Simply the most reliable, rock solid, guaranteed live audio over IP

... Will THE REAL STL-IP Please Stand Up

STL-IP for professional audio over IP networks and the Internet

IP codec live audio over networks Internet LAN WAN

 
  A u d i o T X  S T L - I P AudioTX STL-IP PLUS Plus  -  F e a t u r e s  &  S p e c i f i c a t i o n s . . .
STL-IP Plus for the best audio quality, most reliable, lowest delay audio over IP and Internet for STL, studio connections, remotes and outside broadcast
SYSTEM SUMMARY

AudioTX STL-IP Plus 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 Internet IP STL as reliable as it gets 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 SureFlowSureflow 5 with up to 5 truly independent redundant streams independently from transmission.
  • SureFlowSureflow 5 with up to 5 truly independent redundant streams 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 Internet IP STL as reliable as it gets 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 Internet IP STL as reliable as it gets, 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 Internet IP STL as reliable as it gets 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 Internet IP STL as reliable as it gets 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.

 

AUDIO SPECIFICATIONS AND PROTOCOLS
 

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:

Uncompressed audio at 8kHz to 96kHz sample rate, 16 or 24 bit. Mono or Stereo modes. Full-bandwidth linear audio with a 5ms delay.

FLAC
(lossless):

High-grade audio compression for near transparent professional audio, 16 to 96kHz sample rate, 16 to 24 bits, Mono/Stereo. Variable, depends on programme type, up to 60% reduction in bitrate compared with Linear PCM. 5ms delay.
J.41
(near lossless):
High-grade audio compression for near transparent professional audio, 32kHz sample rate, Mono/Stereo. Mono audio at 384kbps, Stereo at 768kbps. 5ms delay.
DAT12
(near lossless):
High-grade audio compression for near transparent professional audio, 16 to 48kHz sample rate, Mono/Stereo. Mono audio at 192kbps to 576kbps, Stereo at 384kbps to 1152kbps. 5ms delay.
     
MPEG4 AAC: Professional grade AAC coded audio at between 16 and 48 kHz sample rate, 16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. 24-320kbps. Provides full-bandwidth broadcast quality stereo audio at bitrates between 64kbps and 320kbps. Near transparent Stereo audio from 64kbps. 150ms delay.
MPEG4 AAC-Low Delay: Professional grade Low Delay AAC at 16 to 48 kHz sample rate, 16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. Low Delay version. 24-320kbps. Provides full-bandwidth broadcast quality stereo audio at bitrates between 80kbps and 320kbps with just 40ms delay.

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

Offering expectional audio quality at very low bitrates. HE-AAC coded audio at between 32 and 48 kHz sample rate, 16 bit, Mono, Stereo, and Parametric Stereo. 14 to 96 kbps. Provides full-bandwidth excellent quality stereo audio at bitrates between 14kbps and 96kbps. 260ms delay.
     
MPEG Layer 2 coded audio: Professional MPEG Layer 2 coded audio at between 16 and 48 kHz sample rate, 16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. Provides full-bandwidth broadcast quality stereo audio at bitrates between 128kbps and 384kbps. Mono audio from 64kbps. 45ms delay.
MPEG Layer 3 coded audio: Professional MPEG Layer 3 coded audio at between 16 and 48 kHz sample rate, 16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. Provides full-bandwidth broadcast quality stereo audio at bitrates between 128kbps and 384kbps. Mono audio from 64kbps. 125ms delay.
     
OPUS: Professional quality OPUS voice and music modes at between 8 and 48 kHz sample rate, 16 bit, Mono, Stereo modes. Provides full-bandwidth broadcast quality mono or stereo audio at bitrates between 32kbps and 510kbps. 20ms delay.
     
ADPCM: Professional quality compression, 32kHz or 48kHz sample rate, Mono or Stereo mode. Mono audio at 128kbps or 192kbps, Stereo at 256kbps or 384kbps. 5ms delay.
G.722: Good quality algorithm for speech/voice with a 7.5kHz audio bandwidth. Runs at 16kHz sample rate. Mono audio at 64kbps. 5ms delay.
LB-1: Extra Low-Bitrate speech codec offering 7.5kHz audio bandwidth. Runs at 16kHz or 24kHz sample rate and a range of bitrates determine quality. Mono audio from 12kbps. 40ms delay.
     
APTx: Enhanced APTx Coding - low delay, high quality compressed audio, choice of 16 or 24 bits. Bitrates range from 64kbps to 576kbps. 9ms delay.
Source audio selection:

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.

NETWORK CAPABILITIES AND SPECIFICATIONS AND OTHER FEATURES

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.

 

AUDIO, NETWORK & DATA CONNECTIONS
 

Analog audio inputs:

Stereo balanced inputs, 2x XLR (F) -18db nominal signal level. +18db at analog inputs = 0dbFS (digital full scale).
Analog audio outputs: Balanced Stereo 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)  
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 (4) and outputs (4) plus an additional 4 status output signals, D-Sub 25 pin connector. GPIO TTL inputs & outputs provide end-to-end transmission of signals from transmitting to receiving units.
Ancillary Data: RS-232 serial connection for ancillary data in and out, D-Sub 9 pin connector. Serial data can be transmitted/received alongside audio at up to 57600 bps.
Network Connection:

RJ45 Ethernet connector.

10/100 Ethernet connection for TX, RX audio and web-management interface.
POWER
 

AC Power:

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

 

 

... Will the REAL STL-IP Please Stand Up

Copyright (C) MDOUK MMXIV
unauthorised reproduction of contents prohibited
 
Contact us:    sales@stl-ip.com    tel. +44 (0)121 256 0200 (GMT)

 

quicklink STL-IP Plus live IP Codec for broadcast:   home   about STL-IP Plus   STL-IP Plus features/specifications   applications
SureFlow 5 about   SureFlow Compared   Hear the Difference, premium audio quality
quicklink STL-IP Classic:   STL-IP product info   STL-IP features/specifications   applications   sales info/contact
quicklink:   STL-IP-16 and STL-IP-8   models   features   examples

quicklink:   STL-IP Connect Software for reporters and reliable remote / outside broadcasts over Internet

quicklink:   Main AudioTX.com website for audio over IP solutions
quicklink:   The AudioTX STL-IP Blog for all things audio over IP