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

IQOYA CALL for IP audio remote broadcasting

Key Workflow: Build reliable audio contribution links from remote talent to studio with uncompromised broadcast quality over unreliable Internet and Mobile networks

IQOYA CALL builds reliable, ultra-low-latency audio contribution links over unreliable Internet and 3G/4G networks. Specifically designed for non-technical users, it makes IP contribution as simple as placing a call from a smartphone. Its streaming engine FluidDIP enables both interoperability with third party IP codecs and optimized compensation for IP network impairments. Any administrator can configure IQOYA CALL via the cloud for in-the-field contribution scenarios, freeing remote talent from technical hassles.

At a glance

IQOYA*CALL 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 front panel color screen or from the Web page automatically provides an optimized connection without worrying about networks, protocols, 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 anticipate 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 and front panel interface
  • Quick profiles set-up for Audio and Network formats, with and without SIP
  • Direct IP connection to Axia Livewire and Livewire+, AES67 and RAVENNA consoles
  • Full ACIP (Tech 3326 and 3368) compliance with SIP signalling
  • Symmetric RTP mode with auto-answer for simple infrastructures without SIP
  • Ultra-low latency (e.g. typical 10 ms analog end-to-end, with EaptX)
  • Superior 24-bit audio quality (THD < -90 dB)
  • 2 Ethernet IP ports for WAN and LAN-audio connections
  • Superior AAC Fraunhofer error concealmentManaged from anywhere and at anytime by the IQOYA*CLOUD SaaS application


Audio performances

  • Sampling frequencies available:
    • analog I/O: 8 kHz up to 96 kHz
    • digital I/O (AES): 22 kHz 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 line (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 and EaptX)
  • 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

  • LCD + keyboard: 23 key keyboard + TFT color screen, 220×176 pixels
  • 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) and SIP (RFC3261)
  • Symmetric RTP mode with auto answer: ACIP RTP/UDP compliant. Automatic detection of audio format from IQOYA *MOBILE, IQOYA *CALL/LE and IQOYA V*MOTE
  • 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
  • Optional audio stream formats: Enhanced-aptX* Encode and Decode pack, 16/24 bits
  • Auxiliary data transport: in-band RS232 and GPIOs data tunneling
  • SIP included 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
  • 8GPI/8GPO 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

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



  • Simple to operate: smart profiles, multiprotocol management with and without SIP, and call processing integrated into the traditional contacts list
  • High quality of service IP contribution: through a large choice of robust, effective compression algorithms as tricky transport and smart recovery engines.
  • Interoperable: integrate seamlessly into any heterogeneous contribution network. Fully ACIP compliant, IQOYA*CALL also provides a connection to Livewire, 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


“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