Video: PTP/ST 2059 Best Practices developed from PTP deployments and experiences

PTP is foundational for SMPTE ST 2110 systems. It provides the accurate timing needed to make the most out of almost zero-latency professional video systems. In the strictest sense, some ST 2110 workflows can work without PTP where they’re not combining signals, but for live production, this is almost never the case. This is why a lot of time and effort goes into getting PTP right from the outset because making it work perfectly from the outset gives you the bedrock on which to build your most valuable infrastructure upon.

In this video, Gerard Phillips from Arista, Leigh Whitcomb from Imagine Communications and Telestream’s Mike Waidson join forces to run down their top 15 best practices of building a PTP infrastructure you can rely on.

Gerard kicks off underlining the importance of PTP but with the reassuring message that if you ‘bake it in’ to your underlying network, with PTP-aware equipment that can support the scale you need, you’ll have the timing system you need. Thinking of scale is important as PTP is a bi-directional protocol. That is, it’s not like the black and burst and TLS that it replaces which are simply waterfall signals. Each endpoint needs to speak to a clock so understanding how many devices you’ll be having and where is important to consider. For a look a look at PTP itself, rather than best practices, have a look at this talk free registration required or this video with Meinberg.

 

 

Gerard’s best practices advice continues as he recommends using a routed network meaning having multiple layer 2 networks with layer 3 routing between This reduces the broadcast domain size which, in turn, increases stability and resilience. JT-NM TR-1001 can help to assist in deployments using this network architecture. Gerard next cautions about layer 2 IGMP snoopers and queriers which should exist on every VLAN. As the multicast traffic is flooded to the snooping querier in layer 2, it’s important to consider traffic flows.

When Gerard says PTP should be ‘baked in’, it’s partly boundary clocks he’s referring to. Use them ‘everywhere you can’ is the advice as they bring simplicity to your design and allow for easier debugging. Part of the simplicity they bring is in helping the scalability as they shed load from your GM, taking the brunt of the bi-directional traffic and can reduce load on the endpoints.

It’s long been known that audio devices, for instance, older versions of Dante before v4.2, use version one of PTP which isn’t compatible with SPMTE ST 2059’s requirement to use PTP v2. Gerard says that, if necessary, you should buy a version 1 to version 2 converter from your audio vendor to join the v1 island to your v2 infrastructure. This is linked to best practice point 6; All GMs must have the same time. Mike makes the point that all GMs should be locked to GPS and that if you have multiple sites, they should all have an active, GPS-locked GM even if they do send PTP to each other over a WAN as that is likely to deliver less accurate timing even if it is useful as a backup.

Even if you are using physically separate networks for your PTP and ST 2110 main and backup networks, it’s important to have a link between the two GMs for ST 2022-7 traffic so a link between the two networks just for PTP traffic should be established.

The next 3 points of advice are about the ongoing stability of the network. Firstly, ST 2059-2 specifies the use of TLV messages as part of a mechanism for media notes to generate drop-frame timecode. Whilst this may not be needed day 1, if you have it running and show your PTP system works well with it on, there shouldn’t be any surprises in a couple of years when you need to introduce an end-point that will use it. Similarly, the advice is to give your PTP domain a number which isn’t a SMPTE or AES default for the sole reason that if you ever have a device join your network which hasn’t been fully configured, if it’s still on defaults it will join your PTP domain and could disrupt it. If, part of the configuration of a new endpoint is changing the domain number, the chances of this are notably reduced. One example of a configuration item which could affect the network is ‘ptp role master’ which will stop a boundary clock from taking part in BCMA and prevents unauthorised end-points taking over.

Gerard lays out the ways in which to do ‘proper commissioning’ which is the way you can verify, at the beginning, that your PTP network is working well-meaning you have designed and built your system correctly. Unfortunately, PTP can appear to be working properly when in reality it is not for reasons of design, the way your devices are acting, configuration or simply due to bugs. To account for this, Gerard advocates separate checklists for GM switches and media nodes with a list of items to check…and this will be a long list. Commissioning should include monitoring the PTP traffic, and taking a packet capture, for a couple of days for analysis with test and measurement gear or simply Wireshark.

