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Built on the proven, award-winning Intraplex® IP Link codec technology, GatesAir's Intraplex Ascent platform supports up to 32 audio channels with high-end features.

The scalability of Ascent makes it ideal for applications that require multiple channels of audio encoding and decoding at head-end sites. The high-density solution reduces cost and provides a path for convergence of IT and broadcast infrastructure. The interoperability between Ascent and IP Link codecs provides a complementary solution for remote contribution and distribution use cases.

Product Details

The Ascent platform continues the Intraplex tradition of providing an unprecedented level of reliability for audio over IP application. Built with enhanced network reliability and security in mind, Ascent supports the Dynamic Stream Splicing technology of IP Link codec, which provides “hitless” protection for packet losses using the combination of time and network diversity for packet transmission, and FEC.

To further enhance reliability and security, the platform supports the SRT protocol encapsulation. SRT (Secure Reliable Transport) is an open-source protocol that provides low-latency, reliable and secure streaming of audio and video data. The reliability in SRT is accomplished by a built-in packet re-transmission scheme; the security of streams is handled by the built-in AES-128/256 payload encryption.

In addition to “hitless” protection, like the IP Link codecs, the Ascent also supports automatic failover between different incoming streams or stored audio.

This combination of capabilities gives Intraplex codecs a market-leading position in STL over IP technology and unmatched network transport reliability.

Ascent supports both physical audio interfaces (AES3, Analog) and AES67 (AoIP), with a maximum of 16 full-duplex or a total of 32 channels (combination of encode and decode). Physical audio interfaces can be ordered as 4- or 8-channel cards, supporting both AES3 and analog audio signals.    The Ascent product is built on the Commercial-Off-The-Shelf (COTS) x86 architecture to leverage the scalability and cost of the technology. The product is available in a 1RU branded hardware server and as a software-only option.

Main Features of Intraplex® Ascent

• 4, 8, 16 stereo channels. Each channel can be full-duplex, encode-only or decode-only


• Supports AES3, Analog and AES67 audio input and outputs


• Standard Coding: Linear, Opus


• Optional: AAC-HE, AAC-HEv2, AAC- ELD, AAC-xHE, MPEG 2 and MPEG 3 audio coding


• Protocol Encapsulation: Real-time Transport Protocol (RTP), Secure Reliable Transport (SRT)


• Streams: Multicast, unicast, and multi-unicast


• 10 streams per channel with up to 20 destinations per transmit stream; maximum of 100 streams per system


• Three independent IP interfaces for redundant network operation


• Built-in silence sensor with optional stream switchover


• Automatic backup to audio playout from USB drive or external audio source


• Multicoding: can encode the same audio source in multiple formats


• Prioritized stream sources at the output with automatic switch over and switch back between primary and secondary streams and backup sources (streams, USB, external audio source)


• Programmable RTP level Forward Error Correction (FEC) scheme


• Programmable time diversity and Interleaving of streams to combat burst packet losses


• Integrated scheduler for automated scheduled program switching


• Integrated with Intraplex LiveLook (network analytics and monitoring software)


• Web and SNMP for management


• Multiple web account types to restrict access


• GPIOs: up to 32 in, 8 out. Available for stream transport and alarm assignment


• Network reliability
  
  
  
  • Dynamic Stream Splicing with network and time diversity for “hitless” packet loss protection
  
  
  • Programmable RTP level Forward Error Correction (FEC)
  
  
  • Secure Reliable Transport (SRT): new protocol provides automatic packet retransmission method
  
  
  
  


• IP Security
  
  
  
  • Access control per interface
  
  
  • Encryption of streams with AES-128
  
  
  
  

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MAGIC ACip3 is designed as 19” system with integrated wide area power supply and provides optionally an external redundant power supply. Three Ethernet interfaces are available which can be used for Audio over IP transmissions, to control the system with the Windows PC software or to integrate it into a network management system via SNMP. The Audio programmes can – flexibly and freely assignable – be fed in or given out, respectively, via an analogue and two digital stereo interfaces.

The system supports the G.711, G.722, ISO/MPEG Layer 2, Opus coding algorithms and PCM 16/20/24 Bit in the standard delivery version. Optionally, the Audio Codecs can be upgraded with Enhanced apt-X 16/24 Bit, AAC-LD/AAC-ELD and AAC-LC+V1/V2.
One stereo programme is encoded in the standard version, optionally the system can be upgraded to a second stereo programme by software activation.

