Video: AES67 & SMPTE ST 2110 Timing and Synchronization

Good timing is essential in production for AES67 audio and SMPTE ST 2110. Delivering timing is no longer a matter of delivering a signal throughout your facility, over IP timing is bidirectional and forms a system which should be monitored and managed. Timing distribution has always needed design and architecture, but the detail and understanding needed are much more. At the beginning of this talk, Andreas Hildebrand explains why we need to bother with such complexity, after all, we got along very well for many years without it! Non-IP timing signals are distributed on their own cables as part of their own system. There are some parts of the chain which can get away without timing signals, but when they are needed, they are on a separate cable. With IP, having a separate network for distribution of timing doesn’t make sense so, whether you have an analogue or digital timing signal, that needs to be moving into the IP domain. But how much accuracy in timing to you need? Network devices already widely use NTP which can achieve an accuracy of less than a millisecond. Andreas explains that this isn’t enough for professional audio. At 48Khz, AES samples happen at an accuracy of plus or minus 10 microseconds with 192KHz going down to 2.5 microseconds. As your timing signal has to be less than the accuracy you need, this means we need to achieve nanosecond precision.

Daniel Boldt from timing specialists Meinberg is the focus of this talk explaining how we achieve this nano-second precision. Enter PTP, the Precision Time Protocol. This is a cross-industry standard from the IEEE uses in telcoms, power, finance and in many others wherever a network and its devices need to understand the time. It’s not a static standard, Daniel explains, and it’s just about to see its third revision which, like the last, adds features.

Before finding out about the latest changes, Daniel explains how PTP works in the first place; how is it possible to accurately derive time down to the nanosecond over a network which will have variable propagation times? We see how timestamps are introduced into the network interface controller (NIC) at the last moment allowing the timestamps to be created in hardware which removes some of the variable delays that is typical in software. This happens, Daniel shows, in the switch as well as in the server network cards. This article will refer to either a primary clock or a grand master. Daniel steps us through the messages exchanged between the primary and secondary clock which is the interaction at the heart of the protocol. The key is that after the primary has sent a timestamp, the secondary sends its timestamp to the primary which replies saying the time it received the secondary the reply. The secondary ends up with 4 timestamps that it can combine to determine its offset from the primary’s time and the delay in receiving messages. Applying this information allows it to correct the clock very accurately.

PTP Primary-Secondary Message Exchange.
Source: Meinberg

Most broadcasters would prefer to have more than one grandmaster clock but if there are multiple clocks, how do you choose which to sync from? Timing systems have long used strata whereby clocks are rated based on accuracy, either for internal accuracy & stability or by what they are synched to. This is also true for PTP and is part of the considerations in the ‘Best Master Clock Algorithm’. The BMCA starts by allowing a time source to assess its own accuracy and then search for better options on the network. Clocks announce themselves to the network and by listening to other announcements, a clock can decide if it should become a primary clock if, for instance, it hears no announce messages at all. For devices which should never be a grand primary, you can force them never to decide to become grand masters. This is a requisite for audio devices participating in ST 2110-3x.

Passing PTP around the network takes some care and is most easily done by using switches which understand PTP. These switches either run a ‘boundary clock’ or are ‘transparent clocks’. Daniel explores both of these scenarios explaining how the boundary clock switch is able to run multiple primary and secondary clocks depending on what is connected on each interface. We also see what work the switches have to do behind the scenes to maintain timing precision in transparent mode. In summary, Daniel summaries boundary clocks as being good for hierarchical systems and scales well but requires continuous monitoring whereas transparent clocks are simpler to deploy and require minimal monitoring. The main issue with transparent clocks is that they don’t scale well as all your timing messages are still going back to one main clock which could get overwhelmed.

SMPTE 2022-7 has been a very successful standard as its reliance only on RTP has allowed it to be widely applicable to compressed and uncompressed IP flows. It is often used in 2110 networks, too, where two separate networks are run and brought together at the receiving device. That device, on a packet-by-packet basis, is free to derive its audio/video stream from either network. This requires, however, exactly the same timing on both networks so Daniel looks at an example diagram where this PTP sharing is shown.

PTP’s still evolving and in this next section, Daniel takes us through some of the coming improvements which are also outlined at Meinberg’s blog. These are profile isolation, multi-domain clocks, security improvements and more.

Andreas takes the final section of the webinar to explain how we use PTP in media networks. All receivers will have the same clock which could be derived from GPS removing the need to distribute PTP between sites. 2110 is based on RTP which requires a timestamp to be added to every packet delivered to the network. RTP is a wrapper around IP packets which includes a timestamp which can be derived from the media clock counter.