Leigh finishes up the video talking about verifying functionality during redundancy switches and on power-up. Commissioning is your chance to characterise the behaviour of the system in these transitory states and to observe how equipment attached is affected. His last point before summarising is to implement a PTP monitoring solution to capture the critical parameters and to detect changes in the system. SMPTE RP 2059-15 will define parameters to monitor, with the aim that monitoring across vendors will provide some sort of consistent metrics. Also, a new version of IEEE-1588, version 2.1, will add monitoring features that should aid in actively monitoring the timing in your ST 2110 system.

This Arista white paper contains further detail on many of these best practices.

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Speakers

Gerard Phillips Gerard Phillips
Solutions Engineer,
Arista
Leigh Whitcomb Leigh Whitcomb
Principal Engineer.
Imagine
Michael Waidson Mike Waidson
Application Engineer,
Telestream

Video: Creating Interoperable Hybrid Workflows with RIST

TV isn’t made in one place anymore. Throughout media and entertainment, workflows increasingly involve many third parties and being in the cloud. Content may be king, but getting it from place to place is foundational in our ability to do great work. RIST is a protocol that is able to move video very reliably and flexibly between buildings, into, out of and through the cloud. Leveraging its flexibility, there are many ways to use it. This video helps review where RIST is up to in its development and understand the many ways in which it can be used to solve your workflow problems.

Starting the RIST overview is Ciro Noronha, chair of the RIST Forum. Whilst we have delved in to the detail here before in talks like this from SMPTE and this talk also from Ciro, this is a good refresher on the main points that RIST is published in three parts, known as profiles. First was the Simple Profile which defined the basics, those being that it’s based on RTP and uses an ARQ technology to dynamically request any missing packets in a timely way which doesn’t trip the stream up if there are problems. The Main Profile was published second which includes encryption and authentication. Lastly is the Advanced Profile which will be released later this year.

 

 

Ciro outlines the importance of the Simple Profile. That it guarantees compatibility with RTP-only decoders, albeit without error correction. When you can use the error correction, you’ll benefit from correction even when 50% of the traffic is being lost unlike similar protocols such as SRT. Another useful feature for many is multi-link support allowing you to use RIST over bonded LTE modems as well as using SMPTE ST 2022-7

The Main Profile brings with it support for tunnelling meaning you can set up one connection between two locations and put multiple streams of data through. This is great for simplifying data connectivity because only one port needs to be opened in order to deliver many streams and it doesn’t matter in which direction you establish the tunnel. Once established, the tunnel is bi-directional. The tunnel provides the ability to carry general data such as control data or miscellaneous IT.

Encryption made its debut with the publishing of the Main Profile. RIST can use DTLS which is a version of the famous TLS security used in web sites that runs on UDP rather than TCP. The big advantage of using this is that it brings authentication as well as encryption. This ensures that the endpoint is allowed to receive your stream and is based on the strong encryption we are familiar with and which has been tested and hardened over the years. Certificate distribution can be difficult and disproportionate to the needs of the workflow, so RIST also allows encryption using pre-shared keys.

Handing over now to David Griggs and Tim Baldwin, we discuss the use cases which are enabled by RIST which is already found in encoders, decoders and gateways which are on the market. One use case which is on the rise is satellite replacement. There are many companies that have been using satellite for years and for whom the lack of operational agility hasn’t been a problem. In fact, they’ve also been able to make a business model work for occasional use even though, in a pure sense, satellite isn’t perfectly suited to occasional use satellites. However, with the ability to use C-band closing in many parts of the world, companies have been forced to look elsewhere for their links and RIST is one solution that works well.

