Telos 2001-00378 Zephyr iPort PLUS Multi-Codec Gateway & Content Delay

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SKU:
2001-00378
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true
old_prod_key:
22087218265842CBA291CCEE15E6EE21
Support Documentation:
Product Info_Telos_Zephyr iPort PLUS.pdf

Zephyr iPort PLUS is a networked multi-codec gateway that enables transport of multiple channels of stereo audio across any QoS-enabled IP network, including T1 and T3 connections and private WANs with MPLS - perfect for large-scale distribution of audio to single or multiple locations.

Zephyr iPort PLUS is the workhorse of codecs, configurable as eight stereo bi-directional MPEG codecs, or for decode of up to 16 MPEG-encoded stereo streams. Adding aptX® or using Linear PCM coding can enable up to 16 bidirectional or unidirectional codecs. Zephyr iPort PLUS connects to Axia® IP-Audio networks using a single CAT-6 cable for all I/O. Don't have a Livewire network yet? Pair Zephyr iPort PLUS with Axia xNode audio interfaces for use as a standalone multiple-stream codec.

Coding algorithms include AAC, AAC-LD, HE-AAC (plus v2), MP2, MP3, linear, and optional aptX® Enhanced*. Bit rates range from 24 to 320 kbps for MPEG codecs, plus standard fixed rates for aptX and linear to over 2 Mbps. In addition, iPort offers dual, parallel-path end-to-end streaming for ultra- reliability and redundancy. For network operators, a unique Content Delay feature allows independent local storage and scheduled delayed playout of any or all coded audio channels for up to six hours.

  • Distributes multiple channels of coded audio between broadcast facilities over QoS-enabled IP links.
  • Configurable as a CODEC with 8 bi-directional MPEG-coded channels, each with GPIO and PAD - or, as a 16-channel stereo encoder (by adding aptX and/or Linear PCM-coded channels) or 16-channel stereo decoder.
  • PCM Stereo coded channels are available for use simultaneously alongside MPEG coded channels, (dependent upon available bandwidth).
  • Can also deliver streaming audio channels for Internet transmission via SHOUTcast,
  • Steamcast or compatible stream replication server.
  • Wide choice of genuine Fraunhofer codecs, including Standard AAC, high-efficiency AAC-HE (aacPlus), AAC-HEv2, low-delay AAC-LD, and MP3, with a choice of bit rates from 24 kbps to 320 kbps, definable per stream.
  • Optional aptX® Enhanced audio coding may be ordered at time of purchase or added later, as desired.
  • When used as part of a Livewire network, allows audio from remote facilities to be used as if they were local sources, with associated logic and control.
  • Eight 5-input Virtual Mixer (VMIX) channels each allow combining and mixing of up to 5 networked Livewire audio streams on a single channel.
  • Eight Virtual Mode (VMODE) channels allow audio to be split into left/right channels, summed L+R, and more, prior to encoding and transmission.
  • Content Delay option enables delayed playout of any or all selected receive audio channels, along with time-synchronized ancillary data, for up to six hours. Each playback delay time is independently configurable on a per-channel basis, making Zephyr iPort PLUS ideal for network operators, program distribution networks, or delayed playout of received audio at network-affiliated stations.
  • Remote control/configuration via any computer with a standard Web browser.
  • Separate LAN and WAN ports help ensure network security.
  • Fanless, convection-cooled DSP-powered platform with dual-redundant, auto-switching power supplies for maximum uptime. Power supply modules are field-replaceable in minutes.

Audio

  • Zephyr iPort PLUS has no native audio I/O, operating on streams provided by attached
  • Livewire+ audio devices. All audio specifications below are representative of Axia Livewire+ audio interfaces.

 Analog Line Inputs

  • Input Impedance: >40 k ohms, balanced 
  • Nominal Input Range: Selectable, +4 dBor -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 dB(-20 dB FSD) 
  • Impedance: 110 Ohm, balanced (XLR) 
  • Signal Format: AES3 (AES/EBU)
  • AES3 Input Compliance: 24-bit with selectable sample rate conversion, 32 kHz to 96 kHz input sample rate capable.
  • AES3 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
     

Frequency Response 

  • Any input to any output: +/- 0.5 dB, 20 Hz to 20 kHz
     

Network 

  • 1 LAN port, 1 WAN port; 100/1000-BaseT Ethernet interfaces
     

Codecs

  • Standard AAC, high-efficiency AAC-HE (aacPlus), AAC-HEv2, low-delay AAC-LD, MP3, MP2.
  • Optional: Enhanced aptX® from CSR. 

Power 

  • Dual-redundant internal auto-ranging power supplies, 90 – 132 / 187 – 264 VAC, 50Hz/60Hz.
  • Power consumption: 100 Watts.