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At last – a simple solution for MPX signal transport over IP!

At last – a simple solution for MPX signal transport over IP!

At last – a simple solution for MPX signal transport over IP!

Distribution – Infrastructure – Multiplex – Transmitter

D

igigram has developed a solution to transport composite MPX signals from the studio over an IP network with the same quality and reliability as a point-to-point link.

MPX (multiplex) is the composite signal dedicated to the transmitter site. It is the only signal accepted by the transmitter (exciter) for FM diffusion, and it is traditionally delivered via a point-to-point link.

What are the main differences for an MPX signal delivery, between an IP wireless point-to point-link, and an IP network link?

There are two distinct advantages of an MPX-over-IP transport solution for broadcasters:

The first is a significant gain in flexibility and scalability; and the second is a reduction in equipment to be maintained and deployed on the transmitter sites. However, to ensure the same levels of quality and reliability over an IP network as for a point-to-point link, attention needs to be paid to IP transport and jitter mastering.

IP mastering

Built on the universally acclaimed IQOYA audio-over-IP platform, Digigram has developed a highly reliable solution allowing broadcasters to deliver MPX signals over IP networks.   All the mechanisms that have helped to make IQOYA a worldwide success for rock-solid audio-over-IP streaming have been deployed for MPX transport over IP. That includes FEC, dual streaming with time diversity, network jitter mastering, and smart synchronisation on the incoming stream. Plus, additional smart features have been added, such as decoder synchronisation for MFN, and auxiliary data tunnelling.

Jitter mastering

Why is it important to master the jitter? Mastering the jitter guarantees the same reliability as a point-to-point link. That means that transporting MPX signal can be achieved either via point-to-point or over an IP network depending on the broadcaster’s requirements.

Data rate mastering

MPX is a composite signal, hence smart codec design requires far less dynamic range than baseband audio. For instance for Audio stereo + RDS, 144 kHz sample rate @12 bits encoding with analog transmission leads to a very acceptable 2 Mb/s instead of the usual 3.3 Mb/s or more. With AES192 digital MPX transport 2.6 Mb/s can in addition transport an auxiliary subcarrier.

Conclusion:

The IQOYA hardware codec is a highly versatile codec for program delivery. It delivers analog or digital baseband audio or MPX through point-to-point or IP networks links. These baseband audio links are distributed to FM transmitter sites, web radio CDN, and/or DVB multiplexers for satellite, and DAB encoders for digital radio.

Digigram is proud to be one of the few in the world able to transport MPX over IP network with a very high jitter resilience and acceptable bandwidth.

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Five benefits of using a SaaS application for radio outside broadcasting

Five benefits of using a SaaS application for radio outside broadcasting

Five benefits of using a SaaS application for radio outside broadcasting

Contribution – Infrastructure – Application

The high expectations of today’s audiences are driving up content quality. Listener interests, rather than technical constraints, now dictate the nature of content being produced. Looking at these trends as a whole, broadcasters are seeking more flexible outside broadcasting (OB) solutions capable of supporting enriched content creation. Software as a service (SaaS) applications are becoming increasingly popular because they bring a high ease of use while reducing overall costs. Here are five other good reasons for using SaaS for OB.

1. Less money spent on equipment

Thanks to the SaaS model, there is no need to invest in costly equipment or infrastructure. In addition to enabling users to generate a broadcast-quality interview from any connected device, certain SaaS applications are available on a pay-as-you-go model that further reduces costs.

2. More time for content, less for setup

Accessible directly from a web browser and available after a few clicks, a SaaS application can eliminate the need for a technician to set up interviews. Because technical requirements are reduced to their minimum, journalist or producer staff have the freedom to focus on producing quality content instead of sweating over technical matters.

3. Upgrade or downgrade as needed

The best broadcasting SaaS applications give users the ability to extend the number of contributors to a program, adding and removing contributors with ease. This makes a SaaS OB application the perfect tool for radio stations’ editorial managers!

4. Maximum flexibility

Functions such as master control room (MCR) codec operation may be distributed to each studio or centralized according to the workflow habits of each radio station. Accessible from any connected browser on a local PC or a nomadic device, the simplified operation enabled by the SaaS model can allow the technical team to be more adaptable and agile.

