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.
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.
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)
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.