Andreas looks at how accurate RTP delivery is achieved, dealing with offset values, populating the timestamp from the PTP clock for realties streams and he explains how the playout delay is calculated from the link offset. Finally, he shows the relatively simple process of synchronisation art the playout device. With all the timestamps in the system, synchronising playback of audio, video and metadata using buffers can be achieved fairly easily. Unfortunately, timestamps are easily destroyed by secondary processing (for instance loudness adjustment for an audio stream). Clearly, if this happened, synchronisation at the receiver would be broken. Whilst this will be addressed by out-of-band messaging in future standards, for now, this is managed by a broadcast controller which can take delay information from processing stages and distribute this to receivers.

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Speakers

Daniel Boldt Daniel Boldt
Head of Software Development,
Meinberg
Andreas Hildebrand Andreas Hildebrand
RAVENNA Technology Evangelist,
ALC NetworX

Video: PTP in Virtualized Media Environment

How do we reconcile the tension between the continual move towards virtualisation, microservices and docker-like deployments and the requirements of SMPTE 2110 to have highly precise timing so it can synchronise the video, audio and other essence streams? Virtualisation adds fluidity in to computing so it can share a single set of resources amongst many virtual computers yet PTP, the Precision Time Protocol a successor to NTP, requires close to nano-second precision in its timestamps.

Alex Vainman from Mellanox explains how to make PTP work in these cases and brings along a case study to boot. Starting with a little overview and a glossary, Alex explains the parts of the virtual machine and the environment in which it sits. There’s the physical layer, the hypervisor as well as the virtual machines themselves – each virtual machine being it’s own self-contained computer sitting on a shared computer. Hardware must be shared between, often, many different computers. However some devices aren’t intended to be shared. Take, for instance, a dongle that contains a licence for software. This should clearly be only owned by one computer therefore there is a ‘direct’ mode which means that it is only seen by one computer. Alex goes on to explain the different virtualisation I/O modes which allow devices which can be shared, a printer, storage or CPU for instance need to have access computers may need to wait until they have access to the device to enable sharing. Waiting, of course, is not good for a precision time protocol.

In order to understand the impact that virtualisation might have, Alex details the accuracy and other requirements necessary to have PTP working well enough to support SMPTE 2110 workflows. Although PTP is an IEEE standard, this is just a standard to define how to establish accurate time. It doesn’t help us understand how to phase and bring together media signals without SMPTE ST 2059-1 and -2 which provide the standard of the incoming PTP signal and the way by which we can compare timing and media signals (more info here.) All important is to understand how PTP can actually determine the accurate time given that we know every single message has an unknown propagation delay. By exchanging messages, Alex shows, it is quite practical to measure the delays involved and bring them into the time calculation.

We now have enough information to see why the increased jitter of VM-based systems is going to cause a problem as there are non-deterministic factors such as contention and traffic load to consider as well as factors such as software overhead. Alex takes us through the different options of getting PTP well synchronised in a variety of different VM architectures. The first takes the host clock and ensures that is synchronised to PTP. Using a dedicated PTP library within the VM, this then speaks to the host clock and synchronises the VM OS clock providing very accurate timing. Another, where hardware support in the VM’s hardware for PTP isn’t present, is to have NICs with dedicated PTP support which can directly support the VM OSes maintaining their own PTP clock. The major downside here is that each OS will have to make their own PTP calls creating more load on the PTP system as opposed to the previous architecture whereby the host clock of the VM was the only clock synchronising to the system PTP and therefore there was only ever one set of PTP messages no matter how many VMs were being supported.

To finish off, Alex explains how Windows VMs can be supported – for now through third-party software – and summarises the ways in which we can, in fact, create PTP ecosystems that incorporate virtual machines.

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Download the slides
Speakers

Alex Vainman Alex Vainman
Senior Staff Engineer,
Mellanox Technologies

Video: The Basics of SMPTE ST 2110 in 60 Minutes

SMPTE ST 2110 is a growing suite of standards detailing uncompressed media transport over networks. Now at 8 documents, it’s far more than just ‘video over IP’. This talk looks at the new ways that video can be transported, dealing with PTP timing, creating ‘SDPs’ and is a thorough look at all the documents.

Building on this talk from Ed Calverly which explains how we can use networks to carry uncompressed video, Wes Simpson goes through all the parts of the ST 2110 suite explaining how they work and interoperate as part of the IP Showcase at NAB 2019.

Wes starts by highlighting the new parts of 2110, namely the overview document which gives a high level overview of all the standard docs, the addition of compressed bit-rate video carriage and the recommended practice document for splitting a single video and sending it over multiple links; both of which are detailed later in the talk.

