Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation WAN transport-platform designed for all-IP services and all-IP networks, guaranteeing quality for the most demanding live media events while providing agility through automated real-time provisioning. Scaling both switching and link capacities to levels unprecedented in the media WAN space, enabling event sites, stadiums, studios, and production sites to be interconnected using a total of 1,2 Tbps of guaranteed capacity in just a compact 3RU modular form factor.Future proof building practice with fully flexible cooling and power configurations allow deployment in any modern service provider, broadcaster or data center environment. Designed with a virtualization architecture, using one common hardware platform for all media and network functions, enabling flexible deployment of functions anywhere in the network. Modular and redundant, with hot swappable and protected components guarantees uptime and reliability.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Combining native video, audio and high quality ethernet transport with switching, the Nimbra 300 provides a high-quality multiservice solution for demanding media applications. Multi-service capabilities allow you to transport live and file-based content across the same network, and with native support for DTT (ATSC 3.0, DVB-T2, ISDB-T) transport, Broadcast Radio distribution and video contribution the Nimbra 300 series gives you all the flexibility you need.

With support for service admission control and guaranteed bandwidth, performance and characteristics different services can safely be mixed no matter the network load. With hitless 1+1 protection of media and data services you can be sure that the expected service quality will always be delivered.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

With its swiss-army-knife approach, the Nimbra 400 family is the very definition of versatility. Engineered for the harsh realities of the internet, the Nimbra 400 family supports a wide range of video and transport technologies. The combination of native support for both AVC and HEVC encoding/decoding, compressed video over IP and ASI and an open and flexible approach to IP networking in a super-efficient 1 RU form factor makes the Nimbra 400 family a perfect companion for space constrained remote sites and flight kits. With native support for RIST, SRT, Zixi and our own Edge Connect software means you can get your content anywhere. The integrated encryption and firewall paired with transport of any remote management traffic and comms also minimizes the need for third party equipment on site.

Paired with its standalone software counterpart, the Virtual Nimbra, means there are solutions ranging from highly redundant appliances for flight-kit and racks, to flexible software for datacenters and public cloud providers.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » Hybrid SDI-IP video routers »

Designed from the ground up for high-quality media transport the Nimbra 600 is a highly reliable video and audio processing, monitoring, and IP transmission platform. Support for any underlying network, including IP, wavelength systems and dark fiber, means you are never limited by existing infrastructure.

Available in ultra-compact and high density chassis options, the modern modular design leverages multi-purpose media acceleration, media processing modules and IP networking modules to allow each node to be tailored to individual deployment needs. Redundant switches, redundant node controllers and redundant power supplies ensure high availability.

Configure the Nimbra 600 either as a compact multi-function media access solution or as a powerful edge processing and acceleration access to complement the Nimbra 1000.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation of media transport designed for ultimate reliability, flexibility and versatility. Nimbra Edge is a fully cloud-based media transport solution designed for private or public cloud environments.Nimbra Edge augments existing Nimbra devices by drastically simplifying the network architecture through offloading complexity such as redundancy and scalability to its cloud-based backend. With its open approach to networking, Nimbra Edge connects to any media device supporting RIST, SRT, Zixi or Net Insights Edge connect software. The latter designed to cloud-enable any third-party device or workflow.

Service centric orchestration and network management software that automates network operations and control as well as visualizes network performance through dashboards and reports.

Through pre-defined templates for automatic provisioning, automated inventory and both per-service and aggregated service monitoring, Nimbra vision ensures efficiency and control. Nimbra Vision helps you to plan, upgrade, and evaluate your network capacity and performance in order to deliver world class video services over any infrastructure. With a high availability architecture, all your network management capabilities are available at all times.

Connect » Real-Time Signal Transfer » Video Interfacing & Conversion » Modular interfacing and signal processing »

The Aperi A1105 1RU Switch/Compute platform comprises of 10/40 GbE, industry-leading wire-speed switching silicon and five accelerated, pluggable microservers providing unequaled density and integration of low-latency switching and high performance computing.

The consolidation of Ethernet switching fabric and microserver compute elements enables the lowest rack density and power efficiency possible while ensuring optimal reliability for service aggregation in an architecture that scales perfectly for data center and network access low-latency service applications. Depending on applications the microserver hardware acceleration can achieve multiples of compute over traditional x86 processing at a fraction of the consumed power resulting in tens of rack units of servers, associated cabling and switching collapsing into a 1RU A1105. The 10/40GbE switch executes all standard and high performance L2/L3 protocols through the Broadcom Fastpath networking software. The A1105 is Open Flow capable. Integrated network clock synchronization is achieved using PTPv2 or externally fed sync. The A1105 10 Gigabit Ethernet Switch offers 6 ports of 10/100/1000BASE-T with 8 ports of wire-speed 10 Gigabit Ethernet via SFP+ cages and 2 ports of 40 Gigabit Ethernet via QSFP cages. Exceptional reliability and serviceability are ensured with dual redundant hot-swappable AC or DC power supplies and hot-swappable redundant fans featuring front to back airflow.

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

The Aperi H.264 App offers a unidirectional H.264 encoder/decoder supporting up to 2 channels of SD-SDI, HD-SDI, and 3G-SDI SMPTE compliant video signals. Each channel operates completely independent of the other, with support for video sources running in different formats and at different frame rates. The encoder automatically detects any valid video input on each channel and packetizes each H.264 compressed video signal into an elementary stream multiplexed with MPEG-1 Audio Layer II audio compression and ancillary data elementary streams to create an MPEG-2 compliant transport stream. This transport stream is encapsulated in a SMPTE 2022-2 compliant IP flow and aggregated into 10GbE interfaces on the microserver edge connector. The decoder receives an IP flow for processing, which removes the SMPTE 2022 encapsulation and converts the signal into a transport stream (TS). The stream is separated into video, audio, and ancillary elementary streams which are synchronized via a traffic clock that is closely matched to the encoder. The elementary video data is then decoded, producing an uncompressed video signal. The management interface is accessible via an open RESTful API in addition to an intuitive graphical user interface via integrated web server to configure and monitor the app directly through the API.

Technical Information

• Detection of two unidirectional channels of SDI video compliant with SMPTE 259, SMPTE 292/274, and SMPTE 424/274.


• H.264 transport stream mapping compliant with IEC 13818-1. ▪ YCrCb 4:2:2, 8-bit or 10-bit, and 4:2:0, 8-bit code streams.


• Compressed ES rates of up to 150Mbps (I-frame only) and down to 5Mbps for IP frame GOP.


• Per-channel encapsulation and de-encapsulation into RTP/UDP/IP packets per SMPTE 2022-2.


• FEC insertion as per SMPTE 2022-1. ▪ Transparent transfer of ancillary data per video in accordance with SMPTE 2038 and TR-01.


• Configurable to Baseline, Main, High, High 10, and High 4:2:2 H.264 profiles.


• Ultra-low latency, configurable to less than 1ms.


• Configurable GOP structure.


• Automatic Dolby E audio pass-through.


• MPEG-1 Audio Layer II audio compression up to 384 Kbps.


• Dual presentation of packets via two separate physical interfaces for optimum per-channel hitless protection in compliance with SMPTE 2022-7.


• Dual path IGMP V2 and V3 Multicast with fast leave.


• Decoder PDV buffer for network jitter, configurable up to 200ms.


• Decoders with and without B-frame Decoding available.


• RESTful HTTP open media API, automatically discoverable, exposing all primitives.


• Intuitive, automatically discoverable user interface via integrated web server.

Support » Monitoring, Control & Communications » Video Monitoring » Multiviewer display processors »

The Aperi IP Multiviewer App (“Multiviewer”) is an FPGA-based application designed to run on an Aperi microserver and is available from the Aperi App Store.

Multiviewer receives IP video input from three sources: on the leftmost “SFP 1” 10 GbE front panel port, and on the two 10 GbE back plane Ethernet ports. The two 10 GbE backplane Ethernet ports support up to six SMPTE 2022-6 (uncompressed) feeds per port (12 total). The front input supports up to six SMPTE 2022-6 (uncompressed) feeds via the Aperi Uncompressed over IP App.

Multiviewer output is split into a 16 tile HD-SDI for display, forming a 4x4 mosaic which is outputted via HD-SDI on the rightmost front panel “SFP 4” port. Each tile can be labeled to show its multicast address, as well as incorporating dynamically definable and static text.

