Video: AES67 Over Wide Area Networks


AES67 is a widely adopted standard for moving PCM audio from place to place. Being a standard, it’s ideal for connecting equipment together from different vendors and delivers almost zero latency, lossless audio from place to place. This video looks at use cases for moving AES from its traditional home on a company’s LAN to the WAN.

Discovery’s Eurosport Technology Transformation (ETT) project is a great example of the compelling use case for moving to operations over the WAN. Eurosport’s Olivier Chambin explains that the idea behind the project is to centralise all the processing technology needed for their productions spread across Europe feeding their 60 playout channels.

Control surfaces and some interface equipment is still necessary in the European production offices and commentary points throughout Europe, but the processing is done in two data centres, one in the Netherlands, the other in the UK. This means audio does need to travel between countries over Discovery’s dual MPLS WAN using IGMPv3 multicast with SSM

From a video perspective, the ETT project has adopted 2110 for all essences with NMOS control. Over the WAN, video is sent as JPEG XS but all audio links are 2022-7 2110-30 with well over 10,000 audio streams in total. Timing is done using PTP aware switches with local GNSS-derived PTP with a unicast-over-WAN as a fallback. For more on PTP over WAN have a look at this RTS webinar and this update from Meinberg’s Daniel Boldt.

 

 

Bolstering the push for standards such as AES67 is self-confessed ‘audioholic’ Anthony P. Kuzub from Canada’s CBC. Chair of the local AES section he makes the point that broadcast workflows have long used AES standards to ensure vendor interoperability from microphones to analogue connectors, from grounding to MADI (AES10). This is why AES67 is important as it will ensure that the next generation of equipment can also interoperate.

Surrounding these two case studies is a presentation from Nicolas Sturmel all about the AES SC-02-12-M working group which aims to define the best ways of working to enable easy use of AES67 on the WAN. The key issue here is that AES67 was written expecting short links on a private network that you can completely control. Moving to a WAN or the internet with long-distance links on which your bandwidth or choice of protocols is limited can make AES67 perform badly if you don’t follow the best practices.

To start with, Nicolas urges anyone to check they actually need AES67 over the WAN to start with. Only if you need precise timing (for lip-sync for example) with PCM quality and low latencies from 250ms down to as a little as 5 milliseconds do you really need AES67 instead of using other protocols such as ACIP, he explains. The problem being that any ping on the internet, even to something fairly close, can easily take 16 to 40ms for the round trip. This means you’re guaranteed 8ms of delay, but any one packet could be as late as 20ms known as the Packet Delay Variation (PDV).

Not only do we need to find a way to transmit AES67, but also PTP. The Precise Time Protocol has ways of coping for jitter and delay, but these don’t work well on WAN links whether the delay in one direction may be different to the delay for a packet in the other direction. PTP also isn’t built to deal with the higher delay and jitter involved. PTP over WAN can be done and is a way to deliver a service but using a GPS receiver at each location, as Eurosport does, is a much better solution only hampered by cost and one’s ability to see enough of the sky.

The internet can lose packets. Given a few hours, the internet will nearly always lose packets. To get around this problem, Nicolas looks at using FEC whereby you are constantly sending redundant data. FEC can send up to around 25% extra data so that if any is lost, the extra information sent can be leveraged to determine the lost values and reconstruct the stream. Whilst this is a solid approach, computing the FEC adds delay and the extra data being constantly sent adds a fixed uplift on your bandwidth need. For circuits that have very few issues, this can seem wasteful but having a fixed percentage can also be advantageous for circuits where a predictable bitrate is much more important. Nicolas also highlights that RIST, SRT or ST 2022-7 are other methods that can also work well. He talks about these longer in his talk with Andreas Hildrebrand

The video concludes with a Q&A.

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Speakers

Nicolas Sturmel Nicolas Sturmel
Product Manager – Senior Technologist,
Merging Technologies
Anthony P. Kuzub Anthony P. Kuzub
Senior Systems Designer,
CBC/Radio Canada
Olivier Chambin Olivier Chambin
Audio Broadcast Engineer, AioP and Voice-over-IP
Eurosport Discovery

Video: AES67/ST 2110-30 over WAN

Dealing with professional audio, it’s difficult to escape AES67 particularly as it’s embedded within the SMPTE ST 2110-30 standard. Now, with remote workflows prevalent, moving AES67 over the internet/WAN is needed more and more. This talk brings the good news that it’s certainly possible, but not without some challenges.

