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Telos VX Prime Plus VOIP Phone System

VX Prime PLus VoIP Phone System - 8 fixed hybrids 2001-00510


Telos VX Prime Plus
Telos VX Prime PlusTelos VX Prime PlusTelos VX Prime Plus
*VSET6 Sold Separately

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. 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.

Main 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.*


• 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.

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