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    Digigram IQOYA LINK

    IP STL Codec
    Digigram-IQOYA-LINK

    TRU Price Guarantee

    (Subject to conditions)

    2,185.00
    2,185.00
    255.00
    255.00
    386.00
    386.00
    1,571.00
    1,571.00
    255.00
    255.00
    386.00
    386.00
    0 .00
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    Tech Support
    2-Year Warranty

    Description

    IQOYA *LINK is a stereo IP audio codec specially designed for STL and SSL applications based on IP networks. It has been developed in close collaboration with leading European Telcos around the following themes: full IP connectivity, reliability, IP streaming robustness on managed and unmanaged networks, continuity of audio service on transmitter sites, full remote control and monitoring from SNMP supervisors.

    At a glance

    IQOYA *LINK is based on a low consumption, fanless, and robust dedicated hardware platform. It includes a rich set of features allowing for the reliable transport of audio content over managed and unmanaged networks (FECs, redundant dual streaming with time diversity, VLANs, QoS, IGMP), and for continuity of audio service on transmitter sites (3 backup levels, backup on SD card, switchable hardware by-pass of audio inputs to outputs in case of power supply failure). It can be used in analog and AES/EBU environments, as well as Livewire, RAVENNA and AES67 environments. In addition, IQOYA *LINK can be fully  controlled and monitored through SNMP.

    • Designed for 24/7 operations
    • Supports legacy audio connectivity (analog, AES/EBU), as well as AoIP (Livewire, AES67, and Ravenna)
    • Support of analog MPX(*), and digital MPX (MPX over AES192)
    • Reliable IP streaming: FECs, redundant dual streaming with time diversity, management of network errors, real time network metrics
    • Continuity of audio service on transmitter site (3 backup levels, hardware by-pass of audio inputs to outputs in case of power supply failure)
    • Wide range of supported audio formats
    • Full remote control, monitoring, and alarming through SNMP and from WEB GUI
      (*) For analog MPX, it is necessary to use third party 75 Ohms BNC<->XLR adapters.
    • Full-duplex stereo IP audio codec
    • Low consumption and fanless platform adapted to transmitter sites (strong resilience to EMI)
    • Two 10/100 Mbits/s Ethernet ports for separate control traffic and IP audio streaming, or for seamless stream failover (dual redundant streaming), or for a fail-over on a secondary network
    • Balanced analog and AES/EBU inputs/outputs
    • Support for Livewire (standard mode), AES67(*), RAVENNA(*) inputs and outputs (*:excluding simultaneous encoding/decoding in AAC, aptX, or Opus, with AES67/RAVENNA I/Os)
    • Audio formats: PCM, G711, G722, Opus, MPEG L2 & L3, Fraünhofer AAC-LC, AAC-LD, AAC_ELD, HE-AACv1, HE-AACv2, and optional Qualcom aptX™
    • Audio format for optional analog MPX transport: PCM 12, 16, 24 bits. Sampling frequency: 144 kHz, 192 kHz
    • Audio format for optional MPX over AES192 transport: PCM 12, 16, 24 bits. Sampling frequency: 192 kHz
    • Streaming protocols: RTP, Icecast/Shoutcast, MPEG-TS/IP SPTS on RTP or UDP
    • VLAN tagging (IEEE 802.1q, 802.1p), DSCP, SNMP (SET,GET, Traps), supported by Dataminer
    • Unicast, multi-unicast, multi-multicast, IGMPv3
    • FECs and redundant dual streaming with time diversity up to 2 seconds
    • In-band auxiliary data tunneling (serial data, 8 GPIs)
    • Continuity of audio service on transmitter sites:
      – Backup: IP stream (RTP, Icecast/Shoutcast),  playlist/ sound files on local SD card, audio inputs
      – 3 backup levels
      – Switchable hardware by-pass of inputs to outputs in case of power supply failure
    • Real time metrics on received RTP stream
    • Optional synchronization of decoders for multi-frequency networks (MFN)
    • AES/EBU transparency (audio samples + C & U bits) without external synchronization.
    • Intuitive web interface with access rights
    • Front panel LCD display, keypad

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