Video: AES67 Beyond the LAN

It can be tempting to treat a good quality WAN connection like a LAN. But even if it has a low ping time and doesn’t drop packets, when it comes to professional audio like AES67, you can help but unconver the differences. AES67 was designed for tranmission over short distances meaning extremely low latency and low jitter. However, there are ways to deal with this.

Nicolas Sturmel from Merging Technologies is working as part of the AES SC-02-12M working group which has been defining the best ways of working to enable easy use of AES67 on the WAN wince the summer. The aims of the group are to define what you should expect to work with AES67, how you can improve your network connection and give guidance to manufacturers on further features needed.

WANs come in a number of flavours, a fully controlled WAN like many larger broadacsters have which is fully controlled by them. Other WANs are operated on SLA by third parties which can provide less control but may present a reduced operating cost. The lowest cost is the internet.

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. If you’re contributing into the cloud, then you have an extra layer of complication on top of the WAN. Virtualised computers can offer another place where jitter and uncertain timing can enter.

Link

The good news is that you may not need to use AES67 over the WAN. 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 have a varying round trip time of, say, 16 to 40ms. This means you’re guaranteed 8ms of delay, but any one packet could be as late as 20ms. This variation in timing is 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 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

Nocals finishes by summarising that your solution will need to be sent over unicast IP, possibly in a tunnel, each end locked to a GNSS, high buffers to cope with jitter and, perhaps most importantly, the output of a workflow analysis to find out which tools you need to deploy to meet your actual needs.

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Speaker

Nicolas Sturmel Nicolas Sturmel
Network Specialist,
Merging Technologies

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: AES67 over WAN

Deeply embedded in the audio industry and adopted into SMPTE ST 2110, AES67 workflows surround us. Increasingly our workflows are in multiple locations so moving AES67 on the WAN and the internet is essential. If networks were always perfect, this would be easy but as that’s not the case, this RAVENNA talk examines what the problems are and how to solve them.

Andreas Hildebrand introduces the video with an examination of how the WAN, whether that’s a company’s managed wide area network or the internet at large, is different from a LAN. Typical issues are packet loss, varying latency meaning the packets arrive with jitter, lack of PTP and multicast. With this in mind, Nicolas Sturmel from Merging Technologies takes the reins to examine the solutions.

Nicolas explains the typically EBU Tech 3326 (also known as ACIP) is used for WAN contribution which specifies how a sender and receiver communicate and the codecs to be used. Although PCM is available, many codecs such as AptX are also prescribed for use. Nicolas says that ACIP is great for most applications but if you need low-latency, precise timing and PCM-quality staying AES67 may be the best policy, even over the WAN.

Having identified your AES6-over-WAN workflow, the question is how to pull it off. Nicolas looks at three methods, one is 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. This is can work well but requires sending this extra data constantly therefore putting up your bandwidth. It can also only deal with certain losses requiring them to be of a short duration.

Instead of FEC, you can use RIST, SRT or a similar re-transmission technology. These will actively recover any lost packets and have the benefit that you only transmit more data when you have lost data. Lastly, he mentions SMPTE ST 2022-7 which uses two paths of identical data to cover losses in any one of them. Although this is 100% extra data, the benefit is that it can deal with any type of loss including a complete path failure which neither of the others can do. It is, however possible to combine FEC or RIST with a 2022-7 workflow so you can have two levels of protection.

Timing over the WAN is not ideal as PTP loses accuracy over long-latency links and it assumes symmetry. On the internet, it’s possible to get links where the latency is longer in one direction than the other. An easy, though potentially costly, workaround for distributing PTP over the WAN is to use GPS, GLONASS or similar to synchronise grandmaster clocks at each location.

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Speakers

Nicolas Sturmel Nicolas Sturmel
Product Manager & Senior Technologist
Merging Technologies
Andreas Hildebrand Andreas Hildebrand
RAVENNA Evangelist,
ALCNetworx

Video: Audio networking – ask anything you want!

It’s open season with these AES67 audio-over-Ip experts who are all the questions put to them on working with AES67. Not only was AES67 baked in to SMPTE ST 2110-30, it’s also a standard that brings compatability between Dante and RAVENNA as well as other AoIP technologies.

After a quick summary of what AES66 is, this talk quickly moves into answering these, and other questions:

  • How much bandwidth does stereo AES67 require?
  • Can multicast be used within Ravenna
  • Will there be a slipless switching/2022-7 style function?
  • Should receivers automatically adjust to original stream
  • Is it possible to avoid using PTP in an audio-only system?
  • Cost of PTP-capable switches
  • What’s the difference between Boundary Clocks and Transparent Clocks
  • Can AES67 go over the internet?
  • Tools for spotting problems
  • IPMX for Pro-AV update (See this talk)
  • Is NMOS ‘the answer’ for discovery and configuration?
  • Latency for Ravenna and AES67
  • New advancements in the PTP standard.

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Speakers

Andreas Hildebrand Andreas Hildebrand
Evangelist,
ALC NetworX
Claude Cellier Claude Cellier
President & CEO
Merging Technologies SA
Claudio Becker-Foss
CTO,
DirectOut
Daniel Boldt Daniel Boldt
Head of Software Development,
Meinberg
Terry Holton Terry Holton
Audio subgroup Chairman,
AIMS
Roland Hemming Moderator: Roland Hemming
Audio Consultant
RH Consulting