David runs through a number of others including primary and secondary distribution, links aggregation, premium sports syndication with the handoff between the host broadcaster and the multiple rights-holding broadcasters being in the cloud and also a workflow for OTT where RIST is used for ingest.

RIST is available as an open source library called libRIST which can be downloaded from videolan and is documented in open specifications TR-06-1 and TR-06-2. LibRIST can be found in gstreamer, Upipe, VLC, Wireshark and FFmpeg.

The video finishes with questions about how RIST compares with SRT. RTMP, CMAF and WebRTC.

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Speakers

Tim Baldwin Tim Baldwin
Head of Product,
Zixi
David Griggs David Griggs
Senior Product Manager, Distribution Platforms
Disney Streaming Services
Ciro Noronha Ciro Noronha
President, RIST Forum
Executive Vice President of Engineering, Cobalt Digital

Video: FOX – Uncompressed live sports in the cloud

Is using uncompressed video in the cloud with just 6 frames of latency to get there and back ready for production? WebRTC manages sub-second streaming in one direction and can even deliver AV1 in real-time. The key to getting down to a 100ms round trip is to move down to millisecond encoding and to use uncompressed video in the cloud. This video shows how it can be done.

Fox has a clear direction to move into the cloud and last year joined AWS to explains how they’ve put their delivery distribution into the cloud remuxing feeds for ATSC transmitters, satellite uplinks, cable headends and encoding for internet delivery, In this video, Fox’s Joel Williams, Evan Statton from AWS explain their work together making this a reality. Joel explains that latency is not a very hot topic for distribution as there are many distribution delays. The focus has been on getting the contribution feeds into playout and MCR monitoring quickly. After all, when people are counting down to an ad break, it needs to roll exactly on zero.

Evan explains the approach AWS has taken to solving this latency problem and starts with considering using SMPTE’s ST 2110 in the cloud. ST 2110 has video flows of at least 1 Gbps, typically and when implemented on-premise is typically built on a dedicated network with very strict timing. Cloud datacentres aren’t like that and Evan demonstrates this showing how across 8 video streams, there are video drops of several seconds which is clearly not acceptable. Amazon, however, has a product called ‘Scalable Reliable Datagram’ which is aimed at moving high bitrate data through their cloud. Using a very small retransmission buffer, it’s able to use multiple paths across the network to deliver uncompressed video in real-time. The retransmission buffer here being very small enables just enough healing to redeliver missing packets within the 16.7ms it takes to deliver a frame of 60fps video.

On top of SRD, AWS have introduced CDI, the Cloud Digital Interface, which is able to describe uncompressed video flows in a way already familiar to software developers. This ‘Audio Video Metadata’ layer handles flows in the same way as 2110, for instance keeping essences separate. Evan says this has helped vendors react favourably to this new technology. For them instead of using UDP, SRD can be used with CDI giving them not only normal video data structures but since SRD is implemented in the Nitro network card, packet processing is hidden from the application itself.

The final piece to the puzzle is keeping the journey into and out of the cloud low-latency. This is done using JPEG XS which has an encoding time of a few milliseconds. Rather than using RIST, for instance, to protect this on the way into the cloud, Fox is testing using ST 2022-7. 2022-7 takes in two identical streams on two network interfaces, typically. This way it should end up with two copies of each packet. Where one gets lost, there is still another available. This gives path redundancy which a single stream will never be able to offer. Overall, the test with Fox’s Arizona-based Technology Center is shown in the video to have only 6 frames of latency for the return trip. Assuming they used a California-based AWS data centre, the ping time may have been as low as two frames. This leaves four frames for 2022-7 buffers, XS encoding and uncompressed processing in the cloud.

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Speakers

Joel Williams Joel Wiliams
VP of Architecutre & Engineering,
Fox Corporation
Evan Statton Evan Statton
Principal Architect, Media & Entertainment,
AWS

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).

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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

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Speaker

Nicolas Sturmel Nicolas Sturmel
Product Manager, Senior Technologist,
Merging Technologies