- 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.*
- 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.