5. Automatic synchronous update to the latest versions

A SaaS application for audio OB productions is a time-saver. Technical setup time is minimized because it is simply not needed anymore. Furthermore, certain SaaS applications dedicated to radio broadcasting provide automatic updates and allow users to benefit simultaneously from all the latest versions and options.

Conclusion:

Broadcast professionals are looking for simple, reliable solutions that will help them to create enriched audio content. Where they previously needed a mobile studio, technicians, and expensive or bulky equipment, they now need only a laptop, the internet and a professional SaaS application in order to maintain broadcast quality, performance and continuity of service.

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Three Tips for Setting Up and Managing a Standard and AES67 Network

Three Tips for Setting Up and Managing a Standard and AES67 Network

Three Tips for Setting Up and Managing a Standard and AES67 Network

Contribution – Infrastructure – Studio

A lthough large radio stations and groups typically have the resources to develop dedicated networks for AES67 (IP audio) and for corporate traffic, small and mid-sized stations often need to pursue a less-expensive approach. Here we provide three recommendations that help smaller radio broadcasters ensure the peaceful coexistence of AES67 audio-over-IP (AoIP) traffic and standard network traffic on a single network.

Seamless Handling of AoIP and Standard Traffic on One Network

The AES67 standard makes AoIP protocols such as Dante, Livewire and Ravenna interoperable, in turn simplifying connectivity and reducing hardware and overall clutter. To identify the requirements of handling and optimizing AoIP traffic and standard traffic — web, video transfers, and corporate data — on one network, we examined the interoperability of AES67-compatible products from four different manufacturers and then created three recommendations for small and mid-sized radio broadcasters in setting up and managing AES67 (IP audio) on an existing network.

In short, we recommend the use of Internet Group Management Protocol (IGMP) snooping protocol to distribute predictable bandwidth on a high number of streams; installation of PTP (Precision Time Protocol)-enabled switches; and activation of quality of service (QoS) mechanisms to limit disruption and avoid audio glitches.

Making It Happen: Three Keys to AES67/AoIP on a Standard Network

Recommendation 1: Enable IGMP

First step: Activate the IGMP. By realizing select communications, IGMP enables the management of subscriptions to the multicast addresses. It manages the distribution of network and audio packets, in turn preventing saturation of bandwidth and reducing clutter on the network.

Recommendation 2: Consider PTP

The switch used to support this single-network model can be PTP-enabled or not. If it is, it facilitates better synchronization of the network and is less sensitive to the disruptive effects that can be generated by the corporate network. Although a PTP-enabled switch is more expensive than the alternative, its benefits often are worth the extra cost. When the switch is not PTP-enabled, synchronization or jitter problems may occur during heavy use of the network. (One way to avoid this issue is to configure the QoS present on the switches.) Without PTP support, clock jitter on AES67 traffic becomes quite high above 100 audio channels.

Recommendation 3: Configure Packet Priority via QoS

QoS is used to manage the priority of packets over the network, and it improves the network capabilities of a switch that does not contain PTP. The AES67 standard imposes rules on manufacturers regarding QoS eligibility. It integrates the management of the priorities of the IP packets and to which class of service they are associated. The equipment and the network must follow the AES67 recommendations to ensure a uniform understanding of priorities.

Optimizing the Single-Network Model

Enabling the IGMP is the most important step radio broadcasters can take to avoid saturation of the audio network, regardless of whether or not they are using a PTP-enabled switch. That said, the bottom line is that the combination of PTP and Qos makes it possible to increase the quality and quantity of available audio streams. Through our tests, we determined that the combination of a non-PTP-enabled switch with QoS enabled made it possible to exchange 120 full-duplex channels on the network without loss of packets and without any latency problems.

Conclusion: Creating an AES67 Network is an Option for All

Network audio competence is increasingly essential for radio broadcasters, but today’s AES67-compatible IP-based products for AoIP — and a few helpful tips for implementation — can go a long way in enabling small and mid-sized operations to take advantage of IP audio and its many benefits.

