Webinar: RAVENNA and its Relationship to AES67 and SMPTE ST 2110


This webinar is now available on-demand

This first in a series of webinars, this will have a broad scope covering the history of audio networking in, the development of RAVENNA the the consequent developments of AES67 and ST2110. Whether you’re new to or already familiar with RAVENNA and/or AES67 & ST2110 you’ll benefit from this webinar either as revision or as an excellent starting point for understanding the landscape of Audio-over-IP standards and technologies.

This webinar is presented Andreas Hildebrand who has previously appeared on The Broadcast Knowledge giving insight into The Audio Parts of ST 2110, ST 2110-30 and NMOS IS-08 — Audio Transport and Routing amongst others.

This talk looks at how audio IP works and the benefits of using an IP system. Since the invention of RAVENNA, the AES and SMPTE have moved to using audio over IP so the walk will examine how RAVENNA and SMPTE place the audio data on the network and get that to the decoders. Interoperability between systems is important but can only happen if certain parameters are correct, something that Andreas will mention but will also be a subject for future webinars.

Whether you want to revise the basics from an expert or learn them for the first time, now’s the time to register.

Watch now!

Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX GmbH

Video: Reinventing Intercom with SMPTE ST 2110-30

Intercom systems form the backbone of any broadcast production environment. There have been great strides made in the advancement of these systems, and matrix intercoms are very mature solution now, with partylines, IFBs and groups, wide range of connectivity options and easy signal monitoring. However, they have flaws as well. Initial cost is high and there’s lack of flexibility as system size is limited by the matrix port count. It is possible to trunk multiple frames, but it is difficult, expensive and takes rack space. Moreover, everything cables back to a central matrix which might be a single point of failure.

In this presentation, Martin Dyster from The Telos Alliance looks at the parallels between the emergence of Audio over IP (AoIP) standards and the development of products in the intercom market. First a short history of Audio over IP protocols is shown, including Telos Livewire (2003), Audinate Dante (2006), Wheatstone WheatNet (2008) and ALC Networks Ravenna (2010). With all these protocols available a question of interoperability has arisen – if you try to connect equipment using two different AoIP protocols it simply won’t work.

In 2010 The Audio Engineering Society formed the x192 Working Group which was the driving force behind the AES67. This standard was ratified in 2013 and allowed interconnecting audio equipment from different vendors. In 2017 SMPTE adapted AES67 as the audio format for ST 2110 standard.

Audio over IP replaces the idea of connecting all devices “point-to-point” with multicast IP flows – all devices are connected via a common fabric and all audio routes are simply messages that go from one device to another. Martin explains how Telos were inspired by this approach to move away from the matrix based intercoms and create a distributed system, in which there is no central core and DSP processing is built in intercom panels. Each panel contains audio mix engines and a set of AES67 receivers and transmitters which use multicast IP flows. Any ST 2110-30 / AES67 compatible devices present on the network can connect with intercom panels without an external interface. Analog and other baseband audio needs to be converted to ST 2110-30 / AES67.

Martin finishes his presentation by highlighting advantages of AoIP intercom systems, including lower entry and maintenance cost, easy expansion (multi studio or even multi site) and resilient operation (no single point of failure). Moreover, adaptation of multicast IP audio flows removes the need for DAs, patch bays and centralised routers, which reduces cabling and saves rack space.

Watch now!

Download the slides.

If you want to refresh your knowledge about AES67 and ST2110-30, we recomend the Video: Deep Dive into SMPTE ST 2110-30, 31 & AES 67 Audio presentation by Leigh Whitcomb.

Speaker

Martin Dyster
VP Business Development
The Telos Alliance

Video: The Basics of SMPTE ST 2110 in 60 Minutes

SMPTE ST 2110 is a growing suite of standards detailing uncompressed media transport over networks. Now at 8 documents, it’s far more than just ‘video over IP’. This talk looks at the new ways that video can be transported, dealing with PTP timing, creating ‘SDPs’ and is a thorough look at all the documents.

Building on this talk from Ed Calverley which explains how we can use networks to carry uncompressed video, Wes Simpson goes through all the parts of the ST 2110 suite explaining how they work and interoperate as part of the IP Showcase at NAB 2019.

Wes starts by highlighting the new parts of 2110, namely the overview document which gives a high level overview of all the standard docs, the addition of compressed bit-rate video carriage and the recommended practice document for splitting a single video and sending it over multiple links; both of which are detailed later in the talk.

