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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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