Video: Introduction To AES67 & SMPTE ST 2110

While standardisation of video and audio over IP is welcome, this does leave us with a plethora of standards numbers to keep track of along with interoperability edge cases to keep track of. Audio-over-IP standard AES67 is part of the SMPTE ST-2110 standards suite and was born largely from RAVENNA which is still in use in it’s own right. It’s with this backdrop that Andreas Hildebrand from ALC NetworX who have been developing RAVENNA for 10 years now, takes the mic to explain how this all fits together. Whilst there are many technologies at play, this webinar focusses on AES67 and 2110.

Andreas explains how AES67 started out of a plan to unite the many proprietary audio-over-IP formats. For instance, synchronisation – like ST 2110 as we’ll see later – was based on PTP. Andreas gives an overview of this synchronisation and then we shows how they looked at each of the OSI layers and defined a technology that could service everyone. RTP, the Real-time Transport Protocol has been in use for a long time for transport of video and audio so made a perfect option for the transport layer. Andreas highlights the important timing information in the headers and how it can be delivered by unicast or IGMP multicast.

As for the audio, standard PCM is the audio of choice here. Andreas details the different format options available such as 24-bit with 8 channels and 48 samples per packet. By varying the format permutations, we can increase the sample rate to 96kHz or modify the number of audio tracks. To signal all of this format information, Session Description Protocol messages are sent which are small text files outlining the format of the upcoming audio. These are defined in RFC 4566. For a deeper introduction to IP basics and these topics, have a look at Ed Calverly’s talk.

The second half of the video is an introduction to ST-2110. A deeper dive can be found elsewhere on the site from Wes Simpson.
Andreas starts from the basis of ST 2022-6 showing how that was an SDI-based format where all the audio, video and metadata were combined together. ST 2110 brings the splitting of media, known as ‘essences’, which allows them to follow separate workflows without requiring lots of de-embedding and embedding processes.

Like most modern standards, ATSC 3.0 is another example, SMPTE ST 2110 is a suite of many standards documents. Andreas takes the time to explain each one and the ones currently being worked on. The first standard is ST 2110-10 which defines the use of PTP for timing and synchronisation. This uses SMPTE ST 2059 to relate PTP time to the phase of media essences.

2110-20 is up next and is the main standard that defines use of uncompressed video with headline features such as being raster/resolution agnostic, colour sampling and more. 2110-21 defines traffic shaping. Andreas takes time to explain why traffic shaping is necessary and what Narrow, Narrow-Linear, Wide mean in terms of packet timing. Finishing the video theme, 2110-22 defines the carriage of mezzanine-compressed video. Intended for compression like TICO and JPEG XS which have light, fast compression, this is the first time that compressed media has entered the 2110 suite.

2110-30 marks the beginning of the audio standards describing how AES67 can be used. As Andreas demonstrates, AES67 has some modes which are not compatible, so he spends time explaining the constraints and how to implement this. For more detail on this topic, check out his previous talk on the matter. 2110-31 introduces AES3 audio which, like in SDI, provides both the ability to have PCM audio, but also non-PCM audio like Dolby E and D.

Finishing up the talk, we hear about 2110-40 which governs transport of ancillary metadata and a look to the standards still being written, 2110-23 Single Video essence over multiple 2110-20 streams, 2110-24 for transport of SD signals and 2110-41 Transport of extensible, dynamic metadata.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX Gmbh.

Video: Implementing AES67 and ST 2110-30 in Your Plant

AES67 is a flexible standard but with this there is complexity and nuance. Implementing it within ST 2110-30 takes some care and this talk covers lessons learnt in doing exactly that.

AES67 is a standard defined by the Audio Engineering Society to enable high-performance audio-over-IP streaming interoperability between various AoIP systems like Dante, WheatNet-IP and Livewire. It provides comprehensive interoperability recommendations in the areas of synchronization, media clock identification, network transport, encoding and streaming, session description, and connection management.

The SMPTE ST 2110 standards suite makes it possible to separately route and break away the essence streams – audio, video, and ancillary data. ST 2110-30 addresses system requirements and payload formats for uncompressed audio streams and refers to the subset of AES67 standard.

In this video Dominic Giambo from Wheatsone Corporation discusses tips for implementing AES67 and ST 2110-30 standards in a lab environment consisting of over 160 devices (consoles, sufraces, hardware and software I/O blades) and 3 different automation systems. The aim of the test was to pass audio through every single device creating a very long chain to detect any defects.

The following topics are covered:

  • SMPTE ST 2110-30 as a subset of AES67 (support of the PTP profile defined in SMPTE ST 2059-2, an offset value of zero between the media clock and the RTP stream clock, option to force a device to operate in PTP slave-only mode)
  • The importance of using IEEE-1588 PTP v2 master clock for accuracy
  • Packet structure (UDP and RTP header, payload type)
  • Network configuration considerations (mapping out IP and multicast addresses for different vendors, keeping all devices on the same subnet)
  • Discovery and control (SDP stream description files, configuration of signal flow from sources to destinations)

Watch now!

You can download the slides here.

Speaker

Dominic Giambo
Senior Embedded Engineer
Wheatstone Corporation

Video: A Simple Description of SDP in SMPTE ST 2110


Often overlooked, but very important for IP streams such as SMPTE ST 2110, is SDP – Session Description Protocol.

SDP describes the stream by transferring a small text file from the sender to the receiver which defines what the media is (audio, video etc.) the IP address of the sender and other data.

Sithideth Viengkhou from Embrionix takes us through some of the details, shows some examples and explains and why it’s so useful.

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Speakers

Sithideth Viengkhou Sithideth Viengkhou
R&D Project Manager,
Embrionix

Video: Routing AES67

Well ahead of video, audio moved to uncompressed over IP and has been reaping the benefits ever since. With more mature workflows and, as has always been the case, a much higher quantity of feeds than video traditionally has, the solutions have a higher maturity.

Anthony from Ward-Beck Systems talks about the advantages of audio IP and the things which weren’t possible before. In a very accessible talk, you’ll hear as much about soup cans as you will about the more technical aspects, like SDP.

Whilst uncompressed audio over IP started a while ago, it doesn’t mean that it’s not still being developed – in fact it’s the interface with the video world where a lot of the focus is now with SMPTE 2110-30 and -31 determining how audio can flow alongside video and other essences. As has been seen in other talks here on The Broadcast Knowledge there’s a fair bit to know.(Here’s a full list.

To simplify this, Anthony, who is also the Vice Chair of AES Toronto, describes the work the AES is doing to certify equipment as AES 67 ‘compatible’ – and what that would actually mean.

This talk finishes with a walk-through of a real world OB deployment of AES 67 which included the simple touches as using google docs for sharing links as well as more technical techniques such as virtual sound card.

Packed full of easy-to-understand insights which are useful even to those who live for video, this IP Showcase talk is worth a look.

Watch now!

Speaker

Anthony P. Kuzub Anthony P. Kuzub
IP Audio Product Manager,
Ward-Beck Systems