Two operating modes are available for the pure IP version: the system can be used for dial-up AoIP connections or IP Leased Line connections. In AoIP mode, the system can register at 5 different SIP servers and automatically accept incoming calls from this SIP server. Audio connections in IP Leased Line and in AoIP dial-up Mode can be established with the Secure Streaming functionality for a highly reliable transmission.

Via the Ember+ protocol 64 inputs and 64 outputs can be programmed, an easy communication between MAGIC ACip3 and e.g. DHD or Lawo mixing consoles is possible.

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The AVN-AIO8 audio converter and interface converts up to eight analogue inputs and eight analogue outputs to and from the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue inputs and outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for POE, Link, and Clock.

• 8 x balanced analogue inputs on XLR.


• 8 x balanced analogue outputs on XLR.


• 1 x RJ45 Dante connector (1Gb/s Ethernet Port).


• POE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19” rack-mount form factor.

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Flexible in application: The codec is designed for studio-to-studio, studio-to-transmitter links and cross-media tasks. Even synchronization with microsecond accuracy is possible by an implemented latency control (useful for simultaneous signal output of all decoders of a network or for audio description). In addition, audio streams can be combined to multichannel streams. An optionally available FM Tuner offers further functions for fallback or backup and rebroadcasting. The Easy2connect feature provides operators as well as reporters with an easy to use connection establishment (SIP phonebook) while being in the field.

High compatibility: The IP-4c supports a wide range of protocols for streaming, control and monitoring (e.g. EBU TECH 3326, AES67, Ravenna, Livewire+, Dante, SMPTE ST 2110, PTPv2, RTSP, SAP, SIP, Discovery, Bonjour, SNMP, HTTP, HTTPS, FTP, FTPS or Ember+ and more). Furthermore, the exchange of additional information like GPIO and ancillary data between the audio networks is possible.

Pay as you grow: All soft- and hardware components are individually combinable. The scalability from one to four audio channels using software licenses gives you flexibility in planning your network and reducing your costs.

Multi-format audio coding: Another advantage is the variety of possible algorithms like MPEG1 Layer 2, MPEG2 Layer 3, most AAC profiles including the new xHE-AAC and AAC-ELDv2, OPUS, Ogg Vorbis, PCM, Enhanced aptX, Dolby Digital plus (on request) and more.

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The high quality and professional DSP-based hardware audio encoder MAGIC AE1 DAB+ Go is realized as a fanless ½ x 19‘‘ device and offers excellent audio quality thanks to the licenced Fraunhofer DAB+ algorithm.

The supplied Windows software allows the configuration of up to 40 systems.

Both analog and digital audio interfaces are available for the input of the audio signal.
In addition to the usual PAD feed via the multiplexer - which supports all standardized services - the PAD services Dynamic Label via FTP or UECP and Slideshow via FTP can also be fed in locally on the device.

In addition, a traffic announcement (TA) can be triggered very easily via UECP or a GPI contact.

Finally, the Audio Encoder allows direct transmission of the program type (PTy) like Rock, Pop etc. via UECP.

The transmission is monitored via the supplied Windows PC software or SNMP. Alternatively, an alarm can also be output via a GPO contact.
Up to three IP addresses can be assigned to the integrated network interface, so that a network separation for different applications is possible. The system also supports VLANs.

Communication between multiplexer and encoder is carried out via the AVTMUX protocol, which enables the control, monitoring and PAD transmission of the encoders from the multiplexer and guarantees secure transmission via Secure Streaming.
With this method, which has also proved its worth in the field of classical audio transmission, all IP packets are transmitted twice - with a delay. Due to the low bit rates with DAB+, the necessary doubling of the data rate is negligible. In addition, different routes can be implemented in the transmission path by using suitable addressing.

At the multiplexer (MAGIC DABMUX Go and MAGIC DABMUX Plus) all IP packets are reassembled correctly in time and duplicate received packets are rejected.
The system can also be used directly with the Open Source Multiplexer ODR DabMux. The necessary interface adaptations are available free of charge on the GitHub platform.

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The AVN-AO16 audio converter and interface converts up to sixteen analogue outputs from the Dante Audio-over-IP networking standard. This cost effective 1U rack-mount unit offers an easy solution for AV professionals and system integrators. It is simple to configure and operate, with all set-up done via the standard Dante Controller software and power via PoE (Power Over Ethernet).

