VX connects to traditional POTS and ISDN telephone lines via standard Telco gateways. But it also connects to VoIP-based PBX systems and SIP Trunking services to take advantage of low-cost Internet-delivered phone services.
VX also weds modern networking to the remarkable power of digital signal processing. VX uses Ethernet as its connection backbone, significantly cutting the cost of phone system installation, maintenance and cabling. Ethernet is a powerful, yet simple way to share phone lines among studios and connect system components. This also makes VX naturally scalable, capable of serving even the largest of facilities — while remaining surprisingly cost-effective for even single stations with more modest needs.
Don’t have an IP-Audio network yet? No problem; optional Telos Alliance xNodes, like the Telos Alliance Mixed Signal Node, break out audio into analog and digital formats, along with GPIO logic commands.
- World’s first VoIP telephone system designed and built specifically for broadcasting.
- 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 be easily expanded to manage a network of up to 20 studios, each with a dedicated Program-On-Hold input – truly a “whole-plant” solution for on-air phones.
- System capacity of up to 48 standard phone lines; supports up to 250 SIP numbers.
- Up to 24 hybrids, depending on ultimate system configuration.
- 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 Telos Alliance AES/EBU, Mixed Signal, and GPIO xNodes. Audio interfaces feature 48 kHz sampling rate and studio-grade 24-bit A/D converters with 256x oversampling.
- Powerful dynamic line management enables instant re-allocation 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 Axia mixing consoles.
- Included XScreen screening software with built-in soft-phone 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 codec enables high-fidelity phone calls from SIP clients.
- Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost savings, via standard Telco gateways
- Telos 5th-generation Adaptive Digital Hybrids.
- Maximum number of phone lines: 48, when used with a-Law or u-Law codecs for VoIP lines. (Higher quality codecs, such as G.722, consume more system resources and result in a decreased number of total lines available.)
- Maximum number of SIP numbers: 250
- Maximum active on-air calls: 48
- Maximum on-air calls on one fader: 4
Analog Inputs (with Telos Alliance xNode):
- Input Impedance: >40 k Ohms, balanced
- Nominal Level Range: Selectable, +4 dBu or -10dBv
- Input Headroom: 20 dB above nominal input
Analog Outputs (with Telos Alliance xNode):
- 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)
- 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
- Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz
- 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
- One 100BaseT/gigabit Ethernet via RJ-45 LAN connection
- One 100BaseT/gigabit Ethernet via RJ-45 WAN connection
- All processing is performed at 32-bit floating-point resolution.
- Send AGC/limiter
- Send filter
- Gated Receive AGC
- Receive filter
- Receive dynamic EQ
- Sample rate converter
- Line Echo Canceller (hybrid)
- Acoustic Echo Canceller (wideband)
Power Supply AC Input
- Modular, field-replacable auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
- Power consumption: 100 Watts
- -10 degree C to +40 degree C, <90% humidity, no condensation
Studio Audio Connections
- Via Livewire+ IP/Ethernet. Each selectable group and fixed line has a send and receive input/output.
- Each studio has a Program-on-Hold input.
- Each Acoustic Echo Canceller has two inputs (signal and reference) and one output.
- Livewire+ equipped studios may take the audio directly from the network. Telos Alliance xNodes are available for pro analog and AES3 breakout.
- Audio: standard RTP. Codecs: g.711u-Law and A-Law, and g.722
- Control: standard SIP trunking