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IQOYA TALK is a portable IP audio codec dedicated to live remote broadcasting for Radio and TV. Designed with an intuitive user-interface as simple as a smartphone, IQOYA TALK allows remote reporters to perform all the key actions in just 2 clicks. Live reporting or commentary can be performed, as well as studio quality interviews for up to 4 journalists and guests, with a user experience designed for the non technicals. Audio content is streamed through a large number of wired or wireless ‘last mile’ connections.

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  • Main Features:
  • Simple configuration by predefined scenarios in the studio
  • Large number of built-in connectivities: 2 Ethernet, WiFi, 2x 3G/LTE/4G
  • Up to 12H autonomy achieved by two independent batteries
  • Quick access to all relevant on-field settings
  • Specifications:
  • 3 Mic inputs, Microphone / Line level adjustment 12 or 48 V phantom power
  • 1 analog or AES/EBU stereo input Line level
  • 4 headphone outputs, individual mix bus on each output
  • 2 Talkback channels for users 1 and 2 and 1 mono or stereo program on independent IP connections
  • 11 input / 10 output embedded mixer
  • 2 Gigabit Ethernet ports for EBU/ACIP and AES67audio streams
  • 1 integrated dual band wifi / bluetooth module
  • 2 integrated 4G / LTE modules, approved in Europe, USA and Asia
  • 3 GPIOs controllable by GUI
  • 2 USB A ports for files 1 USB C port for streaming audio to PC
  • 2 independent SIP connections for 1 mono or stereo program and 2 talkbacks
  • Encoding : G711/G722, MPEG-1/2 Layer II MPEG-4 AAC-LC, AAC-LD/ELD, HE-AACv1/v2, Opus
  • Dual streaming / Configurable FEC
  • Redundancy of audio streams on available IP networks (Ethernet, WiFi, 4G / LTE, 3G)
  • Compliant with ACIP (EBU Tech 3326 and Tech 3368)
  • Lockable external power supply 12-24 V DC
  • LxWxH: 200 x 216 x 90 mm (77⁄8 x 81⁄2 x 335⁄ 64)
  • Up to 12H autonomy with 2 internal hot-swappable batteries (2 Li-ion packs, 6.4 A.h)
  • Weight : 2,3 Kg (5 lb) incl. 1 battery
  • Operating temperature 0-45 ° C
  • IQOYA X/LINK-LE benefits from all the major features of X/LINK but comes at a lower price point. It can be used in legacy analog or AES/EBU audio environments, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio. Like all the IQOYA products, X/LINK-LE is based on Fluid IP, the Digigram technology for reliable and resilient audio transmission over all types of IP networks including inexpensive unmanaged IP networks. Based on low consumption and fanless powerful hardware platform, IQOYA X/LINK-LE is designed for 24/7/365 use.
  • Main Features:
  • Cost-effective solution with essential features, and no compromises on reliability
  • Adapted to legacy audio infrastructures and full-IP audio infrastructures
  • 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 power supply units, 4 network ports with stream redundancy, audio failovers, audio hardware by-pass, and 1+1 hot device redundancy
  • Control and configuration via SNMP and Web services for easy integration with codec and network management/monitoring systems
  • Specifications:
  • I/Os and POWER:
  • 2 balanced analog audio I/Os with 24 bit converters, and 1 stereo AES/EBU
  • I/O with hardware sample rate converter, and 2 AES67 or Ravenna or Livewire I/Os
  • 4 ethernet ports: 1x100Mbps + 3x1Gbps on RJ-45 connectors
  • 1 RS232 port for serial data tunneling
  • 8 GPI / 8 GPO on Sub-D25 connector
  • 2 internal redundant PSUs 100-250VAC (Optional 100-250VAC / -48VDC) (Max 15W consumption)
  • RTP/UDP, Icecast/Shoutcast, HLS, SIP/SDP, STUN
  • PTP, NTP
  • DHCP, IGMPv2 and v3, QoS (VLAN tagging, DSCP)
  • Audio encoding formats: PCM linear 16/20/24 bits, ISO MPEG-½ La yer II and Layer III, MPEG-4 AAC-LC, AAC-LD, HE-AACv1, HE-AACv2, AAC-ELD, Opus
  • 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)
  • Multi-format encoding of the audio sources Network traffic separation (WAN, LAN, Management)
  • Configuration and monitoring via intuitive Web GUI and via SNMP
  • Optional: Simultaneous multi-protocol streaming
  • Optional: Audio output synchronization of decoders based on NTP or PTP
  • Available clock synchronization are internal, AES/EBU inputs, PTP, livewire clock
  • Hardware by-pass of the audio inputs to the audio outputs (analog and AES/EBU)
  • Unicast, multi-unicast, multicast, multi-multicast addressing
  • 3 decoding priorities per output program with choice of the audio source on each priority: IP stream, playlists on SDHC card, or audio inputs.
  • The switch between decoding priorities is automatic according to adjustable criteria
  • Silence detection on the audio inputs and on the received IP audio streams
  • Place calls from an address book or receive calls for live broadcasts (SIP, Direct SIP, Symmetric RTP)
  • Auxiliary data tunneling of physical or IP serial data and GPIs (including RDS-UECP)
  • Insertion of metadata to Icecast/Shoutcast streams (yellow pages and on-the-fly)
  • Optional: AES transparency without the need of an external GPS synchronized clock