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.
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 |