Video: ST 2110-30 and NMOS IS-08 — Audio Transport and Routing

Andreas Hildebrand starts by introducing 2110 and how it works in terms of sending the essences separately using multicast IP. This talk focusses on the ability of audio-only devices to subscribe to the audio streams without needing the video streams. Andreas then goes on to introduce AES67 which is a standard defining interoperability for audio defining timing, session description, encoding, QOS, transport and much more. Of all the things which are defined in AES67, discovery was deliberately not included and Andreas explains why.

Within SMPTE 2110, there are constraints added to AES67 under the sub-standard 2110-30. The different categories A, B and C (and their X counterparts) are explained in terms how how many audios are defined and the sample lengths with their implications detailed.

As for discovery and other aspects of creating a working system, Andreas looks towards AMWA’s NMOS suite summarising the specifications for Discovery & Registration, Connection Management, Network Control, Event & Tally, Audio Channel Mapping. It’s the latter which is the focus of the last part of this talk.

IS-08 defines a way of defining input and output blocks allowing a channel mapping to be defined. Using IS-05, we can determine which source stream should connect to which destination device. Then IS-08 gives the capability to determine which of the audios within this stream can be mapped to the output(s) of the receiving device and on top of this allows mapping from multiple received streams into the output(s) of one device. The talk then finishes with a deeper look at this process including where example code can be found.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX

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.

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Speaker

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

Video: AES67 Open Media Standard for Pro-Audio Networks

AES67 is a method of sending audio over IP which was standardised by the Audio Engineering Society as a way of sending uncompressed video over networks between equipment. It’s become widespread and is part of SMPTE’s professional essences-over-IP standards suite, ST 2110.

Here, Conrad Bebbington gives us an introduction to AES67 explaining why AES67 exists and what it tries to achieve. Conrad then goes on to look at interoperability with other competing standards like Dante. After going into some implementation details, importantly, the video then looks the ‘Session Description Protocol’, SDP, and ‘Session Initialisation Protocol’, SIP which are important parts of how AES67 works.

Other topics covered are:

  • Packetisation – how much audio is in a packet, number of channels etc.
  • Synchronisation – using PTP
  • What are SDP and SIP and how are they used
  • Use of IGMP multicast
  • Implementation availability in open source software

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For a more in-depth look at AES67, watch this video

Speakers

Conrad Bebbington Conrad Bebbington
Software Engineer,
Cisco

Video: The Audio Parts of ST 2110 Explained

At the IBC 2018 IP Showcase, Andreas Hildebrand explains how AES67 and 2110 work together and how technologies like Dante, RAVENNA and Livewire fit in.

While there are lots of resources for working with 2110 video, but this is one of the few which tackles Audio. Andreas covers one of the ‘gotchas’ in 2110 – the compatability requirements for AES within the standard. He then looks at the timing requirements of 2110 and how they differ to those of AES67 and finally discusses AES3 while explaining the ST 2110-31 standard.

Presenter

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager and Evangelist for the RAVENNA technology developed by ALC NetworX, Germany,