Each input includes an analyzer, which detects for frozen picture, blank screen, or audio loss from any particular source. If an error is detected, the UI will indicate an issue with that particular tile via the alarm system. This alarm system is available via a common interface, and is exposed in the Multiviewer API.

Technical Information

• Supports up to six SMPTE 2022-6 IP video inputs from the 10 GbE SFP 1 port on an Aperi Microserver.


• Supports up to six SMPTE 2022-6 IP video inputs from each of the 10 GbE backplane Ethernet ports (6 + 6) on the Aperi A1105 Chassis.


• 16-tile 4x4 mosaic display of all video sources, outputted via HD-SDI on SFP 4.


• Support for quick-switching any stream from the 16x16 mosaic into full screen display.


• Full labeling support for each tile, displaying multicast address or user-definable text.


• Detection of frozen picture, blank screen, and audio loss via a common alarm interface.


• Graphical user interface available via HTTPS, accessible via any modern web browser.


• Built upon a RESTful API according to industry best practices, along with full API documentation accessible directly via the Multiviewer UI.


• Included Multiviewer User Guide PDF document packaged with the app, and available directly from the UI.

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

The Aperi JPEG 2000 (“J2K”) Encoder and Decoder Apps are FPGA-based applications designed for Aperi's microserver platform, and are available from the Aperi App Store. These apps enable up to a four-channel unidirectional J2K encoder along with up to a four-channel J2K decoder for SD, HD, and up to two channels of 3G SMPTE compliant video signals. The encoder automatically detects any valid video input on each channel, and packetizes each J2K compressed video signal into an elementary stream multiplexed with audio and data elementary streams to create an MPEG-2 compliant transport stream. This transport stream is encapsulated in a SMPTE 2022-2 compliant IP flow, and aggregates them into 10 GbE interfaces on the microserver edge connector. Each J2K compressed IP flow is duplicated at the 10 GbE outputs of the encoder to enable diverse routing. The J2K decoder includes the capability to receive duplicate flows on separate, temporally aligned physical interfaces prior to packet validation & selection. This process provides the hitless switching capability for the apps. The J2K apps are ideal for scenarios requiring preserved picture quality in a compressed format. It delivers visually lossless (and mathematically lossless for some content) reconstruction at higher elementary stream rates. The UI is built above an open, highly accessible RESTful API that provides numerous monitoring and configuration parameters. It's designed to adhere to the standard cloud model and comply with evolving industry standards for device discovery and registration. It seamlessly integrates under any orchestration platform.

Technical Information

• Detection of 4 unidirectional channels of SDI video compliant with SMPTE 259M, SMPTE 292M or SMPTE 424M.


• JPEG 2000 mapping compliant with TR-01.


• YCrCb 4:2:2, 10 bit, J2K ISO/IEC 15444-1 code stream.


• Compressed ES rates of up to 460Mbps for HD-SDI.


• Per-channel encapsulation and de-encapsulation into RTP/UDP/IP packets per SMPTE 2022-2.


• Selectable two or four AES uncompressed audio per video in accordance with SMPTE 302 and TR-01.


• Transparent transfer of ancillary data per video in accordance with SMPTE 2038 and TR-01.


• Dual presentation of packets via two separate physical interfaces for optimum per-channel hitless protection in compliance with SMPTE 2022-7.


• Dual path IGMP V2 and V3 Multicast with fast leave.


• PDV buffer for network jitter, configurable up to 200ms.


• RESTFUL HTTP open media API, automatically discoverable, exposing all primitives.


• Intuitive, automatically discoverable user interface via integrated web server.

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

The Aperi ‘ASI Transport Stream to IP Network Adapter App’ is an FPGA based app that has been designed to run on Aperi’s SDMP microserver. The app can be downloaded to an SDMP from an app store. An Aperi license is needed to create an entitlement to run the app. The app consists of a 4 channel uni-directional transport stream encapsulator and a 4 channel uni-directional de-encapsulator for ASI video signals compliant with DVB and ISO/IEC MPEG-2 transport stream standards. The encapsulation process is compliant with SMPTE 2022-2. The Encapsulator app will automatically detect any valid 188 byte or 204 byte ASI video input on each channel for processing. The Encapsulator app aggregates IP encapsulated MPEG-2 single or multi-program transport streams into a 10GbE interface and also provides a duplicate 10GbE signal for protection purposes. The inverse functions are applied to all receive channels and a PDV buffer is included on each receive channel to absorb network jitter. FEC can be added to each of the 4 IP flows created at the encapsulator which is compliant with SMPTE 2022-1 and the two flows received at the de-encapsulator can be temporally aligned, validated and selected to provide a SMPTE 2022-7 capability. The ASI outputs are presented in packet mode as defined in the DVB-ASI specification. The app can be used in application where it is necessary to connect a standards compliant ASI signal to an Ethernet network or vice-versa. This could be at the ASI output of any Encoder that uses MPEG-2 transport streams. The app is also ideal for transporting ASI singles from over an Ethernet network as a service in a contribution or distribution broadcast network. The user interface is built above an Open RESTful API that exposes all monitoring and configuration parameters. The API has been designed to adhere to the standard cloud model and to enable compliance with the evolving industry standards for device registration and discovery. The API can also be accessed by higher level management systems to allow integration within a single network control platform.

Technical Information

• Detection of 4 uni-directional channels of ASI compliant with EN 50083-9 carrying single or multi-program transport streams.


• Extracts 188 byte or 204 byte transport streams from ASI inputs.


• Encapsulation and de-encapsulation into RTP/UDP/IP packets per SMPTE 2022-2.


• FEC protection compliant with SMPTE 2022-1.


• Dual presentation of packets via two separate physical interfaces for optimum per-channel hitless protection in compliance with SMPTE 2022-7.


• Dual path IGMP V2 and V3 Multicast with fast leave.


• De-encapsulator PDV buffer for network jitter, configurable up to 200ms. - ASI transport stream outputs linearized in packet mode.


• RESTFUL HTTP open media API, automatically discoverable, exposing all primitives.


• Intuitive, automatically discoverable user interface via integrated web server.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » Video-over-internet »

Aperi's Network Address Translation & Firewall app (“NAT app”) is an FPGA-based application designed for the microserver platform, and is available from the Aperi
App Store. Providing seamless inbound and outbound re-routing of RTP streams within your network infrastructure, NAT app also includes RTP stream health monitoring, which will automatically shut down streams that deviate from a specified policy. Stream replication allows for two distinct output streams with independent IP addresses, ports, and VLANs. Failover can merge multiple source streams, selecting the highest quality single stream and automatically switching to a perfect replication should the
original stream degrade in quality or deviate from policy. The supports the SMPTE 2022 standard for transporting both compressed (2022-2) and uncompressed (2022-6) media flows via IP. NAT app provides an open bandwidth budget across the 10 GbE interfaces, and can route varying numbers of compressed (2022-2) streams depending on size, and up to 6 uncompressed (2022-6) streams per SFP port. It also provides support for the SMPTE 2110 standard, and can route video (2110-20), audio (2110-30), and ancillary data (2110-40) flows independently through networks. NAT app provides an open bandwidth budget across the 10 GbE interfaces, and can route up to 64 independent media flows per port. It offers stream duplication for media flows via SFP, and A/B failover for flows outputting through SFPs.