SMPTE ST 2110 is fundamentally different, as highlighted next, in that it splits up all the separate parts of the signal (i.e. video, audio and metadata) so they can be transferred and processed separately. This is a great advantage in terms of reading metadata without having to ingest large amounts of video meaning that the networking and processing requirements are much lighter than they would otherwise be. However, when essences are separated, putting them back together without any synchronisation issues is tricky.

ST 2110-10 deals with timing and knowing which packets of one essence are associated with packets of another essence at any particular point in time. It does this with PTP, which is detailed in IEEE 1588 and also in SMPTE ST 2059-2. Two standards are needed to make this work because the IEEE defined how to derive and carry timing over the network, SMPTE then detailed how to match the PTP times to phases of media. Wes highlights that care needs to be used when using PTP and AES67 as the audio standard requires specific timing parameters.

The next section moves into the video portion of 2110 dealing with video encapsulation on the networks pixel grouping and the headers needed for the packets. Wes then spends some time walking us through calculating the bitrate of a stream. Whilst for most people using a look-up table of standard formats would suffice, understanding how to calculate the throughput helps develop a very good understanding of the way 2110 is carried on the wire as you have to take note not only of the video itself (4:2:2 10 bit, for instance) but also the pixel groupings, UDP, RTP and IP headers.

Timing of packets on the wire isn’t anything new as it is also important for compressed applications, but it is of similar importance to ensure that packets are sent properly paced on wire. This is to say that if you need to send 10 packets, you send them one at a time with equal time between them, not all at once right next to each other. Such ‘micro bursting’ can cause problems not only for the receiver which then needs to use more buffers, but also when mixed with other streams on the network it can affect the efficiency of the routers and switches leading to jitter and possibly dropped packets. 2110-21 sets standards to govern the timing of network pacing for all of the 2110 suite.

Referring back to his warning earlier regarding timing and AES67, Wes now goes into detail on the 2110-30 standard which describes the use of audio for these uncompressed workflows. He explains how the sample rates and packet times relate to the ability to carry multiple audios with some configurations allowing 64 audios in one stream rather than the typical 8.

‘Essences’, rather than media, is a word often heard when talking about 2110. This is an acknowledgement that metadata is just as important as the media described in 2110. It’s sent separately as described by 2110-40. Wes explains the way captions/subtitles, ad triggers, timecode and more can be encapsulated in the stream as ancillary ‘ANC’ packets.

2110-22 is an exciting new addition as this enables the use of compressed video such as VC-2 and JPEG-XS which are ultra low latency codecs allowing the video stream to be reduced by half, a quarter or more. As described in this talk the ability to create workflows on a single IP infrastructure seamlessly moving into and out of compressed video is allowing remote production across countries allowing for equipment to be centralised with people and control surfaces elsewhere.

Noted as ‘forthcoming’ by Wes, but having since been published, is RP 2110-23 which adds back in a feature that was lost when migrating from 2022-6 into 2110 – the ability to send a UHD feed as 4x HD feeds. This can be useful to allow for UHD to be used as a production format but for multiviewers to only need to work in HD mode for monitoring. Wes explains the different modes available. The talk finishes by looking at RTP timestamps and SDPs.

Watch now!
The slides for this talk are available here
Speakers

Wes Simpson Wes Simpson
President,
Telecom Product Consulting

Video: Broadcast 101 – Audio in an IP Infrastructure

Uncompressed audio has been in the IP game a lot longer than uncompressed video. Because of its long history, it’s had chance to create a fair number of formats ahead of the current standard AES67. Since many people were trying to achieve the same thing, we find that some formats are compatible with AES67 – in part, whilst we that others are not compatible.

To navigate this difficult world of compatibility, Axon CTO Peter Schut continues the Broadcast 101 webinar series with this video recorded this month.

Peter starts by explaining the different audio formats available today including Dante, RAVENNA and others and outlines the ways in which they do and don’t interoperate. After spending a couple of minutes summarising each format individually, including the two SMPTE audio formats -30 and -31, he shows a helpful table comparing the,

Timing is next on the list discussing PTP and the way that SMPTE ST 2059 is used then packet time is covered explaining how the RTP payload fits into the equation. This payload directly affects the duration of audio you can fit into a packet. The duration is important in terms of keeping a low latency and is restricted to either 1ms or 125 microseconds by SMPTE ST 2110-30.

Peter finishes up this webinar talking about some further details about the interoperability problems between the formats.

Watch now!

Speaker

Peter Schut Peter Schut
CTO,
Axon