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AES67 for TV applications

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AES67 for TV applications

audio essence will no longer “follow” video; it will be produced independently and dynamically assembled with metadata in the delivery of the final content, leading to new productivity schemes

B y its nature, the delivery of audio over IP infrastructure enables a distributed approach to handling audio streams. Without the need to rely on specific pieces of equipment, connected in a point-to-point model, broadcast and A/V facilities can realize much greater flexibility in routing, scheduling and managing audio streams. At the same time, audio-over-IP (AoIP) technologies simplify operations by allowing users to maintain synchronized content within complex multiple-source, multiple-destination workflows, and to manage metadata more effectively in terms of end-to-end content management and overall operations.

The AES67 standard

The AES67 standard for AoIP interoperability has evolved to the point that its performance is roughly comparable to that of MADI (AES10). With the industry increasingly focusing on system approaches, technology suppliers are addressing the “discovery gap” — which was deliberately omitted by the AES67 Working Group — by bridging stream discovery at the system level or in equipment. At the same time, vendors are pushing forward control and monitoring specifications, such as AES70, NMOS, and others, that further enhance system implementation of IP-based applications.

Audio essence will no longer “follow” video

The value of AoIP to mission critical IP-based audio distribution applications has been widely demonstrated, firstly outside the studio with the ACIP standard and now inside with AES67, and video broadcasters now are considering integrating AES67 into video-over-IP environments. In fact, through the Alliance for IP Media Solutions (AIMS), broadcast equipment and solution suppliers have come together to ensure an easier transition to IP by supporting VSF TR-03/-04, SMPTE 2022-6, and AES67 standards. With such an approach, audio essence will no longer “follow” video; it will be produced independently and dynamically assembled with metadata in the delivery of the final content, leading to new productivity schemes.
When broadcasters and A/V facilities take full advantage of IP infrastructure’s potential to increase their workflow productivity and flexibility far beyond simple gains such as reduced installation or transmission costs, they realize the optimal cost-benefit ratio. To do so, they must undertake solid IP infrastructure engineering, taking care to establish the multicast routing and PTP clock synchronization capabilities essential to mission critical broadcast operation.

Because audio also needs to be transmitted between facilities, several audio manufacturers propose “Media Gateway” solutions to bridge the LAN-synchronous and ultra-low latency AES67 to outside WANs, using the ACIP standard (EBU Tech 3326 and EBU Tech 3368). As an example, for countrywide studio-to-studio transport of AES/EBU, bit transparency provides for the highest PCM audio quality, Dolby E® and user bits transport. This application benefits from a dedicated QoS priority to transport PTP (Precision Time Protocol) clock reference at reasonable jitter on a managed WAN to accurately synchronize and phase audio at the sample level.

Another important use case consists of quickly installing a high-performance audio link between an AES67-enabled OB and an AES67-enabled venue, such as a theater or a stadium.

Since AES67 avoids both dedicated audio lines and costly hardware routing, all applications requiring flexibility, high performance (high channel count, precise time-alignment, low latency…) and low capex could benefit from AES67 interoperability.

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From audio streaming to AES70 device discovery and control

From audio streaming to AES70 device discovery and control

From audio streaming to AES70 device discovery and control

AES67 – Q&A

1) Is the role of AES67 as a sort-of bridging technology that will allow vertically integrated Audio over IP protocols, such as RAVENNA, Livewire+, WheatNet. to transport audio packets to equipment or studios based on other vertically integrated Audio over IP protocols?

Most vertically integrated manufacturers (e.g., Axia and Wheatstone) — in association with Dante-enabled equipment such as Stagetec, DHD or SSL mixing engines, NTP routers — will continue to focus on their own system approaches for control management (that is, everything except audio streaming).
Since most of them are committed to providing an AES67 interface for audio streaming, broadcasters will be able to select the best technology for every function just as they have been able to do for years with AES/EBU or MADI. And as they did with AES/EBU or MADI, broadcasters will continue to rely on transversal studio managers from specialized editors (e.g. BFE and many others) to control their studios with existing protocols, such as EMBER+ or more proprietary protocols.

