TBU-IP Studio TBU / Telephone hybrid for VoIP and SIP phone systems and PBXs

AudioTX TBU-IP - Digital Telephone hybrid / TBU for SIP / VoIP phone systems and PBXs

Features and Specifications


AudioTX TBU-IP is a Dual VoIP/SIP telephone hybrid/TBU. Designed to operate with SIP capable PBX systems or as (two) standalone SIP phones directly with trunks or external telephony providers. Front panel LEDs show the status of each TBU module. External status/signalling outputs and control inputs via TTL GPIO port.

  • Designed for Professional Broadcast applications - Radio and TV studios, control rooms, OB vans. And any other application where you need professional quality audio to and from telephone callers.
  • Can be used with an existing PBX (where each of the 2 TBUs acts as a normal SIP phone extension) allowing operator screening of calls before transfer to the studio. After the on-air call is complete, caller can be reverted back to a pre-programmed extension or group on the PBX or disconnected.
  • Can be used directly, without a PBX, as simply as any SIP phone.
  • GPIO / 5v TTL outputs show Ring, In use, Line Available.
  • GPIO / 5v TTL inputs allow for Manual Answer, Hangup/Revert and dialling of pre-set numbers.
  • Supports standard digital phone quality (G.711) and wideband calls (G.722).

Professional grade analogue balanced mono audio inputs and outputs for each TBU module (2) plus single AES/EBU digital audio in/out (TBU1 on Left channel, TBU2 on Right channel), external wordclock input. Audio in/out at up to 24 bit, 96 kHz sample rate.


Each TBU module appears as a separate SIP phone/extension to a PBX or trunk provider. G.711 and G.722 calls supported for standard (3.5 kHz) and wideband (7 kHz) audio quality.

Detailed Specification:  
Digital telephony audio, 3.5 kHz audio bandwidth, for standard high quality telephone calls.


Wideband telephony audio, 7 kHz audio bandwidth, for enhanced, high quality telephone calls.

Supports all IP networks including Telco, Private/Dedicated circuits, LAN/WAN, Satellite, Wireless (incl. WiFi & WiMAX), ATM, T1/E1 and The Internet via inbuilt 10/100 Ethernet interface.

Audio transmit/receive bitrate:  64 kB/s per channel/call

Optional use of network jitter compensation/safety buffer configurable in 1ms increments


Monitoring and control via:

  • Web-browser control interface
  • Telnet style IP remote control interface (using simple text commands and responses)
  • Logic level (5v TTL) inputs and outputs for connection to studio signalling systems and external operator control panels/buttons.

Analog audio inputs:

Balanced inputs, 2x XLR (F) -18db nominal signal level. +18db at analog inputs = 0dbFS (digital full scale).
Analog audio outputs: Balanced outputs, 2x XLR (M) -18db nominal signal level. 0dbFS (digital full scale) = +18db at analog inputs.
Digital audio input: AES/EBU digital input, XLR (F) Input accepts both AES/EBU and SPDIF type of signals.
Digital audio output: AES/EBU digital output, XLR (M) TBU1 on left AES channel, TBU2 on right.
Clock input: Wordclock input, BNC. System clock-source is user-selectable: internal clock, wordclock input or use clock from incoming AES/EBU source.
GPIO: TTL level inputs and outputs on D-Sub 25 pin connector. GPIO TTL inputs & outputs for interfacing with external studio signalling and control panels/buttons.
Network Connection:

Neutrik Ethercon RJ45 connector, accepts standard RJ45 connector or locking Ethercon cable.


10/100 Ethernet connection for IP telephony connection and web-management interface.

AC Power:

96-264 VAC, 50-60Hz autosensing for worldwide operation.



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Contact us:    sales@audiotx.com    tel. +44 (0)121 256 0200 (GMT)


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