Telos VX Prime Broadcast VoIP Telefoon systeem
Het Telos VX Prime Broadcast VoIP Telefoon Hybride Systeem is het kleinere broertje van het Telos VX ststeem maar dan met maximaal 8 lijnen. De perfecte oplossing voor de wat kleinere regio /lokale omroep.
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Telos VX Prime Broadcast VoIP Telefoon Hybride Systeem
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 two to four 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 also 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.
Why VX Prime?
Cost-Efficient Way to Upgrade to IP
- Lower cost alternative to full VX Broadcast System
- Potentially save thousands monthly on expensive ISDN/POTS lines
- Ideal for smaller studios (2-4 Stations) & smaller budgets
- Native support of G.722 “HD Voice” codec
- Smart AGC ensures consistent caller audio levels
- Digital Dynamic EQ (DDEQ) by Omnia adjusts EQ automatically to ensure call-to-call consistency and the best intelligibility
- Connects to your existing Livewire network with a single Ethernet cable
- Non-Livewire studios can use Telos Multipurpose Node audio interface for audio and GPIO connectivity to studio consoles
- Provides phone hybrids for each of your studios without need for any additional wiring or physical audio connections
- No restriction to the number of SIP lines or phone numbers that can come into the system
- Eight fixed hybrid/faders (not expandable)
- Industry-leading 24/7 support
- Telos standard 5-year warranty
Why VoIP For Broadcast?
VoIP has taken the business world by storm, increasing the flexibility of office phone systems and PBXs while simultaneously lowering maintenance and equipment costs. In fact, most Fortune 500 companies have replaced their old PBX systems with VoIP for just these reasons.
VoIP is a natural for broadcasters, interconnecting the phone system with audio interfaces, phone sets, console controllers, and PCs running screening software by way of efficient, low-cost Ethernet. Using VoIP, you can finally share phone lines among multiple studios and route caller audio anywhere in your facility, easily, and instantly. Got a hot talk-show that suddenly needs more lines in a certain studio? Just a few keystrokes at a computer and you’re ready—no delays, and no cables to pull. VX systems can even interconnect with your business office’s VoIP PBX to allow easy call transfers.
Reduced Cost. Increased Flexibility.
The use of sophisticated, modern IP networking for Telos VX Prime allows rich communication between devices. For example, caller information entered by a producer is displayed on the studio phone’s color LCD. Caller audio is available on studio PCs for easy recording. Operators at mixing consoles can directly control line switching without diverting their attention from the board. The result? Talk shows that run like clockwork, sound better, and flow without errors.
This standards-based VoIP architecture helps you save money, too, by widening your choices in telco providers. Most carriers now offer VoIP services using the SIP protocol, which can deliver substantial savings to stations that need any number of lines. (You can also connect to traditional T-1/PRI, POTS or ISDN phone lines using open-source Asterisk-based phone servers.)
But VX systems don’t stop at providing the benefits of VoIP—they also carry the broadcast-phone technology expertise of Telos.
Every incoming line has its own fifth-generation Telos Adaptive Digital Hybrid, our most advanced ever—packed full of technology engineered to extract the cleanest, clearest caller audio from just about any phone line, even cellular calls. Multiple lines can be conferenced with superior clarity and fidelity. Smart AGC ensures consistent caller audio levels. And calls from mobile callers using SIP clients on their smartphones benefit from native support for the G.722 “HD Voice” codec, improving caller speech quality and intelligibility.
The main component of the VX Prime system is the VX Prime Engine. It works together with Vset telephones, Telos Call Controllers, and Telos Audio, GPIO, or Multipurpose Nodes/Interfaces to make a complete system, all sold separately.
VX PRIME FEATURES
- A true VoIP telephone system designed and built specifically for broadcasting; VX Prime is ideal for small to medium studios with two to four stations.
- SIP call-handling throughout—no internal conversion to analog call handling like some other so-called “VoIP” systems.
- Works with T1/E1, ISDN, SIP telco services and even POTS for maximum flexibility and cost-savings.
- 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 eight 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 Multipurpose, AES/EBU, and GPIO interfaces. Audio interfaces 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 (sold separately) 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 Call Controller modules can integrate VX phone control directly into your 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.
- Support for G.722 “HD Voice” codec enables high-fidelity (7 khz) phone calls from SIP telephone sets and softphones.
VX PRIME SPECIFICATIONS
- 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
- 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
VX Prime Engine
- One 1 Gigabit Ethernet via RJ-45 LAN connection (livewire)
- One 1 Gigabit Ethernet via RJ-45 WAN Connection (SIP provider)
- 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)
- 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
- -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.
- Audio: standard RTP. Codecs: G.711u-Law and A-Law, and G.722.
- Control: standard SIP endpoint