Digigram IQOYA SERV/LINK 88 Dante

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IQOYA SERV/LINK – Multichannel IP Audio Codec (MADI / AES67 / AES/EBU / DANTE / Analog)
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IQOYA SERV/LINK is a high-density 1U rack multi-channel IP audio codec designed for live remote broadcasting and delivery of audio programs in applications such as multiple STL and SSL links, delivery of multiple WEB radios to CDNs, multiple program delivery to DVB/cable operators, multiple live remote broadcasts, multiple intercom and commentary channels, and IP audio transcoding.


IQOYA SERV/LINK handles from 4 up to 64 stereo (8 up to 128 mono) audio input and output channels in 1U, with the possibility to simultaneously encode and stream the audio inputs in multiple formats and protocols, decode IP audio streams to the outputs, and transcode IP audio streams. It features two hot-swappable redundant power supply units, two Ethernet ports, and supports different types of audio I/Os (Analog, or AES/EBU, or MADI, or AES67/RAVENNA, or DANTE, or AES67/RAVENNA+MADI). IQOYA SERV/LINK can be fully controlled and monitored from its embedded WEB pages and through SNMP or web services.


Designed for: Radio Broadcasting (SSL, STL, WEB radio streaming, DVB audio, Live remote) and TV/Video Broadcasting (multiple intercom and commentary)

Scalable number of supported audio I/Os (for MADI, AES67/RAVENNA and DANTE versions) and of transcoding through license upgrade

Simultaneous encoding and delivery of multiple audio programs to transmitter sites, WEB radio CDNs, DVB multiplexers, or studios

Up to 3 decoding priorities per output (with automatic switching between priorities in case of audio source loss). Sources can be IP audio streams, local backup playlists and sound files, audio inputs

MPX transport on AES192 (up to 16)

Intuitive web interface to place calls from an address book or receive calls during multiple live broadcasts (SIP, Direct SIP, Symmetric RTP)

Control, configuration and alarming via SNMP and Web services for easy integration with codec and network management/monitoring systems.

Tunnelling of physical or IP serial data and GPIs (including yellow pages and on-the-fly Icecast/Shoutcast metadata, RDS-UECP metadata)

Key points

High audio channels density in a 1U rack.

Simultaneous encoding and delivery of multiple audio programs to transmitter sites, WEB radio CDNs, DVB multiplexers, or studios

EBU/ACIP compliance for interoperability with third-party codecs and any SIP infrastructure.

Multiple levels of redundancy for audio service continuity and failsafe operation: 2 hot-swappable power supply units, 2 Gbit network ports with stream redundancy, audio failovers, and 1+1 hot device redundancy.

Features Fluid-IP, the Digigram technology for resilient audio transmission over unmanaged IP networks.

Isolation between the internet world and the LAN audio (AES67 or DANTE) to preserve the security of the internal IP-audio infrastructure.


Dimensions: 19, 1RU, 43.9 cm depth B

Weight: 7.9 kg

Power supply: 2 hot swappable redundant PSU 90-264VAC, Optionally, 2 hot swappable redundant PSU -48VDC

Temperature: Operating 0C  50C, Storage -20C TO 70C

Humidity: 85% non-condensing

Power consumption: Max 80W


Ethernet ports: 2 x 10/100/1000 Mbps RJ-45

Analog and AES/EBU versions: XLR connectors on breakout cables

MADI versions: Multimode optical fiber

AES67/RAVENNA and DANTE versions: additional Gbits/s RJ-45 ports (2 ports for 64 channels, 4 ports for 128 channels)

AES/EBU Sync: Female XLR (analog and AES/EBU)

Word Clock Sync: BNC 75 Ohms (all versions)

Optional RS232 ports: 8 or 16 Sub-D 9 pins on 1 or 2  breakout cables

Optional GPIOs: 16 GPIs TTL compatible on terminal block, 16 relay GPOs on terminal block


Type: Balanced

A/D converter resolution: 24 bits

Maximum input level/ impedance: +24 dBu/ >10 k

Adjustable input gain: from 94.5dB +24 dB


Type: Balanced

D/A converter resolution: 24 bits

Maximum input level/ impedance: +24 dBu/ <100

Adjustable output gain: from 94.5dB +24 dB


Hardware sample rate converters: Sample rate conversion = 7.5:1 to 1:8, up to 192 kHz

Programmable input gain: from 15 dB to +15 dB


Sample rate: 32 kHz, 44.1 kHz, or 48 kHz


AES11: Only on analog and AES/EBU versions, and AES/EBU versions.

Word Clock: 32, 44.1 kHz 48 kHz


Sample Rate: 32, 44.1 kHz 48 kHz

MADI OUT channel mode: 64/64 channels mode


Dynamic range (A-weighted): Analog In: >104 dB / Analog Out: >106 dB

THD + noise 1 kHz at 1 dBfs: Analog In: <97 dB / Analog Out: <96 dB

Channel phase difference: 20/20kHz: 0.2 / 2

Crosstalk (Analog in or out) 1 kHz at 22 dBu: <100 dB 15 kHz at 22 dBu: <85 dB


Streaming and signalling protocols:  RTP and UDP with or without MPEG-TS encapsulation (SPTS and MPTS), Icecast/Shoutcast, HLS, SIP/SDP, STUN

Support of unicast, multi-unicast, multicast, multi-multicast addressing (IGMPv2 and v3)

Support of VLAN tagging and DiffServ QoS (DSCP)

Time synchronization protocols: PTP, NTP



Supported formats: MPEG L2 & L3, Opus, Fraunhofer MPEG4-AAC (AAC-LC, AAC-LD, HE-AACv1, HE-AACv2, AAC-ELD), PCM, G711, G722, and optional Qualcomm aptX

Support of multichannel audio 5.1 and 7.1

Selectable FECs for ACIP/RTP streams (from +10% to +100% IP bandwidth), Pro-MPEG CoP #3 FEC for MPEG-TS streams

Dual-port redundant streaming with spatial and time diversity

Adaptive and resilient audio streaming (Fluid-IP)

Full EBU/ACIP compliance (tech3326 and 3368)

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