Home Audio over IP IQOYA *CALL/LE for IP audio remote broadcasting
  • Overview
  • Specifications

IQOYA *CALL/LE for IP audio remote broadcasting

Key Workflow: Build cost-effective, reliable, broadcast quality audio contribution links from remote talent to studio over unreliable Internet and Mobile networks

For non-permanent installations, IQOYA *CALL/LE is all you need to send live, broadcast quality audio to the radio studio quickly and easily. Specifically designed for non-technical users, it makes IP contribution as simple as placing a call from a smartphone. Its streaming engine FluidIP enables interoperability and optimized compensation for unreliable networks. Any administrator can anticipate contribution scenarios via the cloud, freeing remote talent from technical hassles.

At a glance

IQOYA *CALL/LE provides a full-duplex stereo connection between a remote site and a studio

  • Ultra-simple to use: click to dial from contact list just like a smartphone. Dialling from the Web page automatically provides an optimized symmetrical RTP connection without worrying about infrastructure, networks, or audio technical parameters
  • Easy to manage: all setups are enclosed in the Administrator’s web page. Configure locally or via IQOYA*CLOUD app to manage in-the-field contribution scenarios
  • Smart quality administration: a smart bar graph displays connection quality in real time. Even inexperienced users can check and manage set-ups during actual field situations

Key features

  • Super-fast dial-up and configuration via web-based interface
  • Quick profiles set-up for Audio and Network formats
  • Direct IP connection to Axia Livewire and Livewire+, AES67 and RAVENNA consoles
  • ACIP (EBU Tech 3326) compliant in symmetric RTP mode and with optional SIP signalling
  • Symmetrical RTP mode with smart signalling and auto-answer for simple infrastructures without SIP
  • Ultra-low latency (e.g. less than 30 ms analog end-to-end, with Opus)
  • Superior 24-bit audio quality (THD < -90 dB)
  • 2 Ethernet IP ports for WAN and LAN-audio connections
  • Superior AAC Fraunhofer error concealment
  • Contacts and smart Profiles managed from anywhere and at anytime by the IQOYA *CLOUD SaaS application.
  • Optional SIP to access Active phone book and IQOYA *CLOUD Monitoring and Control functions


Audio performances

  • Sampling frequencies available:
    • analog I/O:8kHz up to 96 kHz
    • digital I/O (AES): 22kHz up to 216 kHz, with SRC on inputs
  • Frequency response at 48 kHz: 20 Hz-20 kHz +/-0,3 dB
  • IP audio inputs and outputs:
    • 1 native Axia Livewire(™) stereo channel, slaved to Livewire network clock
    • 1 AES67/RAVENNA stereo channel, slaved to PTP network clock
  • 1 AES/EBU (stereo) and 2 balanced analog line (mono) inputs
  • Maximum analog input level/impedance: +24 dBu / >10 kΩ
  • Analog input gain: 0 dB to +18 dB in 0.5 dB steps
  • Analog input THD + Noise at 1 kHz / –2 dBFS: < -90dB, Input gain 0 dB to +18 dB
  • 1 AES/EBU (stereo) and 2 servo-balanced analog (mono) outputs
  • Maximum analog output level / impedance: + 24 dBu / < 100 Ω
  • Analog output THD + Noise at 1 kHz / -1 dBFS: < -95 dB, load impedance > 5 kΩ

Professional quality streaming

  • Simultaneous encoding and decoding of different audio formats
  • Extremely low back-to-back latency (down to 10 ms in PCM)
  • Real time measurement of network path
  • Management of network conditions (jitter, lost packets, duplicated packets, disordered packets)
  • Selectable FECs as per RFC2733
  • Additional compensation of a lost packet
  • AAC error concealment implementation
  • A smart clock synchronization on incoming IP stream guaranteeing audio integrity

User interface and system management

  • Status: 6 status LEDs, 3 colors
  • Stereo headphones output: 6.35 mm jack, Max output level = 2×50 mW / 2×32 Ω load
  • PC interface: Web pages on local web server
  • SNMP v2 monitoring

Audio-over-IP formats and protocols

  • Audio networking format: Full compliance with ACIP (EBU Tech 3326)
  • Symmetric RTP mode with auto answer: ACIP RTP/UDP compliant. Automatic detection of audio format from IQOYA *MOBILE, IQOYA *CALL and IQOYA *CALL/LE
  • Included Audio stream formats: PCM linear 16/24 bits – ITU G.711/722 – ISO MPEG-1/2 Layer I, Layer II, Layer III – Fraunhofer Encode and Decode MPEG-4 pack (AAC-LC, AAC-LD, HE-AACv1*, HE-AACv2*, AAC-ELD)
  • Opus compression* speech and music
  • Auxiliary data transport: in-band RS232 and GPIOs data tunneling
  • Optional SIP to access Active phone book and IQOYA *CLOUD Monitoring and Control functions
  • Dual Mono mode*

Advanced Networking

  • 2 Ethercon IP ports for separate control traffic and IP audio streaming
  • VLAN tagging (IEEE 802.1Q, 802.1p) DSCP support


  • Analog and digital I/Os: Neutrik XLR-3
  • IP network: 2 Neutrik EtherCon RJ45 for ACIP and Livewire networks
  • 4GPI/4GPO on 2 Sub-D25
  • Auxiliary data: 1 Sub-D9 RS232 port
  • SDHC slot for memory card


  • Size: 19”/1U rackmount
  • Power consumption: 11W typical / 15W max
  • Power supply: 100 – 250 VAC, 50-60 Hz

The IQOYA range of IP audio devices and Visiblu-based radio automation systems developed by Digigram’s development partners offer a direct, low latency connectivity to Axia Livewire networks.

(*) With audio on AES/EBU or analog inputs/outputs


  • Simple to operate: smart signalling without SIP and call processing integrated into the traditional contacts list
  • High quality IP contribution: through a large choice of robust algorithms as tricky transport and recovery engines.
  • Interoperable: integrate seamlessly into any heterogeneous contribution network. ACIP compliant in Symmetrical RTP mode, IQOYA*CALL/LE also provides a connection to Livevwire, Livewire+, AES67, and RAVENNA consoles
  • Ultra-low latency: superior encoding algorithms allow for natural conversations that are more comfortable for talent and ultimately, more enjoyable for the audience
  • Cost-effective: without sacrificing contribution quality of service


“Everything was up and running quickly and perfectly from Chicago. The Digigram audio quality was excellent. We had the impression that the reporter was located in our studio in Brussels, rather than over eight-thousand kilometres away. We were amazed!”

Laurent Flemal, RTBF, Head of Digitization Division