What’s implementing SMPTE ST-2110 like in real life? How would you design your network and what were the problems? In this case study Ammar Latif from Cisco Systems presents the architecture, best practices and lessons learned they gleaned in this live IP broadcast production facility project designed for a major US broadcaster. Based on SMPTE ST-2110 standard, it spanned five studios and two control rooms. The central part of this project was a dual Spine-Leaf IP fabric with bandwidth equivalent of a 10,000 x 10,000 HD SDI router with a fully non-blocking multicast architecture. The routing system was based on Grass Valley Convergent broadcast controller and a Cisco DCNM media controller.
As the project was commissioned in 2018, the AMWA IS-04 and IS-05 specifications providing an inter-operable mechanism for routing media around SMPTE 2110 network were not yet available. Multicast flow subscription was based on a combination of IGMP (Internet Group Management Protocol) and PIM (Protocol Independent Multicast) protocols. While PIM is very efficient and mature, it lacks the ability to use bandwidth as a parameter when setting up a flow path. Ammar explains how Non-Blocking Multicast (NBM) developed by Cisco brings bandwidth awareness to PIM by signalling a type of data (video, audio or metadata).
The talk continues by discussing PTP distribution & monitoring, SMPTE 2022-7 seamless protection switching and remote site production. Ammar also lets us see how the user interfaces on the Cisco DCNM media controller were designed which include a visualisation of multicast flow, network topology and link saturation of ports.
The Broadcast Knowledge exists to help individuals up-skill whatever your starting point. Videos like this are far too rare giving an introduction to a large number of topics. For those starting out or who need to revise a topic, this really hits the mark particularly as there are many new topics.
John Mailhot takes the lead on SMPTE 2110 explaining that it’s built on separate media (essence) flows. He covers how synchronisation is maintained and also gives an overview of the many parts of the SMPTE ST 2110 suite. He talks in more detail about the audio and metadata parts of the standard suite.
Eric Gsell discusses digital archiving and the considerations which come with deciding what formats to use. He explains colour space, the CIE model and the colour spaces we use such as 709, 2100 and P3 before turning to file formats. With the advent of HDR video and displays which can show bright video, Eric takes some time to explain why this could represent a problem for visual health as we don’t fully understand how the displays and the eye interact with this type of material. He finishes off by explaining the different ways of measuring the light output of displays and their standardisation.
Yvonne Thomas talks about the cloud starting by explaining the different between platform as a service (PaaS), infrastructure as a service (IaaS) and similar cloud terms. As cloud migrations are forecast to grow significantly, Yvonne looks at the drivers behind this and the benefits that it can bring when used in the right way. Using the cloud, Yvonne shows, can be an opportunity for improving workflows and adding more feedback and iterative refinement into your products and infrastructure.
Looking at video deployments in the cloud, Yvonne introduces video codecs AV1 and VVC both, in their own way, successors to HEVC/h.265 as well as the two transport protocols SRT and RIST which exist to reliably send video with low latency over lossy networks such as the internet. To learn more about these protocols, check out this popular talk on RIST by Merrick Ackermans and this SRT Overview.
Rounding off the primer is Linda Gedemer from Source Sound VR who introduces immersive audio, measuring sound output (SPL) from speakers and looking at the interesting problem of forward speakers in cinemas. The have long been behind the screen which has meant the screens have to be perforated to let the sound through which interferes with the sound itself. Now that cinema screens are changing to be solid screens, not completely dissimilar to large outdoor video displays, the speakers are having to move but now with them out of the line of sight, how can we keep the sound in the right place for the audience?
This video is a great summary of many of the key challenges in the industry and works well for beginners and those who just need to keep up.
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.
PTP, Precision Time Protocol, underpins the recent uncompressed video and audio over IP standards. It takes over the role of facility-wide synchronisation from black and burst signals. So it’s no surprise that The Broadcast Bridge invited Meinberg to speak at their ‘Real World IP’ event exploring all aspects of video over IP.
David Boldt, head of software engineering at Meinberg, explains how you can accurately transmit time over a network. He summarises the way that PTP accounts for the time taken for messages to move from A to B. David covers different types of clock explaining the often-heard terms ‘boundary clock’ and ‘transparent clock’ exploring their pros and cons.
Unlike black and burst which is a distributed signal, PTP is a system with bi-directional communication which makes redundancy all the more critical and, in some ways, complicated. David talks about different ways to attack the main/reserve problem.
PTP is a cross-industry standard which needs to be interpreted by devices to map the PTP time into an understanding of how the signal should look in order for everything to be in time. SMPTE 2059 does this task which David cover.
PTP-over-WAN: David looks at a case study of delivering PTP over a WAN. Commonly assumed not practical by many, David shows how this ways done without using a GPS antenna at the destination. To finish off the talk, there’s a teaser of the new features coming up in the backwards-compatible PTP Version 2.1 before a Q&A.