2) We understand that AES67 is still evolving, for instance work is ongoing to enable a control layer. Is that true? What other areas might be developed to make AES67 more well-rounded over time -perhaps device discovery?

The AES is currently working on device discovery and control protocols, the whole purpose of which is to extend the interoperability beyond the principle of audio transport. As recent examples, AES70 has been released early 2016, and alliances such as AIMS or AMWA promoting IP technologies for broadcast studios are offically requesting the implementation of the NMOS Open Specifications protocol.

3) What is AES70’s evolution in the industry? What should customers know about AES70 and your interaction with it?

  • Unlike AES67 focused on transport and synchronization, AES70 is a door to ocean: Workflow topology, signal descriptions, metadata, scheduling, priorities…
  • Convergence is still a long road because other open (NMOS, EMBER+) or proprietary (Wheatnet, Livewire+…) solutions already exist and tend to retain the added value of each System manufacturer.
  • Because control is a key point in System approach, Digigram has introduced AES70 control in the roadmap of new 2017 products.

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Radio Broadcasting technology: state of the art and best practices

Radio Broadcasting technology: state of the art and best practices

Radio Broadcasting technology: state of the art and best practices

careful engineering must identify the different failure scenarios

What is the most important development happening in AoIP right now for radio broadcasters?

In Outside Broadcasting, SSL and STL applications, IP is no more an option, since ISDN clearly fades out and because IP brings incredible new possibilities. Inside the studio, AES67-2013 has significantly consolidated audio transport interoperability. 35+ manufacturers implement AES67. Several actual “IP studios” are set-up worldwide to benchmark AoIP with actual operation and CAPEX/OPEX.
By nature, AoIP is distributed, no more connecting point-to-point specific equipments. As a consequence, the consecutive “silos” of Radio workflow are merging in service-oriented vertical applications rather than function-oriented, connected equipments.

How far are we in the AES67 interoperability testing effort as an industry?

2015 plugfest showed both great interest and solid interoperability between representative manufacturers, including RAVENNA, Livewire+ and DANTE adopters. This federating effect was confirmed during MNA interoperability setups prepared for ISE 2016 and NAB 2016 (27 products, 13 manufacturers).
Functional level is now roughly comparable to MADI. Nevertheless, solid IP infrastructure engineering with careful Multicast routing is a confirmed must for mission critical broadcast operation.
The industry now focuses on system approaches : first, plug pragmatically the “Discovery” gap intentionally left aside in AES67. Most manufacturers intend to bridge stream discoveries at system level or in equipments. Second by pushing up Control and Monitoring specifications such as AES70 or NMOS.

Some engineers have reported resistance among users to adopting AoIP networks and products, compared to traditional alternatives, especially in smaller markets and budget-conscious environments. Thoughts?

For sure AoIP engineering is more complex and the practical rules (“cooking receipe”) are not yet established. The manufacturers have to come with complete simple to use solutions, with long term support through service contracts.

Assuring reliable performance seems to be a paramount concern when it comes to IP audio codecs. How has the state of the art changed in this area in the past two years?

Confidence is here: STL are rock solid with plenty of streams backups through network redundancy and FEC as our FluidIP technology offers. For example in France most of the Radio programs have been delivered for 5 years on a managed MPLS IP network to 1500+ towers with 100+ years calculated MTBF and 99.99% QoS.
Still ongoing in Remote Broadcasting but no choice because of vanishing, when available, ISDN. For mobile applications IP becomes a must.

What “best practices” can you recommend to broadcast engineers and users in creating and using their IP networks?

First, careful engineering must identify the different failure scenarios, including accurate clock synchronization, not only transport issues.
Then, select equipments providing the different protections to overcome these issues : Multicast IP redundancy schemes, Clock recovery, failover and backup mechanisms.

What else should our readers know about the state or trends in AoIP?

Broadcasters already consider using uncompressed AoIP outside studio on dedicated managed IP networks. The traditional CoDec becomes a “media gateway” bridging the inside and the outside studio streams, what might be compared to WAN gateways in IT networks.

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Don't miss any articles, Subscribe to our Newsletter!