Technical Information
Firewall
 4-layer inspection for basic ACL:
 Ethernet: mac.dst, mac.src, vlan, vlanid
 IP: ip.dst, ip.src
 UDP: udp.dst, udp.src
 RTP: rtp.src
 Additional inspection of RTP payloads for ACL
 Ensure 2022-2 payloads conform to transport stream structure and have correct packet count
 Ensure 2022-6 payloads include RTP header extension, SDI types in headers match configured values, and detect if frame size of encapsulated SDI matches configured value
 Ensure presence of RFC4175 payload header and indicated frame size matches configured values; ensure encapsulated raster video frame size matches configured value
Address Translation
 4-layer packet header rewrite enforces complete re-addressing of every packet before egress
 All unicast/multicast mappings are supported
 For multicast flows, flood, IGMPv2 and v3 are supported
Monitoring and Stats
 Ingress statistics include:
 Ethernet: Frames received, FCS errors, packet rate, and total average bandwidth
 Firewall-overall & firewall-per-flow counts total packets dropped, accepted, average speed and bandwidth
 RTP flow shows RTP type (and payload header if applicable), sequence number, transport stream checking, and PDV/inter packet arrival time
 For 2022-2, shows TS packets per payload, TS lock, TS rate, and payload alignment
 For 2022-6, shows SDI type indicated by payload header versus one detected by line/sample counting, and SDI TRS lock and errors
 For RFC4175, shows video raster type indicated by payload header versus type detected by line/sample counting
 Egress statistics show total packet count, packet speed, and average rate per flow and for Ethernet
Hitless Merge
 Configurable PDV buffer depth
 Creates a new regenerated hitless transport with new addressing
 Counts total number of recovered and unrecoverable packets, along with the differential delay between input flows

Produce » Real-Time Production » Video Production » Real-time production video servers »

The SDMP-4S or Software Defined Media Platform is a modular microserver designed to run software applications (apps) that require intense computational processing power and very low latency. Although the SDMP has a system architecture that has similarities to conventional servers, it uses state-of-the-art technology that provides the processing capabilities needed in apps such as live television production, financial trading and deep machine learning. Multiple apps can be staged on a microserver simultaneously enabling extensive adaptability for the platform. The microserver is based on the data concept of ‘single unit of deployment’ to run all apps. The apps can be downloaded as containerized bit files from an app store or repository and then run from memory in the low latency recomposable processor (FPGA). Physical access to the SDMP is accommodated via four SFP+/video SFP (small form factor pluggable) cages on the front panel of the microserver and two 10GbE parallel interface card edge connections at the rear. The microserver operates in the Aperi A1105 Ethernet Switch/Compute Platform (see A1105 data sheet). The A1105 platform can house up to five microservers and each microserver interfaces to ports on the Ethernet Switch in the A1105 via the two 10GbE ports on the microserver edge connector. This topology enables the seamless deployment of clusters of Ethernet connected microservers in a truly elastic architecture. The microserver has its own Linux based Aperi operating system. This OS has an associated ‘infrastructure API’ which is independent of the Open APIs that are integrated within each app that runs on the microserver. The microserver API assists with the orchestration and deployment of apps onto a microserver from an app repository.

Any app running on the microserver has as an integrated web server with an automatically discoverable intuitive user interface. The web server is accessed via the management port of the platform in which the microserver running the app is populated. Comprehensive information related to the microserver hardware on which the app is running is also accessible via this UI.

• Chassis - Populatse Aperi A1105 1RU Ethernet Switch/Compute Platform
• Module Form Factor - Industry standard telecommunications form factor defined by PICMG specifications
• Data Plane Processor - Xilinx FPGA with 2.5GBytes of DDR3–1.6GHz (also available with a quad core FPGA)
• Control Plane Processor - Processor with quad core using full Linux operating System
• App Staging on Microserver - Includes 16GBytes of flash memory to enable many different apps and app versions and instantaneously switch between them
• Networking - Dual 10GbE connections via card edge IEEE802.3 (also available in with dual 40GbE connections via card edge)
• Synchronization - Dual reference timing input for GPS, 1PPS or application layer (video) clock. IEEE 1588 capable.
• Application Layer Physical Interface - 4 x small form factor pluggable (SFP/SFP+) cages for any standards compliant modules
• Application Programming Interface - Open RESTFUL API using standard cloud model
• Graphical User Interface - Integrated web server with intuitive UI via HTTPS for all apps including microserver hardware status and authentication and authorization for security
• App Store - Apps can be managed and downloaded from an ‘app
store’ or app repository and run using an Aperi app license
• Applications - The microserver is ideal for low latency and intense computational processing applications including live
production, financial trading and image processing

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

Aperi’s TICO Encoder and Decoder are FPGA-based applications designed for Aperi’s microserver platform and are available from the Aperi App Store.

The TICO Encoder and Decoder app enables a 4K UHD channel that provides virtually lossless quality. The 4K UHD signal is received as four 3G feeds and employs the TICO mezzanine compression to create a codestream that is packetized and encapsulated per SMPTE RDD35, conforming to SMPTE 2110/VSF TR-03 standards. It’s formatted as a quad-link 2SI.

Audio data is de-embedded from the first channel, and mapped into IP with either AES67 or RFC-3190. Ancillary data is encapsulated into IP with the SMPTE 2038 specification.

In the 3G quad link SDI mode, up to four bidirectional channels can simultaneously process all de-embedding, compression, and encapsulation, effectively bypassing the quad-link formatter, and processing independently. SMPTE 2022-7 RTP replication and merging is used for network protection.

Extensive flexibility and precise configuration options make Aperi’s TICO encoder and decoder ideal for a variety of scenarios. It effectively can encode and decode 4K UHD with near mathematically lossless 4K and 3G SDI quality, all at extremely low-latencies.

Technical Information

Supported Formats

• 4K UHD Quad-Link 3G-SDI per ST 425-5 with one simultaneous encoding or decoding channel.


• 4x 3G-SDI with up to four simultaneous encoding or decoding channels. Compression


• Offers TICO Encode (Profile 2) and TICO Decode (Profile 1 & 2).


• Visually lossless at 4:1 CBR, and virtually mathematically lossless at 2:1 CBR. RTP Encapsulation


• Video encapsulated according to SMPTE RDD35.


• Audio encapsulated according to AES67.


• Ancillary data encapsulated according to SMPTE 2038. Timing and Synchronization.


• SMPTE 2059-2 PTP (Precision Time Protocol) used when available.


• High quality reference if PTP isn’t available.

Networking

• RTP replication and hitless merging according to SMPTE 2022-7.


• Dual path IGMP v2 and IGMP v3.


• Multicast with fast leave.


• PDV buffer for network jitter, configurable up to 200ms.

User Interface

• Open RESTful HTTP HAL & JSON open media API


• Automatically discoverable, highly intuitive interface offering extensive monitoring and configuration

Publish » Linear Signal Distribution » Distribution & Re-distribution » Encoding and multiplexing for distribution and re-distribution »

The Aperi Uncompressed SDI over IP (S22-567) App is an FPGA-based application designed for our microserver platform, and is available from the Aperi App Store. Offering up to four-channel bidirectional SDI encapsulator and decapsulator for SD, HD, and 3G SMPTE compliant video signals, it automatically detects any valid source on each input for processing. The app aggregates the IP encapsulated signals into 10 GbE interfaces, with supported FEC capability. FEC packets are added at the RTP layer of each encapsulated IP flow. Inverse functions are applied to all channels, with a PDV buffer on each to absorb network jitter. Uncompressed flows are duplicated at the 10 GbE outputs of the encapsulator, enabling the use of diverse routing. Each decapsulator can receive duplicate flows on separate physical interfaces, which are temporally aligned prior to packet validation and selection. These features provide a hitless switching capability. Our uncompressed SDI to IP network adaptor app provides unparalleled performance in transporting SDI signals over an Ethernet network. Additional use cases, such as replacing an SDI router with an Ethernet switch in an SDI plant. The versatility of transporting SDI via Ethernet lends the app to numerous present uses cases, along with uses in larger Ethernet networks in the future.

An integrated web server is included with the app, which is automatically detected and can be easily accessed via the chassis management port and viewed in any modern internet browser. The UI is built above an open, highly accessible RESTful API that provides numerous monitoring and configuration parameters. It's designed to adhere to the standard cloud model and comply with evolving industry standards for device discovery and registration. It easily integrates under any orchestration platform.

• Detection of 4 bi-directional channels of SDI video compliant with SMPTE 259M, SMPTE 292M and SMPTE 424M.


• SDI jitter management per SMPTE RP184YCrCb 4:2:2, 8-bit or 10-bit, and 4:2:0, 8-bit code streams.


• Per-channel encapsulation and de-encapsulation into/from RTP/UDP/IP packets per SMPTE 2022-6.


• Per-channel protection via FEC in compliant with SMPTE 2022-5.


• Dual presentation of packets via two separate physical interfaces for optimum hitless protection in compliance with SMPTE 2022-7.


• Dual path IGMP V2 and V3 Multicast with fast leave.


• PDV buffer for network jitter, configurable up to 200ms.


• Network over-subscription policing on all SDI inputs.


• RESTful HTTP open media API, automatically discoverable, exposing all primitives.


• Intuitive, automatically discoverable user interface via integrated web server.

Support » Monitoring, Control & Communications » System Monitoring & Control » Network management, monitoring and orchestration »

V-Stack is the next-generation cloud management software developed by Aperi for managing the A1105 platform. V-Stack offers powerful tools to manage your chassis, SDMP microservers, and all of your apps.