SMPTE ST 2110 is fundamentally different, as highlighted next, in that it splits up all the separate parts of the signal (i.e. video, audio and metadata) so they can be transferred and processed separately. This is a great advantage in terms of reading metadata without having to ingest large amounts of video meaning that the networking and processing requirements are much lighter than they would otherwise be. However, when essences are separated, putting them back together without any synchronisation issues is tricky.

ST 2110-10 deals with timing and knowing which packets of one essence are associated with packets of another essence at any particular point in time. It does this with PTP, which is detailed in IEEE 1588 and also in SMPTE ST 2059-2. Two standards are needed to make this work because the IEEE defined how to derive and carry timing over the network, SMPTE then detailed how to match the PTP times to phases of media. Wes highlights that care needs to be used when using PTP and AES67 as the audio standard requires specific timing parameters.

The next section moves into the video portion of 2110 dealing with video encapsulation on the networks pixel grouping and the headers needed for the packets. Wes then spends some time walking us through calculating the bitrate of a stream. Whilst for most people using a look-up table of standard formats would suffice, understanding how to calculate the throughput helps develop a very good understanding of the way 2110 is carried on the wire as you have to take note not only of the video itself (4:2:2 10 bit, for instance) but also the pixel groupings, UDP, RTP and IP headers.

Timing of packets on the wire isn’t anything new as it is also important for compressed applications, but it is of similar importance to ensure that packets are sent properly paced on wire. This is to say that if you need to send 10 packets, you send them one at a time with equal time between them, not all at once right next to each other. Such ‘micro bursting’ can cause problems not only for the receiver which then needs to use more buffers, but also when mixed with other streams on the network it can affect the efficiency of the routers and switches leading to jitter and possibly dropped packets. 2110-21 sets standards to govern the timing of network pacing for all of the 2110 suite.

Referring back to his warning earlier regarding timing and AES67, Wes now goes into detail on the 2110-30 standard which describes the use of audio for these uncompressed workflows. He explains how the sample rates and packet times relate to the ability to carry multiple audios with some configurations allowing 64 audios in one stream rather than the typical 8.

‘Essences’, rather than media, is a word often heard when talking about 2110. This is an acknowledgement that metadata is just as important as the media described in 2110. It’s sent separately as described by 2110-40. Wes explains the way captions/subtitles, ad triggers, timecode and more can be encapsulated in the stream as ancillary ‘ANC’ packets.

2110-22 is an exciting new addition as this enables the use of compressed video such as VC-2 and JPEG-XS which are ultra low latency codecs allowing the video stream to be reduced by half, a quarter or more. As described in this talk the ability to create workflows on a single IP infrastructure seamlessly moving into and out of compressed video is allowing remote production across countries allowing for equipment to be centralised with people and control surfaces elsewhere.

Noted as ‘forthcoming’ by Wes, but having since been published, is RP 2110-23 which adds back in a feature that was lost when migrating from 2022-6 into 2110 – the ability to send a UHD feed as 4x HD feeds. This can be useful to allow for UHD to be used as a production format but for multiviewers to only need to work in HD mode for monitoring. Wes explains the different modes available. The talk finishes by looking at RTP timestamps and SDPs.

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The slides for this talk are available here
Speakers

Wes Simpson Wes Simpson
President,
Telecom Product Consulting

Video: Broadcast 101 – Audio in an IP Infrastructure

Uncompressed audio has been in the IP game a lot longer than uncompressed video. Because of its long history, it’s had chance to create a fair number of formats ahead of the current standard AES67. Since many people were trying to achieve the same thing, we find that some formats are compatible with AES67 – in part, whilst we that others are not compatible.

To navigate this difficult world of compatibility, Axon CTO Peter Schut continues the Broadcast 101 webinar series with this video recorded this month.

Peter starts by explaining the different audio formats available today including Dante, RAVENNA and others and outlines the ways in which they do and don’t interoperate. After spending a couple of minutes summarising each format individually, including the two SMPTE audio formats -30 and -31, he shows a helpful table comparing the,

Timing is next on the list discussing PTP and the way that SMPTE ST 2059 is used then packet time is covered explaining how the RTP payload fits into the equation. This payload directly affects the duration of audio you can fit into a packet. The duration is important in terms of keeping a low latency and is restricted to either 1ms or 125 microseconds by SMPTE ST 2110-30.

Peter finishes up this webinar talking about some further details about the interoperability problems between the formats.

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Speaker

Peter Schut Peter Schut
CTO,
Axon