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: How to Successfully Commission a SMPTE ST 2059/PTP System

PTP is the beating heart behind video- and audio-over-IP installations. As critical as black and burst reference, it pays to get it right. But PTP is a system, not a monolithic signal distributed around the facility. Unlike genlock, it’s a two-way conversation over networked infrastructure and whilst that brings great benefits, it changes how we deal with it. The system should be monitored, both at the ST 2059 layer and network layer. But before we even get to that point, implementation requires care particularly as the industry is still in the early phases of developing tools and best practices for project deployments.

Leigh Whitcomb from Imagine Communications has stepped up to bring us his experiences and best practices as part of the Broadcast Engineering and IT Conference at NAB. This talk assumes an existing level of knowledge of PTP. If you would like to start at the beginning, then please look at this talk from Meinberg and this from Tektronix.

Leigh starts by explaining that, typically, the best architecture is to have a red and a blue network. A grand master would then be on both networks and both would be set to lock to GPS. He explains how do deal with prioritisation and preventing other devices from becoming grand masters. He also explains some of the basic PTP parameter values such as setting the Announcement time outs. Other good design practices he discusses are where to use Boundary Clocks, avoiding PTP Domain numbers of 0 and 127 plus using QoS and DSCP.

As part of the commissioning piece, Leigh goes through some frequently-seen problems such as locking up slowly due to an incorrect Delay Request setting or the Grand Master announce rate being the same as the timeout. To understand when your system isn’t working properly, Leigh makes the point that it’s vital to understand in detail how you expect the system to behave. Use checklists to ensure all parameters and configuration have been applied correctly but also to verify the PTP packets themselves leaving the GM. Leigh then highlights checklists for other parts of the network such as the switches and Media Nodes.

There are a number of tools available for faultfinding and checking compliance. As part of commissioning, the first port of call is the device’s GUI and API which will obviously give most of the parameters needed but often will go further and help with fault finding. WireShark can help verifying the fields in the packets, the timing and message rates. Whilst Meinberg’s Track Hound is a free program which allows you to verify the PTP protocol and Grand Masters. The EBU List project also covers PTP/ST 2059. Helpfully, Leigh talks through how to use Wireshark to verify fields and message rates.

In terms of Testing, Leigh suggests running a packet capture (PCap) for 48 hours after commissioning to verify any issues. He then highlights the need for redundancy testing. This is where understanding how you intend the network to work is important as redundancy testing should also be combined with network testing where you deliberately pull down part of your network and see the GMs change as intended. This changeover will be managed by the Best Master Clock Algorithm (BMCA). When troubleshooting, you should use your monitoring system to help you visualise what’s happening. A good system should enable you to see the devices on the network and their status. Many companies would want to test how successfully the system recovers from a full failure as this will represent the maximum traffic load on the PTP system.

How to watch
1) Click on ‘Add to favourites’
2) Register for free – or log in if you are already part of NAB Express

3) You will then see the video on the left of the screen.

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Speakers

Leigh Whitcomb Leigh Whitcomb
Architect,
Imagine Communications

Video: Introduction To AES67 & SMPTE ST 2110

While standardisation of video and audio over IP is welcome, this does leave us with a plethora of standards numbers to keep track of along with interoperability edge cases to keep track of. Audio-over-IP standard AES67 is part of the SMPTE ST-2110 standards suite and was born largely from RAVENNA which is still in use in it’s own right. It’s with this backdrop that Andreas Hildebrand from ALC NetworX who have been developing RAVENNA for 10 years now, takes the mic to explain how this all fits together. Whilst there are many technologies at play, this webinar focusses on AES67 and 2110.

Andreas explains how AES67 started out of a plan to unite the many proprietary audio-over-IP formats. For instance, synchronisation – like ST 2110 as we’ll see later – was based on PTP. Andreas gives an overview of this synchronisation and then we shows how they looked at each of the OSI layers and defined a technology that could service everyone. RTP, the Real-time Transport Protocol has been in use for a long time for transport of video and audio so made a perfect option for the transport layer. Andreas highlights the important timing information in the headers and how it can be delivered by unicast or IGMP multicast.

