Telos VX is the worlds first VoIP (Voice over IP) talkshow system a broadcast phone system thats so powerful, it can run all the on-air phones for your entire plant. Telos VX Enterprise, with built-in support for AES67, is the next evolution of Telos VX VoIP phone system 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 expandable up to 120 hybrids/faders, VX Enterprise is ideal for medium to large facilities and can grow with your station over time. (For smaller facilities, check out VX Prime+ with 8-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 Enterprise into any AES67 environment. With plug-and-play connectivity, you can network multiple channels of audio with any manufacturers 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 Enterprise gives you remarkable-sounding on-air phone calls with no gotchas. VX Enterprise weds modern networking to the remarkable power of digital signal processing. VX Enterprise uses Ethernet as its connection backbone, significantly cutting the cost of phone system installation, maintenance, and cabling. 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 Enterprise can also connect to traditional telco lines via Asterisk PBX systems, which can be customized for specific facility requirements.Dont have an IP-Audio network yet? Optional Telos Alliance xNodes, like the Telos Alliance Mixed Signal Node, break out audio into analog and digital formats, along with GPIO logic commands. And with informative VSet phones, talent finds it easier than ever to take control of their callers, moving and sharing lines between studios at the touch of a button.Main Features:VX is the worlds first VoIP telephone system designed and built specifically for broadcasting.Includes support for AES67, giving broadcasters added flexibility of integrating VX Enterprise into any AES67 network, in addition to our own Axia Livewire network.Works directly with SIP endpoint telco or PBX services, and in conjunction with a PBX may support POTS, T1/E1, and ISDN BRI for maximum flexibility and cost savings.*Standards-based SIP/IP interface integrates with most VoIP-based PBX systems to allow transfers, line-sharing, Caller ID 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 and configured to manage a network of 60 or even more studios, each with a dedicated Program-On-Hold inputtruly a whole-plant solution for on-air phones.Base system is licensed for up to 24 hybrids, May be expanded in license increments of 8 up to a total of 120 hybrids. A Telos system engineering consultation is required for any system configuration over 72 hybrids.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 Enterprise 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 reallocation of call-in lines to studios requiring increased capacity.VSet Call 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 Enterprise phone control directly into your Axia 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.Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker feedback.Support for G.722 codec enables high-fidelity phone calls from iPhone and Android SIP softphones using an SIP server.Specifications:GeneralTelos 5th-generation Adaptive Digital HybridsMaximum number of hybrids: 120, 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: unlimitedMaximum active on-air calls: 120 Maximum on-air calls on one fader: 12Analog Inputs (with Telos Alliance xNode)Input Impedance: >40 k Ohms, balancedNominal Level Range: Selectable, +4 dBu or -10dBvInput Headroom: 20 dB above nominal inputAnalog Outputs (with Telos Alliance xNode)Output Source Impedance: <50 Ohms balancedOutput Load Impedance: 600 Ohms, minimumNominal Output Level: +4 dBuMaximum Output Level: +24 dBuDigital Audio Inputs and OutputsReference 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 capableAES-3 Output Compliance: 24-bit Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppmInternal Sampling Rate: 48 kHzOutput Sample Rate: 44.1 kHz or 48 kHzA/D Conversions: 24-bit, Delta-Sigma, 256x oversamplingD/A Conversions: 24-bit, Delta-Sigma, 256x oversamplingLatency <3 ms, mic in to monitor out, including network and processor loopFrequency ResponseAny input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHzDynamic RangeAnalog Input to Analog Output: 102 dB referenced to 0 dBFS, 105 dB A weighted to 0 dBFSAnalog Input to Digital Output: 105 dB referenced to 0 dBFSDigital Input to Analog Output: 103 dB referenced to 0 dBFS, 106 dB A weightedDigital Input to Digital Output: 138 dBTotal Harmonic Distortion + NoiseAnalog Input to Analog Output: <0.008%, 1 kHz, +18 dBu input, +18 dBu outputDigital Input to Digital Output: <0.0003%, 1 kHz, -20 dBFSDigital Input to Analog Output: <0.005%, 1 kHz, -6 dBFS input, +18 dBu outputCrosstalk Isolation, Stereo Separation, and CMRRAnalog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kHzAnalog Line Stereo separation: 85 dB isolation minimum, 20 Hz to 20 kHzAnalog Line Input CMRR: >60 dB, 20 Hz to 20 kHzVX Enterprise IP/Ethernet ConnectionsOne 100/1000BASE-T Ethernet via RJ-45 LAN connectionOne 100/1000BASE-T Ethernet via RJ-45 WAN connectionProcessing FunctionsAll processing is performed at 32-bit floating-point resolutionSend AGC/limiterSend filterGated Receive AGCReceive filterReceive dynamic EQDuckerSample rate converterLine Echo Canceller (hybrid)Acoustic Echo CancellerPower Supply AC InputHot-swap capable dual-redundant internal auto-ranging power supplies. 90 132 / 187 264 VAC, 50Hz/60Hz. IEC receptacle, internal fuse.Power consumption: 150 WattsOperating Temperatures-10 degree C to +40 degree C, <90% humidity, no condensationDimensions and WeightOne rack unit - 1.75H x 19W x 15.5D (44 x 483 x 394 mm)Studio Audio ConnectionsVia Livewire IP/Ethernet. Each selectable group and fixed line has a send and receive input/outputEach studio has a Program-on-Hold inputEach Acoustic Echo Canceller has two inputs (signal and reference) and one outputLivewire+ AES67 equipped studios may take and supply audio directly to/from the network. Telos Alliance xNodes are available for pro analog and AES3 breakout.VX Enterprise supports AES67 connectivity.Telco ConnectionsAudio: standard RTP. Codecs: g.711-Law and A-Law, and g.722.Control: standard SIP Endpoint, ISDN PRI/T-1, ISDN BRI and POTS may be supported with the appropriate interfaces using an Asterisk Open source PBX.