Speaking at the SMPTE technical conference, Nicolas Sturmel from Merging Technologies outlines the work being done within the AES SC-02-12M working group to define the best ways of working to enable easy use of AES67 on the WAn. He starts by outlining the fact that AES67 was written to expect short links on a private network that you can completely control which causes problems when using the WAN/internet with long-distance links on which your bandwidth or choice of protocols can be limited.

To start with, Nicolas urges anyone to check they actually need AES67 over the WAN to start with. Only if you need precise timing (for lip-sync for example) with PCM quality and low latencies from 250ms down to as a little as 5 milliseconds do you really need AES67 instead of using other protocols such as ACIP, he explains. The problem being that any ping on the internet, even to something fairly close, can easily take 16 to 40ms for the round trip. This means you’re guaranteed 8ms of delay, but any one packet could be as late as 20ms known as the Packet Delay Variation (PDV).

Link

Not only do we need to find a way to transmit AES67, but also PTP. The Precise Time Protocol has ways of coping for jitter and delay, but these don’t work well on WAN links whether the delay in one direction may be different to the delay for a packet in the other direction. PTP also isn’t built to deal with the higher delay and jitter involved. PTP over WAN can be done and is a way to deliver a service but using a GPS receiver at each location is a much better solution only hampered by cost and one’s ability to see enough of the sky.

The internet can lose packets. Given a few hours, the internet will nearly always lose packets. To get around this problem, Nicolas looks at using FEC whereby you are constantly sending redundant data. FEC can send up to around 25% extra data so that if any is lost, the extra information sent can be leveraged to determine the lost values and reconstruct the stream. Whilst this is a solid approach, computing the FEC adds delay and the extra data being constantly sent adds a fixed uplift on your bandwidth need. For circuits that have very few issues, this can seem wasteful but having a fixed percentage can also be advantageous for circuits where a predictable bitrate is much more important. Nicolas also highlights that RIST, SRT or ST 2022-7 are other methods that can also work well. He talks about these longer in his talk with Andreas Hildrebrand

Watch now!
Speaker

Nicolas Sturmel Nicolas Sturmel
Product Manager, Senior Technologist,
Merging Technologies

Video: RAVENNA AM824 & SMPTE ST 2110-31 Applications



Audio has a long heritage in IP compared to video, so there’s plenty of overlap and there are edge cases abound when working between RAVENNA, AES67 and SMPTE ST 2110-30 and -31. SMPTE’s 2110 suite of standards currently holds two methods of carrying audio including a way of carrying encoded audio such as Dolby AC4 and Dolby E.

RAVENNA Evangelist Andreas Hildebrand is joined by Dolby Labs architect James Cowdrey to discuss the compatibility of -30 and -31 with AES67 and how non-PCM data can be carried in -31 whether that be lightly compressed audio, object audio for immersive experiences or even just pure metadata.

Andreas starts by revising the key differences between AES67 and RAVENNA. The core of AES67 fits neatly within RAVENNA’s capabilities including the transport of up to 24-bit linear PCM with 48 samples per packet and up to 8 channels of 48kHz audio. RAVENNA offers more sample rates, more channels and adds discovery and redundancy with modes such as ‘MADI’ and ‘High performance’ which help constrain and select the relevant parameters.

SMPTE ST 2110-30 is based on AES67 but adds its own constraints such that any -30 stream can be received by an AES67 decoder, however, an AES67 sender needs to be aware of -30’s constraints for it to be correctly decoded by a -30 receiver. Andreas says that all AES67 senders now have this capability.


In contrast to 2110-30, 2110-31 is all about AES3 and the ability of AES3 to carry both linear PCM and non-PCM data. We look at the structure of the AES3 which contains audio blocks each of which has 192 Frames. These frames are split into 2, in the case of stereo, 64 in the case of MADI. Within each of these subframes, we finally find the preamble and the 24-bit data. Andreas explains how this is linked to AM824 and the SDP details needed.

James Cowdery leads the second part of today’s talk first talking about SMPTE ST 337 which details how to send non-PCM audio and data in an AES3 serial digital audio interface. It can carry AC-3, AC-4 for object audio delivering immersive audio experiences, Dolby E and also the metadata standards KLV and Serial ADM.

