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Multi-Codec Gateway

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.

Additional information

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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




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STUDIO Telephone Interfaces Telos Zephyr iPort PLUS