As a single point of view for chassis management, V-Stack manages all of the microservers, Aperi Apps logistics, and provides dynamic information about the operation of a chassis. With V-Stack, Aperi SDMP microservers are automatically configured for use and kept track of within the software. Aperi Apps can be easily imported via V-Stack from the Aperi App Store, and then launched directly to installed microservers with just a few clicks.

V-Stack also offers an intuitive web-based user interface that adheres to standard design principles, along with a documented REST API for developers to automate their chassis management.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The wide range of Acme Packet platforms is designed to help customers deliver trusted, real-time communications across internet protocol (IP) network borders based on their different needs for performance and capacity.

Technical and Business Benefits

• Wide range of platforms to meet different customer needs


• Consistent set of features across platforms provides the extra flexibility in deployment


• Scalable network performance and capacity


• Advanced DoS/DDoS protection, utilizing hardware-assisted, dynamic, fine-grained controls


• Choices in network interfaces and DSP modules for customized solutions

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

With its swiss-army-knife approach, the Nimbra 400 family is the very definition of versatility. Engineered for the harsh realities of the internet, the Nimbra 400 family supports a wide range of video and transport technologies. The combination of native support for both AVC and HEVC encoding/decoding, compressed video over IP and ASI and an open and flexible approach to IP networking in a super-efficient 1 RU form factor makes the Nimbra 400 family a perfect companion for space constrained remote sites and flight kits. With native support for RIST, SRT, Zixi and our own Edge Connect software means you can get your content anywhere. The integrated encryption and firewall paired with transport of any remote management traffic and comms also minimizes the need for third party equipment on site.

Paired with its standalone software counterpart, the Virtual Nimbra, means there are solutions ranging from highly redundant appliances for flight-kit and racks, to flexible software for datacenters and public cloud providers.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Combining native video, audio and high quality ethernet transport with switching, the Nimbra 300 provides a high-quality multiservice solution for demanding media applications. Multi-service capabilities allow you to transport live and file-based content across the same network, and with native support for DTT (ATSC 3.0, DVB-T2, ISDB-T) transport, Broadcast Radio distribution and video contribution the Nimbra 300 series gives you all the flexibility you need.

With support for service admission control and guaranteed bandwidth, performance and characteristics different services can safely be mixed no matter the network load. With hitless 1+1 protection of media and data services you can be sure that the expected service quality will always be delivered.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation WAN transport-platform designed for all-IP services and all-IP networks, guaranteeing quality for the most demanding live media events while providing agility through automated real-time provisioning. Scaling both switching and link capacities to levels unprecedented in the media WAN space, enabling event sites, stadiums, studios, and production sites to be interconnected using a total of 1,2 Tbps of guaranteed capacity in just a compact 3RU modular form factor.Future proof building practice with fully flexible cooling and power configurations allow deployment in any modern service provider, broadcaster or data center environment. Designed with a virtualization architecture, using one common hardware platform for all media and network functions, enabling flexible deployment of functions anywhere in the network. Modular and redundant, with hot swappable and protected components guarantees uptime and reliability.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Next generation of media transport designed for ultimate reliability, flexibility and versatility. Nimbra Edge is a fully cloud-based media transport solution designed for private or public cloud environments.Nimbra Edge augments existing Nimbra devices by drastically simplifying the network architecture through offloading complexity such as redundancy and scalability to its cloud-based backend. With its open approach to networking, Nimbra Edge connects to any media device supporting RIST, SRT, Zixi or Net Insights Edge connect software. The latter designed to cloud-enable any third-party device or workflow.

Connect » Provide Connectivity » Connectivity & Bandwidth » Fiber and private IP connectivity »

The LTN® Network is a broadcast-quality IP multicast network with 99.999% reliability and ultra-low latency. The Network connects 3,000+ locations with global audiences and powers LTN solutions to drive content value.

With proprietary protocols for dynamic multicarrier routing and rapid error recovery, the LTN Network is engineered to outsmart congestion and preempt failure, ensuring consistent live video delivery.

Backed by the LTN Network, major media producers and distributors expand their reach, increase their revenue, and deliver transformative video experiences.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

LTN Leaf Flypack™ is a portable connection to the LTN Network that enables professional broadcasters to deliver flawless live events — easily, at scale, and from any location. Packing the power of a satellite truck in a cost-efficient carry-on, Flypack provides broadcast quality, ultra-low latency, and global reach, backed by our fully managed and monitored multicast IP network.

A complete IP solution for any live event, Flypack features plug-and-play setup, offers encryption capabilities, and encodes and decodes at up to 20 mbps with SD and HD support.

By connecting Flypack to the LTN Network, you can manage and control video from origin to destination, monitoring every signal with LTN Portal and always-on NOC support.

Power the future of video with LTN Leaf Flypack.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Whether it’s for contribution or distribution, full-time or occasional use, LTN Leaf is a cost-effective, fully managed solution for transmitting or receiving broadcast-quality video.

The Leaf appliance requires minimal on-site setup, and features a compact, integrated design that harnesses the flexibility of multiple encode and decode configurations. Leaf also enables seamless access to the LTN Network, with real-time monitoring and always-on support.

The Leaf encode mode supports interactive interviews and live shots with lower end-to-end latency than satellite. Leaf also features broadcast-quality encoding for fast-motion sports and complex tape playout content at bitrates up to 20 Mbps, with high-quality multi-codec decoding of up to four simultaneous feeds.

Create the future of video with LTN Leaf.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

CatonNet Video Platform (CVP), powered by Caton Transport Protocols, provides broadcast grade media transmission services. CVP is designed as a highly available, low latency and secure service ensuring your video can be transmitted and received without compromising on quality, security or speed. With a point of presence in over 60 countries worldwide, some of the largest content creators, broadcasters, satellite operators and service providers are using this fully managed end-to-end service with up to 99.999% stream availability to securely deliver superior video to their customers locally, regionally and globally. All while enjoying significant cost savings compared to traditional network services.

Support » Monitoring, Control & Communications » System Monitoring & Control » Broadcast system control SW and HW »

 

HOME is a management platform for IP-based media infrastructures. It is designed to connect, manage and secure all aspects and instances of live production environments. HOME provides the tools and centralized services for swift and effective interaction of engineers with their tools.

HOME is cloud-native by design and ready to run anywhere, irrespective of the system’s size. With HOME, the cloud starts on your campus, private and locally. It turns an array of devices, setups, sites, hubs and data centers into a powerful, agile network — quickly and in a perfectly secure way.

Inside HOME, discovery of devices is automatic, while registering and admitting them to the network is only a button press away.

With the adoption of IP well underway, the focus of operators has shifted from whether to adopt IP to how to use its potential with minimal effort and maximum effect. This is where HOME shines: it addresses all pressing issues real-world operators face today and tomorrow. In one place and via a single, platform-agnostic, intuitive user interface.

Lawo’s HOME platform is based on open standards such as ST2110, NMOS, IEEE802.1x and RADIUS and has been designed and built from the ground up using LUX—the Lawo Unified Experience, a framework for conceiving, designing, and building solutions that put you first and defines the standard for user experience and design across the Lawo portfolio.

Discovery and Registration: HOME solves IP complexity with automatic plug & play discovery of IP audio and video devices, which are registered with their name, location, status and type. This applies not only to Lawo products but to third-party solutions as well via NMOS. Discovered devices are managed in a central inventory list, ready for access and configuration.

Device Management: In today’s hectic live broadcast environments, operators rely on speedy, unified device configuration routines, especially when setting generic device parameters or configuring senders and receivers. The ability to save and recall configurations is key to speed up tasks. HOME provides a centralized “mission control” for these processes, providing fast and unified access to device parameters for easy tweaking, irrespective of the end point being controlled.

Operability: With its simple, user-friendly UI, HOME allows users to organize and access processing services. With all required facilities accessible in one place, operators can set up and change stream configurations, and route them across an infrastructure without the need for a separate controller. For large infrastructures HOME works seamlessly with a broadcast controller in the same set-up and helps to speed up configuration and operation. HOME is based on LUX, a UI language common to all Lawo devices and many of their functionalities. Through HOME’s user interface, operators can access and edit device parameters quickly utilizing integral mechanisms that help get the job done efficiently. With HOME, operators quickly get right to what they’re looking for, without distractions and complications, to focus 100% on the task at hand.