All analogue outputs are on high-quality Neutrik XLR connectors and there are front panel status/confidence LEDs for POE, Link, and Clock.

• 16 x balanced analogue outputs on XLR.


• 1 x RJ45 Dante connector (1 Gb/s Ethernet Port).


• POE, Link, and Clock LED status indicators.


• Configuration using Dante Controller.


• AES67 operation & Dante Domain Manager compliant.


• Powered by PoE.


• 1U 19" rack-mount form factor.

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The reference audio codec

IQOYA X/LINK is a 1U rack IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and SSL links, but also DVB audio and WEB radio. It can be used in legacy analog or AES/EBU audio infrastructures, 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 is based on the Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable IP streaming over inexpensive IP links.  Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK is designed for 24/7/365 use.

Key points

Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure

Simultaneous delivery of your audio program to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats

Designed for audio service continuity and failsafe operation

Easy integration into existing SNMP based supervisors (SET, GET, Traps)

Ongoing product support with flexible options

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2wcom’s MoIN multi-format, multi-channel multimedia over IP Network Server (hardware, software or cloud): provides professional IP-based audio network technologies for real-time streaming, transcoding, protocol transforming, routing and mixing.

FLEXIBLE IN APPLICATION: designed for studio to studio and studio to transmitter links, as well as cross-media tasks. Multipurpose usage – e.g. audio routing, managing, levelling, loudness, monitoring and mixing between different protocols and environments. Audio streams are combinable to multichannel streams. PTPv2 provides accurate synchronization. The Easy2connect feature is a SIP phonebook that allows for an uncomplicated connection set-up. The number of channels can scale easiliy depending on your needs with permanent or temporary licenses.

HIGH COMPATIBILITY: MoIN supports all currently established standards and protocols like AES67, RAVENNA, Livewire+, Dante, SMPTE ST 2110, SRT, Ember+, SIP, SAP, PTPv2 and RTSP, as well as transcoding from one format into the other.

MULTI-FORMAT AUDIO CODING: The device provides a wide range of codec algorithms – e.g. MPEG Layer 2, MPEG Layer 3, most AAC profiles including the new xHEAAC and AAC-ELDv2, OPUS, Vorbis, Enhanced aptX, AC3 and more.

TRANSMISSION ROBUSTNESS: Dual Streaming and Pro-MPEG FEC ensure rock-solid IP transmission or go beyond with Stream4Sure. The Reliable User Datagram Protocol (RUDP) ensures highest packet recovery with minimum bandwidth and low latency.

SMART MANAGEMENT: configuration set-up via an easy to use web interface for general settings as well as for backup or fallback options. Remote control via HTTP/HTTPS, FTP, Telnet/SSH, NMS, SNMP, JSON, Ember+. Configuration for transcoding in 15 seconds. PTPv2 synchronization and latency control. Control via centralized Network Management System offering individual settings for e.g. adjustable silence detection, IP buffer and jitter check and PLL control. Sophisticated and elaborated alarm concept: forwarding of alarm messages (e.g. via SNMP, Ember+, HTTP/HTTPS…), optional source switching and event logging for documentation.

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MAGIC DABMUX Go RF is the smallest and only DSP-based Ensemble Multiplexers worldwide, thus providing high availability at an excellent performance. With MAGIC DABMUX Go RF RM there is also 19“ versions that can be mounted in a rack.

Up to 20 program providers can be connected via external Audio Encoders. An installation of the Encoders directly in the studio avoids effectively an interference in Audio quality because of Codec cascading.

Special value was set on the easy configuration of the Ensemble Multiplexer via web browser, so that even users without DAB expert knowledge are able to set up the system.

The highly compact DAB Ensemble Multiplexers facilitate a very simple Multiplex generation in accordance with standard ETSI EN 300 401. Despite its size, all features such as re-configuration (manually and scheduled), extraction of Sub Channels of other Multiplexers, integration of PAD and NPAD data services, creation of Service Information etc. are integrated.
Audio Services can be supplied via the AVTMUX or the EDI(ETI) protocol from external Multiplexers.
As output signal both Multiplexer variants supply an EDI signal for transmission to the transmitters.
With the RF version, which has an integrated modulator, you can alternatively activate a power amplifier directly. This possibility is particularly of interest if you have only one transmitter site.
The synchronisation is effected via NTP or in case of the RF version via the integrated GPS receiver.
The RF input is intended for future applications.

