrates with Asterisk open-source SIP phone servers and most VoIP-based PBX systems to allow transfers 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.

System capacity of 8 hybrids. Each call placed on the air 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 systems to any third-party radio console or other broadcast equipment using available Telos Alliance Mixed Signal, AES/EBU, and GPIO xNodes. xNodes feature 48 kHz sampling rate and studio-grade 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.

The “Drop-in” Vset Call Controller™ modules can integrate VX phone control directly into your mixing consoles.

XScreen Lite screening software included.

Clear, clean caller audio from 5th-generation Telos Adaptive Hybrid technology, including Digital Dynamic EQ, AGC, adjustable caller ducking, and send- and receive-audio dynamics processing by Omnia.

Support for G.722 codec enables high-fidelity phone calls from iPhone and Android SIP softphones using an SIP server.

Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker feedback.

Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost savings, via Asterisk servers.*

*Due to the wide variation in how traditional phone service can be delivered, and the complexities that can be involved in converting those services to SIP, we really want to talk with you about your system design before you order. Telos has VX System engineers standing by to help you draw up a configuration that will ensure your VX purchase will perform to your expectations when using traditional POTS and ISDN lines.

Specifications

 

System

  • Maximum number of simultaneous calls on-air, VX Prime+: 8 (more with conferencing)
  • Maximum number of SIP numbers, VX Prime+: 96

 

Audio Performance (Node)

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

 

VX Prime+ Engine

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

 

Dimensions and Weight

  • One rack unit - 1.75" H x 19" W x 15.5" D (44 x 483 x 394 mm)

 

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 Alliance xNodes are available for professional-level analog and AES3 connection breakouts for clients without Livewire AoIP networking.
  • VX Prime+ supports AES67 connectivity.

 

Telco Connections

  • Audio: standard RTP. Codecs: G.711u-Law and A-Law, and G.722.
  • Control: standard SIP endpoints, ISDN PRI/T-1, ISDN BRI and POTS may be supported with the appropriate interfaces using an Asterisk Open source PBX.