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

Connectivity

• 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

Environment

• 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