The configuration, operation and monitoring are effected via a HTML5-compatible web browser.
An external alarm can also be triggered via SNMP.
The system has a GBit Ethernet network interface, which allows the configuration of up to three IP addresses as well as VLANs.

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The AVN-CU2-DANTE is a portable commentator unit using Dante® AoIP.

The AVN-CU2-DANTE provides two mic/line inputs with a wide, adjustable gain range and has two stereo headphone outputs with lockable jack sockets, suitable for operation by two commentators.

It’s powered using Power over Ethernet (PoE), using Neutrik EtherCON connectors, with primary and secondary ports for power and data redundancy. There’s an additional 4 pin XLR 12V DC input. The unit supports up to 16 input and output AoIP channels and up to 16 simultaneous input and output AoIP streams.

The 6 x push-button rotary encoders and 12 x key-cap buttons are fully configurable, to control input & output levels and panning.

Each rotary encoder has a separate colour-coded meter section showing the channel name, detailed level metering, left/right panning and a limiter indication, on a bright daylight reading display. Colours can be programmed per encoder to quickly identify particular source groups, so headphone source selection becomes intuitive.

Metering is available per input/output, with output metering configurable as pre or post level adjustment. The top of the display shows output metering, a limiter indication and the name of the output, a pre-defined logo or nothing. A limiter is available on every output.

The unit has 2 x locking mic/line inputs with +48V phantom power indication and 2 x headphone outputs on locking 6.35mm jack sockets.

Four wire I/O on rear panel RJ45 connectors provide an AES3 or analogue input and output that can be assigned as mic outputs (line level), talkback outputs, programme inputs or talkback inputs as desired.  In addition, the AES/analogue connections can be used as an insert or exit point into/out from the AoIP network.

The unit has dual redundant network ports on both RJ45 (PoE using 2 x Neutrik EtherCON® connectors) and SFP cages.

There are 10 x configurable GPIO on a 15 way D-type connector with 1 x switched changeover output.

All of the buttons have key-cap text and can be configured. There are some standard operations available:

2 x On-Air buttons can be used to connect mic audio to the main output, either over AoIP or via the AES digital audio connection. The On-Air buttons can be locked if required.

A Menu button can be used to access limited setup options on the TFT display.

2 x Page buttons change the display and encoders to monitor an additional set of sources, mix points or outputs. Up to 4 pages can be pre-programmed, e.g. one for talkback inputs, one for outputs, one to monitor other sources.

2 x Cough buttons take the commentator off-air while pressed.

A User button can be programmed to perform any function using the web server.

4 x T/B (talkback) buttons can be configured to initiate talkback over AoIP or AES digital audio connection, using 4 x talkback busses. The talkback buttons operate with lazy talkback, taking the commentator off-air when invoked.

The illuminated ‘Sonifex’ logo acts as a power indication and there are illuminated LEDs to indicate network clock status, AoIP Primary and AoIP Secondary link status, PoE Primary, PoE Secondary and DC power active.

Stream setup to and from the unit is initially via Dante® Controller with more detailed configuration performed by using the built-in web GUI.

-end-

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The AVN-Portal series is a group of three powerful audio mix engine interfaces (AVN-PA8, AVN-PD8 & AVN-PM8) capable of audio routing, metering and equalisation. They have both physical audio I/O and AES67 stream I/O and can mix and route freely from any input to any output, useful for a multitude of applications.

Communication between products is via RAVENNA/AES67 AoIP allowing simple CAT 5 cabling and expansion. They advertise streams using Avahi/Bonjour and SAP so can be used for Dante™ AES67 enabled streams too.

Applications:

• 8 output analogue zone mixer, with individual output gain control.


• 8 channel clean-feed generator, with input mixing and gain control on inputs and outputs.


• Distribute 8 stereo channels of audio over an SFP fibre connection.


• IFB generator to send 64 x AES67 streams to individual belt-packs.


• 8 output headphone distribution system, with separate input mix for each headphone output and individual gain control.


• Input mixer with input/output metering and AES67 stream generation.

Webserver Software

A built-in responsive web server provides complete remote configuration & control of the unit including matrix mixing and routing, and also allows for firmware updates and configuration backup. Complete product configurations can be saved and loaded for use in different situations and system logs can be saved for device information.

Mix Matrix

The key to the success of the AVN-PA8 is the mix matrix where physical inputs can be freely mixed and routed with AES67 streams, in a simple and intuitive way to both physical outputs and AES67 streams.

