AudioTX STL-IP Plus   IP codec provides for live audio transmission over IP networks with transmission grade audio quality & robustness and extremely low delays - as low as 5ms!

• IP Codec for live audio: Transmit and receive audio using point-to-point UDP or TCP/IP, and point-to-multipoint UDP Multicast network protocols over ANY IP Network - including private networks (LAN/WAN, MPLS), Point to point, Satellite, Wireless networks, T1/E1, ATM or the Internet.
• A single STL-IP Plus  system can encode your audio separately using up to 6 different audio coding types and bitrates and can transmit audio on up to 66 simultaneous connections using different network protocols if required - TCP/IP or UDP. Using Multicast, audio can be sent to an unlimited number of destination units. Up to 66 different multicasts can be generated each with unlimited receiving units. Audio receiver can receive from up to 5 remote units, each with different algorithms/bitrates for multiple levels of fallback and/or use SureFlow independently from transmission.
• SureFlow allows you to combine multiple different networks, routes or connections to maximise reliability. Up to 5 independent, redundant streams can be sent on one or multiple different networks. Each of the 5 streams is independently encoded and so can use different bitrates and audio coding types and each is independently decoded at the receiving end allowing STL-IP Plus  to choose the best available audio for every single sample before you hear it.
• Exclusive Virtual Networks system allows you to set up mutiple IP addresses in STL-IP Plus , each with its own subnet, DNS and gateway/routing, allowing you to send different streams via different networks and/or separate control and audio data.
• AudioTX STL–IP Plus  works with linear (uncompressed) audio at up to 24 bits and 96kHz sample rate, lossless compressed audio with up to 60% saving in bitrate, compressed audio via built-in Professional Grade MPEG Layer 2 or MPEG Layer 3 coding/decoding, near lossless J.41 and DAT12, pro-grade OPUS encoding using the newest versions, ADPCM, G.722 or our extra Low-Bitrate speech codec. Plus MPEG4 AAC, AAC Low-Delay and HE-AAC v2 with the optional AAC Coding Pack for Stereo audio from just 14kbps! and optional APTx Coding (Enhanced APTx, 16 and 24 bits) with the APTx Codec Pack.
• Two types of switchable Forward Error Correction (FEC), separately switchable for every individual stream even with SureFlow/5.
• Up to 60 seconds of network jitter compensation (buffer) where required (per independent stream) configurable in 1ms increments.
• Synchronous transmission of serial ancillary data and/or contact closures (TTL GPIO).
• 256 bit AES encryption (government standard) for your audio - to keep it private but also to guarantee the genuine source of your audio at point of reception.
• Built-in silence and audio overload detectors.
• Backup audio can be received from alternate STL-IP sending units (up to 5), sourced from audio files stored within the unit (which can be remotely updated of course), from an internet web stream, or even sourced from the local device analogue or digital input where you have other backup audio sources onsite.
• Monitor and control via web-browser control interface which works on any device, SNMP traps and queries, E-mail alerts, , included software and logic level status outputs.
• Telnet style IP remote control interface (using simple text commands and responses) easily integrates with existing automated control, management and scheduling systems.
• Incredibly flexible and cost-effective solution.

AudioTX STL-IP  can encode your live audio using up to 6 different audio encoding types and bitrates. And can send up to 66 copies using any combination of UDP, TCP/IP or UDP Multicast (to an unlimited number of destinations for each UDP Multicast connection). The IP Codec system can receive from up to 5 remote units, each with different algorithms/bitrates for multiple levels of fallback and/or use SureFlow, in both cases independently from transmission.

Summary:

Professional grade analogue balanced Stereo audio inputs and outputs plus AES/EBU digital audio in/out, external wordclock input. Audio in/out at up to 24 bit, 96 kHz sample rate.

Encode your audio up to 6 ways with 6 different audio encoding types and bitrates. Send it to up to 66 remote units or an unlimited number using multicast.

Mono/Stereo audio transmit/receive using Linear (uncompressed) audio, Lossless FLAC, near lossless J.41 and DAT12, pro-grade OPUS codec, Broadcast Quality MPEG Layer 2, MPEG Layer 3, Mono, Stereo, Joint-Stereo, Dual-Mono operation.
MPEG4 AAC, AAC Low-Delay and HE-AAC v2 with the optional AAC Coding Pack.
Enhanced APTx coding with the APTx Codec Pack.

Detailed specification:

Linear PCM
(uncompressed) audio:

FLAC
(lossless):

MPEG4 HE-AAC and HE-AAC v2:
(High Efficiency AAC, AACPlus):

User-selectable channel inputs to audio transmission modules - Left channel, Right Channel, Stereo or MonoMix (L+R).

** AAC Codec Pack required for MPEG4 AAC types, APTx Codec Pack for Enhanced APTx coding.

Supports all IP networks including Telco, MPLS, Private/Dedicated circuits, LAN/WAN, Satellite, Wireless (incl. WiFi), ATM, T1/E1 and The Internet for IP codec operation.

Network modes: UDP, TCP/IP, UDP Multicast modes

Audio transmit/receive bitrates between 24 kB/s and 4.6 mB/s

Optional transmission of ancillary serial data at up to 57600 bps, up to 4 in / 4 out GPIO (contact closures)

Optional use of 2 types of FEC (forward error correction) and/or network jitter compensation/safety buffer configurable in 1ms increments from zero to 60 seconds

Encode your audio up to 6 ways with 6 different audio encoding types and bitrates. Send it to up to 66 remote units or an unlimited number using multicast.

Monitoring and control via:

• Web-browser control interface.
• SNMP traps and queries.
• E-mail alerts.
• Telnet style IP remote control interface (using simple text commands and responses).
• Included software.
• Logic level (TTL) status outputs.

Built-in silence and audio overload detectors.

SureFlow for up to 5 independent, redundant multi-streams, sent using one or more available networks/connections.

• Up to 5 redundant streams, encoded using any of the audio coding types and at any bitrate.
• Streams can be sent using the same or different networks/connections.
• All streams are independently decoded at the receiver and each individual sample of audio is chosen from the best quality available according to your stream priorities.
• Individual streams can use switchable FEC (forward error correction), encryption.
• Network jitter compensation/safety buffer configurable in 1ms increments from zero to 60 seconds.

Virtual Networks:

• Give STL-IP up to 6 different IP addresses each with their own subnet, DNS, gateways.
• Can be used to send audio over different networks/connections/routes with or without SureFlow
• Can be used to separate audio and control data.

Backup audio options - in addition to SureFlow.

• Receive audio from up to 5 different sending units for multiple levels of fallback.
• Switch to audio playback from files stored in the unit (files can be updated remotely).
• Choose to receive an internet webstream as a backup source.
• Use live audio from the local analogue or digital inputs as backup audio.

256 bit audio encryption:

• Government grade 256bit audio encryption using your own passphrases or direct key entry.
• Guarantees the source of your audio so you know it's 'you' at the other end - essential for internet connections.
• Secures your audio in confidential applications.

Custom/project based options.

• MPEG TS output (MPEG SPTS, Single Programme Transport Stream).
• MPEG Raw encoding output.
• IEEE 802.1Q VLAN support.
• Others on request.

Analog audio inputs:

RJ45 Ethernet connector.

AC Power:

96-264 VAC, 50-60Hz autosensing for worldwide operation.
DC Power option available.