Telos Zephyr iPort PLUS – Multi-CODEC Gateway (w/ Time Zone Delay)

Zephyr iPort is a Livewire-to-MPEG 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 is the workhorse of codecs, configurable as eight stereo bi-directional MPEG codecs, or for encode / decode of up to 16 uni-directional stereo streams.

Zephyr iPort 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 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 Time Zone Delay feature allows independent delay of any or all coded audio channels for up to six hours.

Features:
– Connects two or more Livewire-equipped facilities over a wide-area network with QoS.
– Transmits/receives as many as 8 bi-directional IP channels per encoder, each with GPIO and PAD.
– Encodes up to 16 streaming audio channels for Internet transmission to the public, or for internal distribution, via SHOUTcast, Steamcast or compatible stream replication server.
– Multiple-protocol dual (redundant) stream send and buffer-aligned receive paths with per-path VLAN tagging.
– UDP, TCP, and Multicast stream types, independently configurable per WAN stream.
– Complete WAN connection management – enable/disable, stream routing, origin and destination, audio and GPIO.
– 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, with audio encoding parameter control via IP.
– Increased GPIO capacity: 20 bi-directional closures per codec channel, and Virtual Endpoints with the same logic as attached to hardware circuits.
– Optional aptX® Enhanced audio coding may be ordered at time of purchase or added later, as required.
– When used as part of a Livewire network, allows audio from remote facilities to be used as if it were a local source.
– Axia GPIO xNode emulation logic with cut-in logic for snake between two hardware GPIO nodes, and cut-in logic for Axia Element to hardware GPIO link, plus Element console emulation logic.
– Remote status signaling and control using virtual GPIO pins.
– 3 bi-directional user data channels per codec.
– Two 5-input Virtual Mixer (VMIX) channels each allow combining 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.
– Time Zone Delay option enables delayed playout of select received audio channels for up to six hours.
– Remote control/configuration via any computer with a standard Web browser.
– Separate LAN and WAN jacks 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.
– SNMP monitoring with traps and attribute read supported.
– NTP synch­ronization for content delay on absolute time.
– Extended remote control and monitoring support via LWRP.
– Optional Time Zone Delay with SSD-based dynamic storage space allocation, configurable per codec, with synchronized delay of GPIO and user data channels