The unit can stream RAVENNA & AES67 AoIP streams or AES67-enabled Dante® flows (discovered using SAP). It can receive AoIP streams from 16 additional AES67 sources and can send to 64 additional AoIP destinations.

Input and output AES67 streams can be individually added/modified and the SDP of each stream can be checked and edited.

DSP functions, such as gain and filtering, can be added at inputs, outputs and cross-points.

The unit can act as a PTP masterclock or slave clock and supports IEEE1588-2008 PTPv2 media and default profiles.

Front Panel Displays, Metering & Controls

The AVN-PA8 can be supplied with different front and rear panels. As standard it has a front panel display to show product information and it uses XLRs and D-types for rear panel connectivity.

Using an OLED display, the front panel provides detailed status information on device name, network addresses, PTP clocking info, power status/voltages and version information. The display and navigation controls allow editing of certain functions, limited to networking (IP addresses, friendly name, etc) and display (brightness and contrast). The front panel controls also include user configurable buttons which can be set-up to perform actions such as activating a GPIO or as a shortcut button to jump to a specified menu on the OLED display.

Front panel LEDs show the AoIP network status, synchronisation status and the status of the AC and DC power supply inputs. The brightness of the OLED display and LED indicators can be continuously adjusted for low or high lighting conditions.

A front panel power button is available to turn the unit on and off. The power button is disabled by default but can be enabled through the ‘Display Settings’ web page.

Detailed Metering Option

The ‘D’ version of the portal (e.g. AVN-PA8D) has two bright TFT meter displays which provide a live display of the levels of the physical inputs and outputs respectively. A rotary navigation control can be used to select a single input or output and view its metering data in a more detailed horizontal view.

The metering scale used is user configurable to one of 9 different metering scales, with relevant ballistics. The metering scales available are: Dual PPM + Standard VU, EBU PPM, BBC PPM, Nordic PPM, AES Digital PPM, DIN PPM, German PPM, SMPTE RP.0155, Standard VU & Extended VU.

Metering can be set to be either ‘Discrete’ or ‘Continuous’, which changes the appearance of the meter bar.

Phase metering can be displayed per stereo channel and channel idents can be shown either above or below the metering to identify each input/output.

On devices without a meter display, a smaller set of monochrome meters are shown on the main OLED display.

Physical Inputs & Outputs

For the analogue audio, the AVN-PA8 uses D-type sockets with AES59 analogue pinout, paralleled with 8 x RJ45 connectors using StudioHub® pinout.

The ‘T’ version (e.g. AVN-PA8T) uses rear panel terminal block connectors for all physical inputs and outputs.

The rear panel contains IEC mains and secondary DC power inputs which provide power redundancy to the product.

There are two Ethernet RJ45 connections (control and AoIP) and there is an Ethernet SFP module that, when used, replaces the AoIP RJ45 connection, e.g. for a 1Gbit/s copper or optical SFP transceiver. When an SFP is used, this replaces the AoIP RJ45 connection.

A rear panel GPIO connector provides 10 local ports which can be user configured as inputs or outputs and provide software-controlled functionality. A voltage free relay contact can be used to operate external equipment. There are virtual GPIO ports which can be used to trigger events over the network between devices.

For remote operation and monitoring, SNMP V2 is supported and the units can be controlled using Ember+ commands.

8 Way Analogue Headphone Distribution System

The AVN-PA8 can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers.

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Designed for FULL IP environments

IQOYA X/LINK-AES67 is a 1U rack IP audio codec designed for the delivery of  stereo and/or mono audio sources over IP networks, for STL, SSL, DVB audio, and WEB radio applications. It is dedicated for full-IP audio infrastructures based on  AES67,  Ravenna, or Livewire technologies.  Like all the IQOYA products, X/LINK-AES67 is based on the Fluid IP technology which offers redundant dual streaming, allowing  reliable IP streaming over inexpensive IP links. Based on a low consumption, fanless and  powerful hardware platform, IQOYA X/LINK-AES67 is designed for 24/7/365 use.

Key points

Designed for full IP environments

Supports AES67, RAVENNA, and Livewire technologies

Scalable number of supported IP audio input and ouput channels (from 1 to 8 stereo I/Os)

Simultaneous delivery of input audio programs to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios, in multiple audio formats

Ongoing product support with flexible options