Security: The content created by a production crew and transported over a network is any operation’s most valuable asset and deserves strong protection. While a robust security system needs to cover all aspects of media infrastructure and content creation, the key lies in its simplicity. HOME provides a variety of security strategies, first of which is quarantining unknown devices when they come online. Only after being deliberately approved, via an intuitive IEEE802.1X-based routine, can they begin exchanging signals with the HOME network. Secondly, HOME uses an authentication strategy based on a centralized user management system, with dedicated user roles and groups. The LDAP based service allows users to authenticate either locally – within HOME – or via their own corporate IT infrastructure, e.g. Microsoft® Active Directory. Finally comes the arbitration of devices and individual streams based on pinpointed rights management. HOME’s architecture is prepared to manage services such as transport layer security, network segmentation and other IT security mechanisms such as RADIUS.

Scalable Architecture: Home is cloud-native by design, which means that its architecture is built to run detached from hardware constraints. This does not automatically mean that services must be outsourced to an external service provider whose meter is running 24/7; with HOME, the cloud starts on your campus, private and locally, on COTS hardware. The HOME platform is designed as functional blocks that provide microservices, which are self-contained and supply functionality to operators or other services.

HOME can be expanded with additional services at any time to increase its functionality — the platform scales on demand. Should there be a need for a larger RDS, because the installation grows, additional instances of the required resources can be added anytime. One of the core principles of HOME is its focus on the utilization of open standards wherever possible, for broadest compatibility and future-proof integration. With HOME, flexibility and resource utilization in IP media infrastructure maximizes.

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Connect » Provide Connectivity » Access Connectivity »

The next generation of digital STL
Our challenge: to design a studio-to-transmitter link (STL) for your most demanding 950 MHz applications — one as reliable and robust as Intraplex® T1 and IP audio links. Digital STL manages much more than audio, so installation and configuration need to be straightforward, not a science project. Data should not be optional, requiring additional boxes and complexity. Your STL should be ready, out of the box, for AM, FM and HD Radio™, as well as future multimedia applications.

Earlier digital STLs were not designed for IP data transport. Adding IP data to them required optional modules and external add-ons, and many engineers working on HD Radio installations reported spending much time and money trying to eliminate glitches.

Our solution: HD Link™ is designed to manage all HD Radio™ transport scenarios, regardless of where you place your importer and exporter. Its two prioritized Ethernet paths give preference to HD Radio™ data over control and other LAN/WAN data. It supports both UDP and TCP, and even handles the switching of TCP return packets over asymmetric IP paths with plug-and-play simplicity.

HD Link offers RF power to spare, an integrated IP gateway with sophisticated data handling capabilities, and multiple channels of audio. The intuitive front panel and remote interfaces tap into the most complete feature set of any microwave STL, yet take less time to configure. It can even operate on both RF and IP simultaneously, and allows automatic backup of all services from one to the other.

Intraplex is broadcasting’s first choice for rock-solid, full-time operation of T1 STLs. With HD Link, you can now count on the same dependable performance, superior support and long-term value for your microwave links.

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The Intraplex® IP Link family of IP audio codecs provides high-end features at an affordable price. Offering an array of audio coding options along with IPConnect technology for data tunneling, the IP Link codecs are suitable for use in Studio-to-Transmitter Links (STLs) as well as audio contribution and distribution networks. Support for IP multicast and multiple unicast streams enables one encoder to feed multiple decoders.

By incorporating three IP Interfaces that can be used for streaming and management, the IP Link systems can provide a level of reliability not seen in comparably-priced codecs.

As the latest additions to the Intraplex line of data transport products, the IP Link family of audio codecs bring legendary Intraplex reliability to the IP codec market.

IP Link 100/100p: Single bidirectional stereo audio channel IP Link 200: Two bidirectional stereo audio channels IP Link 200A: Two bidirectional stereo audio channels with one channel being AES67 Standard: Linear, AAC-LC, Opus and G.722 audio coding Optional: AAC-HE, AAC-HEv2, AACELD, MPEG2, MPEG3 and Enhanced aptX audio coding Optional: Automatic audio loudness leveling and metering compliant with EBU R-128 and ITU-R Optional: IPConnect capability to reliably transport external IP packets Other transport modes: Transparent AES up to 192 kbps to support composite FM multiplex signal over AES Protocol Encapsulation: RTP, SHOUTcast/Icecast, MPEG-TS Three independent IP interfaces for redundant network operation Optional redundant power supply: 12VDC or 48VDC Built-in silence sensor with optional stream switch over Automatic backup to audio playout from USB drive or external audio source Multicoding can encode the same audio source in multiple formats for STL, backup, and web streaming Optional Dynamic Stream Splicing providing “hitless” operation and T1/ E1 circuit like performance on less predictable IP networks 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 with Intraplex IP Link Scheduler for automated scheduled program switching Integrated with Intraplex LiveLook (network analytics and monitoring software) N+1 redundancy with integrated control of external switching equipment IP Link 200/200A/100p: SynchroCast™ option provides dynamically managed precision delay for Single Frequency Network (SFN) broadcasting and simulcasting Support for IP multicast and multi-unicast Web browser user interface and SNMP network management Eight multipurpose contact closure inputs and outputs provide: Transport of logic signals with time-alignment to audio Stream control Alarm notification

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Intraplex® IPConnect is an IP networking gateway solution for fortification of mission-critical communication across wide area networks
IPConnect solves the problem of reliable transport across complex networks that traditionally suffer data loss through dropped packets. IPConnect provides an added layer of reliability for many types of IP traffic such as HD Radio, DAB/DAB+, or DRM by employing a combination of packet protection schemes with network/time diversity and packet-level forward error correction. Additionally, IPConnect can bridge local area network (LAN) segments across wide-area networks with seamless tunneling to enhance reliability of program and signal transport across large geographical regions with multiple receiving and transmitting sites.

With its three network interfaces, IPConnect’s data gateway provides extra protection through packet encapsulation, which encloses external IP data packets in a GatesAir protocol wrapper as it moves across IP networks. IPConnect provides this protection for studio-generated data, as well as IP data from external sources. While the former might include RDS data or SNMP control signals to trigger a command at the transmitter site, the latter may incorporate program or control data coming from a network operations center, satellite feed or advertising service.

As the next-generation member in GatesAir Intraplex family of IP codecs, the award-winning IPConnect leverages the strengths of previous IP Link codecs—including GatesAir’s patented Dynamic Stream Splicing (DSS) technology to mitigate IP packet loss, eliminate off-air time, and optimize stream reliability.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK1616S is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 16 channels of audio sent from the Dark1616S into the network. The Dark1616S has 8 of AES3 inputs and 16 of analogue inputs. Using the GlenController App it is possible to set which inputs (AES3 or Analogue) are being sent to the Dante/ AES67 network, an auto mode is available that sends AES3 (when a valid signal is detected) in preference to the analogue input.

Simultaneously there are 16 channels of audio being received from the network by the Dark1616S and these incoming circuits are provided as outputs from the Dark1616S in both AES3 and analogue.

The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz, the incoming AES3 circuit is always sample rate converted to match the Dante network frequency.

The AES3 outputs have sample rate converters on them and can be locked to the sample frequency of the Word clock input, the first AES3 input or to the Dante network.

A Word clock output is also provided, this is clocked at the same sample rate as the Dante network and can be used for locking external equipment to the network's sample rate.

The unit can be remote controlled using our GlenController Windows 10 App. The App allows such things as the dBFs levels to be set, clock masters and also provides input/ output metering. See the App tab for further details.

For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Being designed for resilient broadcast applications, the Dark1616S features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616S lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616S units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The Glensound DARK1616D is a digital to digital converter designed to connect AES3 audio circuits to/ from a Dante/ AES67 audio network.

Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Dark1616D Dante audio interface will be compatible with any other manufacturers Dante audio interface.

Being designed for live on-air broadcast applications the Glensound Dark1616D has been designed with multiple redundancy capabilities. It has 2 mains power sources and it also has fully redundant network connections for both Copper & Fibre circuits.

The Dark1616D provides 8 balanced AES3 inputs and 8 balanced AES3 outputs to the Dante network on rear panel D25 connectors wired to AES59 (also known as the Tascam standard).

As per our other Dante equipment 0dBu = -18dBFs

Create » Content Acquisition and Capture » Audio Aquisition » Audio acquisition accessories »

The GS-FW012 ip is the Dante network audio version of the GS-FW012 4W unit.

