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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Speakers

Alex Vainman Alex Vainman
Senior Staff Engineer,
Mellanox Technologies

Video: Timing Tails & Buffers

Timing and synchronisation have always been a fundamental aspect of TV and as we move to IP, we see that timing is just as important. Whilst there are digital workflows that don’t need to be synchronised against each other, many do such as studio productions. However, as we see in this talk from The Broadcast Bridge’s Tony Orme, IP networks make timing all the more variable and accounting for this is key to success.

To start with Tony looks at the way the OBs, also known as REMIs, are moving to IP and need a timing plane across all of the different parts of production. We see how traditionally synchronisation is needed and the effect of timing problems not only in missed data but also with all essences being sent separately synchronisation problems between them can easily creep in.

When it comes to IP timing itself, Tony explains how PTP is used to record the capture time of the media/essences and distribute through the system. Looking at the data on the wire and the interval between that and the last will show a distribution of, hopefully, a few microseconds variation. This variation gives rise to jitter which is a varying delay in data arrival. The larger the spread, the more difficult it will be to recover data. To examine this more closely, Tony looks at the reasons for and the impacts of congestion, jitter, reordering of data.

Bursting, to make one of these as an example, is a much overlooked issue on networks. While it can occur in many scenarios without any undue problems, microbusting can be a major issue and one that you need to look for to find. This surrounds the issue of how you decide that a data flow is, say, 500Mbps. If you had an encoder which sent data at 1Gbps for 5 minutes and no data for 5 minutes, then over the 10 minute window, the average bitrate would have been 500Mbps. This clearly isn’t a 500Mbps encoder, but how narrow do you need to have your measurement window to be happy it is, indeed, 500Mbps by all reasonable definitions? Do you need to measure it over 1 second, 1 millisecond? Behind microbursting is the tendency of computers to send whatever data they have as quickly as possible; if a computer has a 10Gbe NIC, then it will send at 10Gbps. What video receivers actually need is well spaced packets which always come a set time apart.

Buffers a necessary for IP transmission, in fact within a computer there are many buffers. So using and understanding buffers is very important. Tony takes us through the thought process of considering what buffers are and why we need them. With this groundwork laid, understanding their use and potential problems is easier and well illustrated in this talk. For instance, since there are buffers in many parts of the chain to send data from an application to a NIC and have it arrive at the destination, the best way to maximise the chances of having a deterministic delay in the Tx path is to insert PTP information almost at the point of egress in the NIC rather than in the application itself.

The talk concludes by looking at buffer fill models and the problems that come with streaming using TCP/IP rather then UDP/IP (or RTP). The latter being the most common.

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Speakers

Tony Orme Tony Orme
Editor,
The Broadcast Bridge

Video: 5 PTP Implementation Challenges & Best Practices

PTP is an underlying technology enabling the whole SMPTE 2110 uncompressed ecosystem to work. Using PTP, the Precision Time Protocol, the time a frame of video, audio etc. was captured is recorded and so when decoded can be synchronised with other media recorded around that same time. Though parts of 2110 can function without it, when it comes to bringing media together which need synchronisation, vision mixing for instance, PTP is the way to go.

PTP is actually a standard for time distribution which, like its forerunner NTP, was developed by the IEEE and is a cross-industry standard. Now on version IEEE-1588-2019, it defines not only how to send time onto a network, but also how a receiver can work out what the time actually is. Afterall, if you had a letter in the post telling you the time, you’d know that time – and date for that matter – was old. PTP defines a way of working out how long the letter took to arrive so that you can know the date and time based on the letter and you new-found knowledge of the delivery time.

Knowing the time of day is all very well, but to truly synchronise media, SMPTE ST 2059 is used to interpret PTP for professional media. Video and audio are made from repeating data structures. 2059 relates these repeating data structures back to a common time in the past so that at any time in the future, you can calculate the phase of the signal.

Karl Khun from Tektronix starts by laying out the problems to be solved, such as managing jitter and the precision needed. This leads in into a look at how timestamps are used to make a note of when, separately, video and audio were captured. The network needed to implement PTP, particularly for redundancy and the ability of GPS allowing buildings to be co-timed without being connected.

Troubleshooting PTP will be tricky for many, but learning the IT side of this is only part of the solution. Karl looks at some best practices and tips on faultfinding PPT errors which leads on to a discussion of PTP domains and profiles. An important aspect of PTP is that it is bi-directional. Not only that but it’s much more than a distribution of a signal like the previous black and burst infrastructure. It is a system which needs to be managed and deserves to be monitored. Karl shows how graphs can help show the stability of the network and how RTP/CC errors can show network packet losses/corruptions.

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Speakers

Karl Kuhn Karl J. Khun
Principal Solutions Architect
Telestream/Tekronix

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