Video: Keeping Time with PTP

Different from his talk of the same name we covered last week, Mike Waidson from Telestream explains the fundamentals of PTP joined by Leigh Whitcomb from Imagine Communications and Robert Welch from Arista. Very few PTP talks include a live BCMA quiz plus, with more time than the IP Showcase talks, this is a well-paced, deep look into the basics.

Mike starts by reviewing how the measurement of time has been more and more accurately measured with us now, typically using atomic clocks. In the TV-domain analogue video used signals for B&B which gave frequency information in the subcarrier and allowed frequency locking and to keep in sync with other signals. NTP has allowed computers and routers on IP networks to keep lock allowing sub-millisecond synchronisation over LANs. Now we have IEEE 1588 PTP which harnesses hardware for maximum precision providing sub-microsecond precision.

Traditionally an SPG would create many different synchronising signals, distributed by DAs. With PTP however, the idea is creating a single time signal on to the network (as well as older signals if necessary). Although, the important thing to remember is that PTP both sends and receives data from the endpoints. GPS is made from 31 active satellites of which only 4 are needed for a lock. But other systems such as the Russian GLONASS, the Chinese BAIDU Navigational system or the European Galileo can also be used, sometimes in conjunction with each other to improve locking speed or give resilience.

Mike and his co-hosts give an overview of the standards that make all this possible, starting with the PTP standard itself IEEE 1588-2019 which is added to by SMPTE 2059. The latter is two standards which, together ensure broadcast devices can usefully harness PTP which is a general, cross-industry standard and track all signals back to a single point in time in 1970. Whilst this may seem extreme, the benefit of doing this is that if we know that all possible types of signal were in-phase at this one point in time, we can extrapolate how each signal should be phased now and use that information to synchronise the system. Upcoming to PTP, we hear, are standardised ways to monitor PTP plus additional security around the standard.

The next section looks at the types of Grandmaster and the fact that each clock works in its own domain. Typically, all your system will be in the same domain, but if you have incompatible situations such as older Dante networks or if you want to have a testing environment, you can use domains to separate your equipment. The standard, as defined by SMPTE 2059 is 127.

Mike then looks at the different types of PTP Message types: Announce, Sync & Follow up, Delay Request, Delay Response and Management Messages (broadcast information, drop second, time zone etc.) He then brings some of these up in Wireshark and talks us through the structure and what can be found within.

The most original part of the talk is the live walkthrough of three different scenarios where Leigh and Robert talk through their thinking on which clock will be the grandmaster and for what reason. This comes down to their understanding of the order of precedence of the metrics such as the manually-allotted priority, then the class of clock, clock accuracy and other values. One value worth remembering is that if your clock is locked to GPS it will have a class of 6, but if it then loses lock, it will become 7.

PTP talks are not complete without an explanation of the sync message exchanges needed to actually determine the time (and the relative delays in order to compute it) as well as the secondary clock types, boundary and transparent. Boundary clocks take on much of the two-way traffic in PTP protecting the grandmasters from having to speak directly to all the, potentially, thousands of devices. Transparent switches, simply update the time announcements with the delay for the message to move through the switch. Whilst this is useful in keeping the timing accurate, it provides no protection for the grandmasters.

Before the talk finishes with a Q&A, the team finish by explaining the difference between operating in unicast and multicast, prioritising PTP traffic using the differentiated services protocol and adding redundancy to the PTP system.

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Speakers

Robert Welch
Technical Solultions Lead,
Arista
Leigh Whitcomb Leigh Whitcomb
Principal Engineer.
Imagine
Michael Waidson Mike Waidson
Application Engineer,
Telestream

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: 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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Speakers

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