There are still four inputs for the top panel loudspeaker or headphone monitoring, each with their own level controls and each being derived from the Dante network.  There are still four talkback outputs each being routed to the Dante network.

Both the inputs and outputs are presented via a single rear panel Neutricon RJ45/CAT5 connection that is Dante/AES67 compliant.

The four talkback outputs are identical.  Each has a 3 position lever key talk switch that is either off, locked on, or in a sprung push to talk mode.  The output of each circuit can be off unless the talkback button is pressed, looping the the associated Dante input input, or outputting the local cue input.

The local cue input is presented on XLR and has it's own level pot for the speaker or headphone monitoring.

A good quality microphone amplifier with adjustable gain followed by a compressor limiter is built into the unit making its talkback outputs to the Dante network crystal clear. Conversion of the audio signals to/ from the Dante network are by low noise, high bandwidth 48K 24bit converters providing superb performance.

Audio I/O can be routed via the digital Dante router.  The free Dante Controller software configures all routes and can be downloaded by clicking here.

There is an internal switch mode power supply, or the GS-FW012 ip can be powered via PoE via the CAT5 connection. If both supplies are present then they will act as redundant supplies.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK16O is a versatile break out box for receiving analogue audio from a network utilising the Dante and AES67 audio over IP (AoIP) protocols.

In total there are 16 channels of audio being received from the AoIP network and converted via low noise DACs (Digital to Analogue Converters) to balanced analogue outputs.

For ease of cabling audio outputs are presented as balanced circuits on Neutrik XLRs.

Being designed for resilient broadcast applications the DARK16O features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16O lost its primary link circuit, the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16O units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK16I is a versatile box for converting analogue audio inputs to an AoIP system utilising the Dante and AES67 audio over IP (AoIP) protocols.

In total there are 16 audio input channels, all being simultaneously converted to the AoIP network via high quality low noise analogue to digital converters.

For ease of cabling audio inputs are balanced circuits on Neutrik XLRs.

Being designed for resilient broadcast applications, the DARK16I features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16I lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16I units can be directly connected with each other with no external network hardware.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK88 MKII is a versatile break in/ out box for sending/ receiving analogue audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 8 channels of audio sent from the DARK88 MKII into the network. The DARK88 MKII has 8 of analogue electronically balanced audio inputs on Neutrik XLRs.

Simultaneously there are 8 channels of audio being received from the network by the DARK88 MKII and these incoming circuits are provided as outputs from the DARK88 MKII in analogue.

Network sample rates of up to 192KHz are accommodated seamlessly within the DARK88 MKII.

Being designed for resilient broadcast applications the DARK88 MKII features both redundant power supplies and redundant Dante network links. Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK88 MKII lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK88 MKII units can be directly connected with each other with no external network hardware.

On the front panel, LEDs indicate the status of the 2 power supplies and the 2 network links. GPO status outputs are also provided for external indication of the power supply & network status.

Network connections are placed on the front panel of the DARK88 MKII in order that the network cables (or fibres) match those of a rack mounted professional network switch, making installation and tracing interconnecting cables easy. Fibre connections are via SFP slots, meaning that users can select their own preferred fibre type & connector style by installing their own fibre SFP modules (a selection of modules is available from Glensound if preferred).

The DARK88 MKII features the Brooklyn module from Audinate which is AES67 compliant.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

The DARK8ADI (Analogue & Digital Input) is a very powerful Dante®/ AES67 network audio interface, in a robust 1RU 19" subrack It has a 8 of low noise analogue line inputs and 4 of transformer balanced AES3 digital audio inputs. These inputs are transmitted on a fully redundant Dante/ AES67 network interface.

In total 16 audio circuits are sent to the Dante/ AES67 network. Eight of these circuits are always derived from the 4 x AES3 inputs, the other eight are automatically switched between the AES3 and analogue inputs. If the DARK8ADI detects valid AES3 signal on a channel then this will be routed to the network output.  If no valid AES3 is detected then the analogue input will be routed to the network output instead of the AES3.

The rugged design, redundant mains powering & redundant network facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.

Eight Analogue Audio InputsThe DARK8ADI has 8 electronically balanced analogue line level audio inputs. Each input is on its own Neutrik 3 pin XLR socket.Quality Analogue To Digital ConvertersTo get the best possible results from your analogue audio inputs, the very best widest range analogue to digital converters (ADCs) currently available are used to make sure the digital audio on your network is as good as it possibly can be.Four AES3 Audio InputsThe DARK8ADI has 4 transformer balanced digital AES3 audio inputs. Each input is on its own Neutrik 3 pin XLR socket. These inputs can accept sample rates up to 192kHz.

Analogue/ AES3 Auto SwitchingEight of the audio outputs are derived from both the 4 x AES3 inputs and the 8 analogue inputs, whereby the AES3 inputs take priority to the analogue circuits. Our input circuitry looks for valid AES3 data streams on the AES3 inputs.  If one is detected then the pair of audio outputs (AES3 is two audio channels) associated with that circuit will be routed to the Dante/ AES67 output and if no valid AES3 signal is detected then the analogue input will be routed to that output.

AES3 Network OutputsAs well as the switched outputs, the outputs of the 4 x AES3 input circuits are always routed to 8 audio channels of the AoIP network.

Sample Rate ConvertersThe AES3 audio inputs are fed through sample rate converters so that they match the AoIP networks sample frequency. The AoIP network supports up to 192kHz sampling and the AES3 inputs can be between 32 & 192kHz.

Network InterfaceThere are 4 network interfaces on the DARK8ADI. There are 2 x Neutrik Ethercon (RJ45) connectors and there are also 2 x SFP slots for customers to fit their own preferred fibre interfaces.

Redundant Network InterfaceWhen using the Dante protocol it is possible to set the DARK8ADI to have a fully redundant network interface whereby a completely glitch free automatic redundant audio network link is provided across 2 of the network interfaces.

PowerEach DARK8ADI can be powered from two independent sources to provide a multi-redundant power option.Two wide range switch mode power supplies are fitted as standard to provide redundant mains power supplies.

Alarms & LEDsThe front panel features two LEDs (one indicating OK and the other fault) for both power supplies and both network ports. This data OK/ Fault information is also provided on solid state relay outputs on front panel D connectors.

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

GRID encompasses a technology and a methodology to enable guaranteed data flows with effectively-enforced QoS in any IP network: a studio network, a post-production network, a data center network, etc.

GRID’s software transforms the IP network and the data flows to be fully predictable, non-blocking and 100% reliable. As a result, your network will be used much more cost-efficiently. GRID also provides several wizards to make configuration and management of the network easy and intuitive. The software enables a 100% service guarantee with a specific bandwidth, low latency and no packet-loss per service or flow.

GRID is compatible with the hardware of multiple vendors.

Publish » Non-Linear Publishing » Non-Linear Publishing Systems » Cable/IPTV VOD servers and systems »

Software solution for sophisticated transport stream splicing. Runs on generic enterprise grade Windows server hardware. Scalable and flexible configuration. Low propagation delay and avoids full steam decoding when operating in frame accurate mode,

Applications include Ad Insertion, Slate insertion (covering content) or live content replacement.

Support » Design, Build, Run » Implementation Services » Consulting services »

Edge, is available as a lightweight functional block on a VM, container in-line with other software or on bare metal which is specified according to the required throughput.

It can either be used standalone through a web interface or in most cases connects over websockets to our control and monitoring platform called CORE (deployed on prem or in the cloud).

Typical use-cases are secure and reliable Internet transit, a monitoring probe, 2022-7 protection, Input protection and (De)Scrambling.

 

Functions that set itself apart from a native SRT implementation are:

•    SMPTE 2022-7 seamless input with per path statistics

•    SRT RTP header passthrough allowing 1+1 paths followed by downstream 2022-7 seamless protection

•    BISS-2 scrambling and descrambling with seamless odd/even key transition

•    Thumbnailing (based on ffmpeg library)

•    ETR290 Priority 1, Priority 2 & Priority 3 (fully written in house)

•    UDP and RTP output with packet pacing and launch delay offset

•    Threshold based input redundancy with support passive or active backup sources

•    CORE (CORE is the name of our control platform) - Centralised control and monitoring capable of controlling and monitoring through firewalls and presenting a northbound REST control and monitoring API

•    CORE - High availability horizontally scalable cluster based control and monitoring capable of managing tens of thousands of EDGEs.