As for the audio, standard PCM is the audio of choice here. Andreas details the different format options available such as 24-bit with 8 channels and 48 samples per packet. By varying the format permutations, we can increase the sample rate to 96kHz or modify the number of audio tracks. To signal all of this format information, Session Description Protocol messages are sent which are small text files outlining the format of the upcoming audio. These are defined in RFC 4566. For a deeper introduction to IP basics and these topics, have a look at Ed Calverley’s talk.

The second half of the video is an introduction to ST-2110. A deeper dive can be found elsewhere on the site from Wes Simpson.
Andreas starts from the basis of ST 2022-6 showing how that was an SDI-based format where all the audio, video and metadata were combined together. ST 2110 brings the splitting of media, known as ‘essences’, which allows them to follow separate workflows without requiring lots of de-embedding and embedding processes.

Like most modern standards, ATSC 3.0 is another example, SMPTE ST 2110 is a suite of many standards documents. Andreas takes the time to explain each one and the ones currently being worked on. The first standard is ST 2110-10 which defines the use of PTP for timing and synchronisation. This uses SMPTE ST 2059 to relate PTP time to the phase of media essences.

2110-20 is up next and is the main standard that defines use of uncompressed video with headline features such as being raster/resolution agnostic, colour sampling and more. 2110-21 defines traffic shaping. Andreas takes time to explain why traffic shaping is necessary and what Narrow, Narrow-Linear, Wide mean in terms of packet timing. Finishing the video theme, 2110-22 defines the carriage of mezzanine-compressed video. Intended for compression like TICO and JPEG XS which have light, fast compression, this is the first time that compressed media has entered the 2110 suite.

2110-30 marks the beginning of the audio standards describing how AES67 can be used. As Andreas demonstrates, AES67 has some modes which are not compatible, so he spends time explaining the constraints and how to implement this. For more detail on this topic, check out his previous talk on the matter. 2110-31 introduces AES3 audio which, like in SDI, provides both the ability to have PCM audio, but also non-PCM audio like Dolby E and D.

Finishing up the talk, we hear about 2110-40 which governs transport of ancillary metadata and a look to the standards still being written, 2110-23 Single Video essence over multiple 2110-20 streams, 2110-24 for transport of SD signals and 2110-41 Transport of extensible, dynamic metadata.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX Gmbh.

Video: Benefits of IP Systems for Sporting Venues

As you walk around any exhibitions there seems to be a myriad of ‘benefits’ of IP working, many of which don’t resonate for particular use cases. Only the most extraordinary businesses need all of the benefits, so in this talk, Imagine Communication’s John Mailhot discusses how IP helps sports venues.

John sets the scene by separating out the function of OB trucks and the ‘inside production’ facilities which have a whole host of non-TV production to do including driving scoreboards, displays inside the venue, replays and importantly has to deal with over 250 events a year, not all of which will have an OB truck.

We see that the scale that IP can work at is a great benefit as many signals can fit down one fibre and 2022-7 seamless switching can easily provide full redundancy for every fibre and SFP. This is a level of redundancy which is simply not seen in SDI systems. With stadia being very large, necessitating cable runs of over 500m, the fact that IP needs fewer cables overall is a great benefit.

John shows an example of an Arista switch only 7U in height which provides 144x 100G ports meaning it could support over 4000 inputs and 4000 outputs. Such density is unprecedented and for OB trucks can be a dealbreaker. For sports venues, this can also be a big motivator but also allow more flexibility in distributing the solution rather than relying on a massive central interconnect with a 1100×1100 SDI router in a central CTA.

TV is nothing without audio and the benefits to audio in 2110 are non trivial since with the audio being split off from the video, we are no longer limited to dealing with just 16 channels per video and de-embedding from a video frame any time we want to touch it.

Timing is an interesting benefit. I say this because, whilst PTP can end up being quite complex compared to black and burst, it has some big benefits. First off, it can live in the same cables as your data where as black and burst requires a whole separate cable infrastructure. PTP also allows you to timestamp all essences which helps with lip-sync throughout your workflow.

John leads us through some examples of how this works for different areas finishing by summing up the relevant benefits such as scalability, multi-format, space efficient, and timing amongst others.

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Download the slides
Speakers

John Mailhot John Mailhot
CTO, Networking & Infrastructure,
Imagine Communications