‘Why use Dolby E?’ asks James. Dolby E has a number of advantages although as bandwidth has become more available, it is increasingly replaced by uncompressed audio. However legacy workflows may now be reliant on IP infrastructure between the receiver and decoder, so it’s important to be able to carry it. Dolby E also packs a whole set of surround sound within a single data stream removing any problems of relative phase and can be carried over MPEG-2 transport streams so it still has plenty of flexibility and uses cases.

Its strength can bring fragility and one way which you can destroy a Dolby E feed is by switching between two videos containing Dolby E in the middle of the data rather than waiting for the gap between packets which is called the guardband. Dolby E needs to be aligned to the video so that you can crossfade and switch between videos without breaking the audio. James makes the point that one reason to use -31 and not -30 to carry Dolby E, or any other non-PCM data, is that -30 assumes that a sample rate converter can be used and so there is usually little control over when an SRC is brought in to use. A sample rate converter, of course, would destroy any non-PCM data.

RAVENNA 824 and 2110-31 gateways will preserver the line position of Dolby data. Can support Dolby E transport can therefore be supported by a vendor without Dolby support. James notes that your Dolby E packets need to be 125 microseconds to achieve packet-level switching without missing a guardband and corrupting data.

Immersive audio requires metadata. sADM is an open specification for metadata interchange, the aim of which is to help interoperability between vendors. sADM metadata can be embedded in SDI, transported uncompressed as SMPTE 302 in MPEG-2 Transport Streams and for 2110, is carried in -31. It’s based on XML description of metadata from the Audio Definition Model and James advises using the GZip compression mode to reduce the bitrate as it can be sent per-frame. An alternative metadata standard is SMPTE ST 336 which is an open format providing a binary payload which makes it a lower-latency method for sending Metadata. These methods of sending metadata made sense in the past, but now, with SMPTE ST 2110 having its own section for metadata essences, we see 2110-41 taking shape to allow data like this to be carried on its own.

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Speakers

James Cowdery James Cowdery
Senior Staff Architect
Dolby Laboratories
Andreas Hildebrand Andreas Hildebrand
RAVENNA Evangelist,
ALC NetworX

Video: Introduction to IPMX

The Broadcast Knowledge has documented over 100 videos and webinars on SMPTE ST 2110. It’s a great suite of standards but it’s not always simple to implement. For smaller systems, many of the complications and nuances don’t occur so a lot of the deeper dives into ST 2110 and its associated specifications such as NMOS from AMWA focus on the work done in large systems in tier-1 broadcasters such as the BBC, tpc and FIS Skiing for SVT.

ProAV, the professional end of the AV market, is a different market. Very few companies have a large AV department if one at all. So the ProAV market needs technologies which are much more ‘plug and play’ particularly those in the events side of the market. To date, the ProAV market has been successful in adopting IP technology with quick deployments by using heavily proprietary solutions like ZeeVee, SDVoE and NDI to name a few. These achieve interoperability by having the same software or hardware in each and every implementation.

IPMX aims to change this by bringing together a mix of standards and open specifications: SMPTE ST 2110, NMOS specs and AES. Any individual or company can gain access and develop a service or product to meet them.

Andreas gives a brief history of IP to date outlining how AES67, ST 2110, ST 2059 and the IS specifications, his point being that the work is not yet done. ProAV has needs beyond, though complementary to, those of broadcast.

AES67 is already the answer to a previous interoperability challenge, explains Andreas, as the world of audio over IP was once a purely federated world of proprietary standards which had no, or limited, interoperability. AES67 defined a way to allow these standards to interoperate and has now become the main way audio is moved in SMPTE 2110 under ST 2110-30 (2110-31 allows for AES3). Andreas explains the basics of 2110, AES, as well as the NMOS specifications. He then shows how they fit together in a layered design.

Andreas brings the talk to a close looking at some of the extensions that are needed, he highlights the ability to be more flexible with the quality-bandwidth-latency trade-off. Some ProAV applications require pixel perfection, but some are dictated by lower bandwidth. The current ecosystem, if you include ST 2110-22’s ability to carry JPEG-XS instead of uncompressed video allows only very coarse control of this. HDMI, naturally, is of great importance for ProAV with so many HDMI interfaces in play but also the wide variety of resolutions and framerates that are found outside of broadcast. Work is ongoing to enable HDCP to be carried, suitably encrypted, in these systems. Finally, there is a plan to specify a way to reduce the highly strict PTP requirements.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Evangelist,
ALC NetworX