Video: Introduction to IPMX

The Broadcast Knowledge has documented over 100 videos and webinars on SMPTE ST 2110. It’s a great suite of standards but it’s not always simple to implement. For smaller systems, many of the complications and nuances don’t occur so a lot of the deeper dives into ST 2110 and its associated specifications such as NMOS from AMWA focus on the work done in large systems in tier-1 broadcasters such as the BBC, tpc and FIS Skiing for SVT.

ProAV, the professional end of the AV market, is a different market. Very few companies have a large AV department if one at all. So the ProAV market needs technologies which are much more ‘plug and play’ particularly those in the events side of the market. To date, the ProAV market has been successful in adopting IP technology with quick deployments by using heavily proprietary solutions like ZeeVee, SDVoE and NDI to name a few. These achieve interoperability by having the same software or hardware in each and every implementation.

IPMX aims to change this by bringing together a mix of standards and open specifications: SMPTE ST 2110, NMOS specs and AES. Any individual or company can gain access and develop a service or product to meet them.

Andreas gives a brief history of IP to date outlining how AES67, ST 2110, ST 2059 and the IS specifications, his point being that the work is not yet done. ProAV has needs beyond, though complementary to, those of broadcast.

AES67 is already the answer to a previous interoperability challenge, explains Andreas, as the world of audio over IP was once a purely federated world of proprietary standards which had no, or limited, interoperability. AES67 defined a way to allow these standards to interoperate and has now become the main way audio is moved in SMPTE 2110 under ST 2110-30 (2110-31 allows for AES3). Andreas explains the basics of 2110, AES, as well as the NMOS specifications. He then shows how they fit together in a layered design.

Andreas brings the talk to a close looking at some of the extensions that are needed, he highlights the ability to be more flexible with the quality-bandwidth-latency trade-off. Some ProAV applications require pixel perfection, but some are dictated by lower bandwidth. The current ecosystem, if you include ST 2110-22’s ability to carry JPEG-XS instead of uncompressed video allows only very coarse control of this. HDMI, naturally, is of great importance for ProAV with so many HDMI interfaces in play but also the wide variety of resolutions and framerates that are found outside of broadcast. Work is ongoing to enable HDCP to be carried, suitably encrypted, in these systems. Finally, there is a plan to specify a way to reduce the highly strict PTP requirements.

Watch now!
Speaker

Andreas Hildebrand Andreas Hildebrand
Evangelist,
ALC NetworX

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 Calverly’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.

Webinar: RAVENNA and its Relationship to AES67 and SMPTE ST 2110


This webinar is now available on-demand

This first in a series of webinars, this will have a broad scope covering the history of audio networking in, the development of RAVENNA the the consequent developments of AES67 and ST2110. Whether you’re new to or already familiar with RAVENNA and/or AES67 & ST2110 you’ll benefit from this webinar either as revision or as an excellent starting point for understanding the landscape of Audio-over-IP standards and technologies.

This webinar is presented Andreas Hildebrand who has previously appeared on The Broadcast Knowledge giving insight into The Audio Parts of ST 2110, ST 2110-30 and NMOS IS-08 — Audio Transport and Routing amongst others.

This talk looks at how audio IP works and the benefits of using an IP system. Since the invention of RAVENNA, the AES and SMPTE have moved to using audio over IP so the walk will examine how RAVENNA and SMPTE place the audio data on the network and get that to the decoders. Interoperability between systems is important but can only happen if certain parameters are correct, something that Andreas will mention but will also be a subject for future webinars.

Whether you want to revise the basics from an expert or learn them for the first time, now’s the time to register.

Watch now!

Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX GmbH