•    CORE has a REST API with existing integrations with ATOS BNCS and Skyline Dataminer.

•    CORE supports local, LDAP, SAML and Azure AD authentication.

•    CORE - Grafana native driver for correlated monitoring of all software components

•    Floating centralised license model with both OPEX and CAPEX options

•    EDGE Debugging tools - top talkers & PCAP

•    HLS input and output

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

Scalable multiple channel transport stream switching running on generic enterprise grade Windows servers. Flexible control options and auto redundant configurations

Connect » Real-Time Signal Transfer » Intra-Facility Connectivity » IP networks and switches and SDVN »

L-GRID tackles the problem of speed mismatches at the source and avoids an extra investment in large buffer core switches.

L-GRID will configure the network card to only allow bursts of the configured bandwidth. Data flows to the preconfigured destinations are sent out of the network card perfectly shaped to the given bandwidth. As the L-GRID eliminates bursts before they enter the network, no exhaustive buffer space is required from the connected switches. The traffic is controlled/managed at the source. The cause for congestion is remediated before it can happen.

L-GRID enables the user to configure specific bandwidths for specific/configured clients. L-GRID runs as a user-space app on a Linux server and is configured via a config file or a REST interface. A certified network card is required

L-GRID will thereby improve the efficiency of your installed network.

Publish » Internet Distribution » Internet Distribution » Content delivery networks (CDNs) »

Network 1 is a reliable and cost-effective internet-based platform, for the primary distribution of linear content. It offers content owners 24/7/365, point to multi-point channel distribution to global audiences. Network 1 empowers customers to focus on the content, not the distribution.

From public service broadcasters, to global media giants, the UK's oldest commercial network and premier league football teams, Network 1 customers are increasingly utilising global connectivity to transport linear video content.

Network 1 offers significantly quicker implementation than traditional fibre and satellite distribution methods, with set-up normally measured in hours rather than months. Network 1 is multi-cloud compatible, protocol agnostic and delivers over managed or unmanaged networks. Infinite scalability ensures on-boarding new channels is easy and if no hardware is required, customers can deliver to new affiliates or broadcast platform operators straight away.

Network 1 is a fully managed service, with network management and monitoring capabilities, offering complete reliability.

Publish » Internet Distribution » Internet Distribution » Internet distribution platforms (incl. social media) »

Livelink is a robust IP delivery solution for transporting live linear and OTT content from point to single-point or multi-point. It allows users to self-manage their live content for delivery to any destination worldwide by utilizing cloud environments available in every region. Livelink is cloud agnostic and compatible with any transport protocol including Zixi, RIST, SRT, HLS and RTMP.

Livelink’s expand on demand infrastructure ensures that teams can quickly scale their requirements within minutes. This flexible approach enables stakeholders managing live broadcasts and events to adapt to changing circumstances. Livelink does not rely on traditional infrastructure which makes it extremely cost-effective. Its intuitive self-service interface enables users to easily transport, manage and extract value from live content. The platform integrates rights management, automatic booking updates and customisable monitoring and alerts so that customers retain control of all aspects of delivery.

Livelink offers broadcasters the ability to deploy and deliver live feeds with minimal on-site presence. The platform offers global reach and is the ideal solution to reduce live event overheads and also supports current social distancing measures which the broadcasting industry has implemented. The self-management interface monitors availability, routing and scheduling for all feeds, as well as tracking cloud environment costs. Cerberus Tech can fulfil hardware requirements, direct to the end user if required. However, in many instances no additional hardware is required for set-up, just an internet connection.

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The DARK1616 is a versatile break in/ out box for sending/ receiving both analogue and digital (AES3) audio to/ from a network utilising the Dante audio over IP (AoIP) protocol.

In total there are 16 channels of audio sent from the Dark1616 into the network. The Dark1616 has 8 of AES3 inputs and 16 of analogue inputs, but they cannot be used at the same time.   The AES3 inputs take priority over the analogue (if an AES3 input is receiving a valid AES3 signal then it will turn off the equivalent analogue input pair and route its output to the network).

Simultaneously there are 16 channels of audio being received from the network by the Dark1616 and these incoming circuits are provided as outputs from the Dark1616 in both AES3 and analogue.

The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz.  The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.

The AES3 outputs are locked to the sample frequency of the Dante network.

For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Being designed for resilient broadcast applications the Dark1616 features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616 lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.

On the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.

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The DARK 1616M is an audio interface that provides connections to a Dante or AES67 IP network.

Inputs & OutputsThere are 16 digital inputs and 16 digital outputs, available on 8 x AES connections.  There are also 16 analogue inputs and 16 analogue outputs, available in parallel to the digital.  The analogue and digital outputs are always available, and the inputs work either on AES input priority or can be selected via the control app.

The AES3 inputs have sample rate converters and can accept input frequencies up to 192kHz.  The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.  The AES3 outputs are locked to the sample frequency of the Dante network.

ConnectionsFor ease of cabling, audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.

Remote Mic AmpsThe analogue inputs are switchable between mic, line and 48v phantom power.  These are controllable across the network via the Dark Controller Windows 10 app.  This allows remote control by an engineer of the input selection, gain adjust and input on/off control.  A meter is also provided for each input on the app, to monitor the input level and help with setting the gain.

Network InterfaceBeing designed for resilient broadcast applications the Dark1616M features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616M lost its primary link circuit the secondary link would automatically take over with no loss of audio.

The primary and secondary network interfaces are routed internally via a network switch.  It is possible to set this switch to work as a traditional network switch instead of the default redundant mode.  This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.

AlarmsOn the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.

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The MDX series of high performance switching solutions unlock the benefits of a media-tuned, non-blocking architecture and are ideal for end-to-end contribution to distribution networks with guaranteed bandwidth and low latency. All Media Links’ MDX switches offer 100% Quality of Service (QoS) and our patented non-blocking switch architecture for feature-rich solutions across multicast applications. They are designed to combine compressed and uncompressed video / audio streams as well as TCP traffic over IP.

• Fault tolerant switching hardware


• TICO and JPEG compression for bandwidth conservation


• Fast OpenFlow switching


• Non-blocking switch architecture


• Integrated provisioning and network management


• SMPTE 2110 enabled


• NEBS Certified

Available Models:

• MDX32C Core Switch comes equipped with 32 x QSFP28 switch ports. Each port supports  1 x 40/100GbE, 2 x 50GbE, or 4 x 10/25GbE.


• MDX48x6C Aggregation Switch comes equipped with 48 x SFP28 (1G/10G/25G) + 6 QSFP28 (100G), a switching capacity of 1.8 Tbps, plus OpenFlow and Media Links’ Non-blocking Technology.

Applications & Use Cases:

• Carrier Class Media Delivery Networks


• High Performance Studio Interconnects


• Flawless Contribution Video Transport


• Contribution Video over Terrestrial and Satellite Networks

DOWNLOADS:

• DOWNLOAD Datasheet for MDX-32C


• DOWNLOAD Datasheet for MDX-48X6C


• WATCH 100G Eco-System Video


• DOWNLOAD Media Links Solutions Brochure


• DOWNLOAD North America Case Study (Telecom)


• DOWNLOAD Metropolitan Distribution Network Application Note


• DOWNLOAD Live Sports Production Application Note


• DOWNLOAD Centralized IP Switching Application Note


• DOWNLOAD WAN IP Network Application Note

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The AVN-PA8/D and the AVN-PM8/D Portals 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 and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PA8/D and AVN-PM8/D can be combined with multiple Sonifex AVN-HA1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• Analogue audio input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• Loop through audio output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector (if a portal is not being used to supply the unit with power).

Note: The AVN-HA1 is an analogue input product taking an analogue audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate analogue input.

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The AVN-PD8/D Portal can be combined with multiple AVN-HD1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.

The AVN-PD8/D can be combined with multiple Sonifex AVN-HD1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.

• Front panel 6.35mm (1/4”) headphone socket and volume control knob.


• Front panel Mute/GPO push button.


• AES3 digital input on RJ45 (the connector provides power to the unit and a GPO back to the portal).


• AES3 digital output on RJ45 (power and GPO signal are not connected).


• Locking DC power connector if a portal is not being used to supply the unit with power.

Note: The AVN-HD1 is an AES3 digital input product taking an AES3 audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate AES3 audio input.

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RouteMaster uses Rascular’s proven router control and emulation technology to form a powerful yet flexible router control system. Built on tried and tested software modules, RouteMaster can be used with a wide range of video and audio routers  from all major manufacturers.

It’s equally suitable for new router installations or increasing the capabilities – and lifetime – of existing systems. You can add third-party hardware panels to existing systems, add IP connectivity to older serial routers, or simply increase the number of control ports available.

The integrated web server means operators can control routers from any web browser, using custom control panels built with Helm Designer. And because it’s built on HTML5 standards, there’s no need for browser plugins.

 

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Video signals have always depended on precise synchronization along the entire delivery chain from camera to consumer display, whether in analog, digital or compressed formats. Timing drives every aspect of signal delivery, particularly synchronizing video to audio to achieve “lip sync.”

With the transition to IP networks for all aspects of the signal processing path, accurate timing becomes more difficult, due to the fundamentally asynchronous, non-deterministic nature of packet-based networks. Fortunately, a solution is available, in the form of IEEE 1588 Precision Time Protocol (PTP). When used properly, this technology can synchronize device clocks to within nanoseconds across a large network with many hundreds of nodes. When these clocks are derived from GPS (Global Positioning System) signals, PTP can provide a very accurate and stable timebase for all types of signals within modern media operations.

The Quarra line of PTP IP solutions compliment Artel's media delivery portfolio offering end users an excellent solution set for delivering real-time media IP network. As the broadcast industry continues to migrate toward all-IP networks, providing reliable solutions to manage IP-based workflows becomes paramount to maintaining efficiencies and ensuring the delivery of mission-critical multimedia streams.

The Quarra PTP Ethernet switches provide leading-edge precision timing protocol switch technologies and have become de-facto standards for the AES67/ SMPTE ST-2059-2 (IEEE1588) audio market. Designed for Carrier Class applications where accurate timing and control is required, Quarra switches are aimed at the professional markets of Audio/Video Broadcast, Defense & Security, Finance, Utilities, Telecom, and Enterprise IT. The Quarra switches are designed as either a standard 1RU rackmount unit with dual power supplies, or a half-width 1RU unit with a single power supply, the switches are available in several configurations of 1 Gigabit and 10 Gigabit interfaces, supporting a best-in-class layer 2 IEEE 1588v2 algorithm. A 1pps external reference input from GPS is included as standard, and options include an internal layer 1 Synchronous Ethernet module. The Quarra switches support SMPTE Standards-based Timing ST 2110-10 and ST 2059-2 and are RAVENNA AES67 approved. The Quarra switches also support IGMP v2/3, QoS, protection switching, VLAN, MEF service delivery and network OAM.

Ideal for:
• Live video/audio broadcast/production over IP utilizing IEEE1588 timing, QoS and Multicast filtering.
• Broadcast DTT/DTV IP distribution, utilizing IEEE1588 and SyncE timing to replace or backup GPS timing at base stations.
• Telecom networks, providing high level of network timing.
• Financial/trading networks, utilizing our IEEE1588 precise nanosecond time synchronization.

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Artel’s SMART Media Delivery Platform™ is the conduit to simplifying the convergence of IT and broadcast ecosystems. As a base platform supporting native IP media delivery, the SMART platform’s architecture provides feature flexibility and future functional expandability via software download. By using the latest networking and processor technology the SMART Media Delivery Platform has achieved a high level of integration and the agility to navigate media through next-generation networks.

•Integrated Switching Capability - A non-blocking layer 2/3 switching and routing engine has been incorporated into the SMART platform to interface with the network eliminating the need for an external switch or router. The integrated switching capability allows for deep packet inspection, traffic management, flow control, protection switching, and link aggregation that can be easily configured through an SNMP interface.

•Control and Management - Simplifying the migration to an IP infrastructure, the SMART platform’s control and management interface allows the ease of integration, operation, problem detection and isolation.

•Designed for Service Provider and Broadcast Networks - The SMART platform is designed to support transport and data services, and broadcast network video, audio, and IT network applications. With a single module, multiple ports of media may be aggregated and injected into the network without the need for different physical elements. The SMART supports SMPTE ST 2022-7 hitless switching, VLAN tagging, QoS, and traffic management. These functions along with greater port density drastically reduces power consumption, size and cost.

•Software Defined Functionality - The migration to IP has increase the need for more agile solutions. The SMART platform is designed to update functionality and add new applications via software download providing a solution to manage multi-platform content delivery.

The SMART Media Delivery Platform’s small footprint, true data networking and transport versatility, software-defined functionality, and high density opens up new opportunities for media transport and broadcast applications and allows for a seamless transition as IP networks evolve.

A finalist for the 2018 IABM BaM Awards in the Connect category, the SMART Media Delivery Platform™ is a carrier-grade, software-defined platform with integrated nonblocking Layer 2/3 switching and routing capabilities. Designed to attach seamlessly to the IP network without the need for external network elements, the SMART platform supports SMPTE ST 2022: 1, 2, 5, 6, and 7 hitless switching; QoS; VLAN tagging; and traffic management. The platform features four video ports for transport of video, audio, and ancillary data and four GigE data ports bridged to a 10G interface. The SMART solution is software-enabled, providing an easy and efficient platform for adding or upgrading functionality via software download.

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Main Advantages
High reliability through the dual redundant power supply
Increases service availability significantly thanks to the automatic or manual operational mode, the Automatic Switch Back and the automatic memorisation of the configuration
Cost-effective solution through GbE logical switching
Easy setup and installation through the ability to copy the full configuration from one device to another
Easy to use, control and operate through its user-configurable switching logic and the Automatic Switch Back
Easy integration into NMS systems through Monitoring & Control via SNMP
Highly compact

Key features
Dual redundant power supply with monitoring
Automatic or manual operational mode
Logical Ethernet switching
Automatic Switch Back
Copy the full configuration between devices
Automatic memorization of configuration
Reference device
GUI with synoptic view
User-configurable switching logic
Switching of ASI, IP, IF, L-band and RF signals
Stand alone operation or integrated in a network management system
Monitoring & control via SNMP
Management of device configuration

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The Macnica MPA1000 4K AV over 1GbE Module merges the promise of AV over IP with the reliability and performance of ST 2110 in a module that has that has what it takes to deliver encrypted UHD 60P 4:4:4 content and control over a standard 1GbE connection.

The MPA1000 Module is field configurable for transmit or receive. Only one part number is all that is needed for delivering routable UHD video, audio, USB, and HDMI control signals (EDID, HDCP, HPD, CEC, etc) over IP.

Once connected to the network, the module’s traffic is discoverable through the RESTful NMOS protocol - an open standard for publishing and discovery of content streams in broadcast. Simple configuration and control is the core principle in the design of this module. Every module includes a built-in web server with a RESTful API and a web application that is ready for customization. Host access is provided through an optional dedicated standard interface (Ethernet or serial).

Additionally, Macnica offers the MPA1000 Development Kit for test and development. Design files available upon request (currently for limited alpha customers only). Please contact Macnica directly for price and availability.

• 4K AV over 1GbE based on SMPTE ST 2110 standards


• Capable of transporting HDMI2.0 4K60p 4:4:4 HDMI 2.0 AV and Control Signals (Configurable as Rx or Tx) over standard 1Gigabit Ethernet network


• Encrypted USB 2.0


• Sub-frame Latency


• JPEG-XS based Visually-Lossless Compression


• Synchronous Sources


• Full-stack API with Web Interface and Scripting Built In

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The ME10 SoC is Macnica’s full-stack AV over IP system on a chip (SoC), transporting HDMI 2.0 4K 4:4:4 video, audio and control data over 1GbE networks in a small, 23 x 23 mm BGA package. The ME10 allows manufacturers to include IPMX interoperability in any video product with a minimal investment in engineering. The ME10 is ideal for larger production runs or for use in more compact products. It features Low-latency synchronized transport based on AIMS IPMX open standard combining SMPTE ST 2110, AMWA NMOS and JPEG-XS compression standards, AES128 Encryption, RESTful API and Lua scripting environment as well as a customizable Web User Interface.

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Everything you need to begin developing 4K AV-over-IP products with IPMX interoperability, all preconfigured so you can begin using it right away.

Use the Launch Kit to evaluate how standards-based video will work with your current products, to develop new products, and as a reference to the connections and configurations you will need as you go into production.