Video: Introduction to Precision Time Protocol (PTP)

Posted on by Russell Trafford-Jones

As we’ve seen in so many videos, PTP is fundamental to large-scale SMPTE ST 2110 and pro-audio installations. On The Broadcast Knowledge we’ve looked at a large range of talks on PTP on architecture, scaling, and how it fits in to the broadcast industry. Few of these break down PTP into the fundamentals like today’s article about a video from Albert Mitchell from Cisco.

The key to a PTP network is having one grandmaster clock which can provide time for the rest of the network. In this article, the clocks running in the end devices are called ‘ordinary clocks’. Whilst there are ways to avoid using PTP with uncompressed video such as ST 2110, for live, studio-style productions where you will be bringing them together in a video mixer or similar, keeping these videos effectively zero latency is important and frame syncs on every input of the mixer are discouraged. A grandmaster clock can provide the timing the whole network needs to make this work, usually fed by GPS time.

 

 

SMPTE’s ST 2110 suite has built itself on the timing mechanism of PTP in form of IEEE-1588. SMPTE ST 2059 standards suite provides a method to accommodate all legacy reference and media signals using IEEE-1588 Precision Time Protocol (PTP), delivered over an IP network.


Albert moves on to how this all works. He keeps it simple explaining that there are two measurements needed to get the timing right. You need to know how long it takes to get a message from the grandmaster to the clock and how long it takes to get a message from the clock to the grandmaster. If the grandmaster sends a message with the time in it, it’s trivial for the ordinary clock to look at the time when the message arrived and work out the time it took. It can do the same; an ordinary clock can put the time into a message and send it to a grandmaster. The grandmaster will look at the current time and reply saying how long the transmission delay was. The ordinary clock averages these two measurements and can use the result and the time from the grandmaster to correct its own clock.

Albert finishes by explaining that if there are other switches between the grandmaster and the ordinary clock, those switches should be expected to identify the ‘residence time’ and add this extra delay of simply going through the switch to the time message. Changes in network delay due to congestion or path changes are the reason this timing calculation happens once a second.

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Speakers

Albert Mitchell Albert Mitchell
Technical MArketing Engineer,
Cisco

Video: What is IPMX? – The IPMX Stack

Posted on by Russell Trafford-Jones

“The AV over IP market has really matured [giving us] great quality, low latency and the kind of stability and features that customers are looking for,” says Andrew Starks from Macnica Technology. If that’s the case, why do we need another standard by the name of IPMX? Intended to open up the AV-over-IP market and provide customers with a better deal, Andrew takes us through the motivations of AIMS, AMWA, VSF, SMPTE and the other organisations involved.

IPMX is a set of open standards and specifications which seek to bring a technology platform to the Pro AV industry on which all vendors can interoperate and innovate. Built on SMPTE’s ST 2110 suite of standards and the accompanying NMOS APIs from AMWA, IPMX adds essential capabilities such as HDMI, HDCP and USB support to create a complete and reliable foundation for AV events and installations.

 

 

Whilst there are a number of successful AV initiatives such as SDVoE, these are typically alliances built around a single-vendor hardware solution which is available to vendors in the alliance. This provides interoperability within that ecosystem but, explains Andrew, it prevents wider interoperability between vendors of different alliances. It also makes it hard to any vendor to innovate in the core feature set since that’s delivered from the single source relegating innovation to ‘plumbing’. For the vendors, at best, this means they have to contend with multiple, incompatible product lines and complicated support. Overall this results in a bad end user experience as they operate multiple islands which can have conflicting network requirements, i.e. 10GbE vs 1GbE.

IPMX can be implemented in software as well as hardware using compressed or uncompressed video with a focus on fully featured discovery as this has been identified as being as important as the ability to carry video. Timing has been made flexible such that it can operate with or without PTP which is one of a number of ways that it’s anticipated IPMX will be able to merge in with ST 2110 infrastructures.

Andrew finishes off his talk with a look at the tech stack of IPMX with layer 2 options from 1 to 100GbE connections supported on which RTP and PTP run. SMPTE’s ST 2110 standards feature heavily alongside a new standard for HDCP in 2110, a VSF spec for FEC and new specifications from AMWA for asynchronous control traffic like EDID, Serial, CEC, USB etc. Finally, there are the main APIs such as IS-04, -05 etc. as well as the application layer which uses OAuth2 for authenticating and has an RDS server for discovery. Lastly, there is a look at the JT-NM roadmap to see how the IPMX work will continue to advance throughout this year.

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Speakers

Andrew Starks Andrew Starks
Director of Product Management,
Macnica America’s Inc.

Video: Build The New Generation Of Real Time Streaming Solutions With WebRTC

Posted on by Russell Trafford-Jones

WebRTC continues to live two lives; one of massive daily use in video conferencing in apps from Google, Facebook as well as many others, and one as a side-lined streaming protocol in hte broadcast and streaming industry. WebRTC is now an IETF/W3C standard, is a decade old and is seeing continued work and innovation from Google, other large companies and smaller specialists pushing it forward.

In this extended Streaming Media Connect video with Millicast’s Ryan Jespersen, we explore where WebRTC is up to now, how it can replace RTMP, how real-time AV1 not only shows the innovation within the technology but also enables several use cases and upcoming technologies such as end-to-end encryption for streaming workflows. The video is in sections: product demos, technology discussion and overviews of use cases.

A clear first question is why bother with WebRTC at all. Ryan’s quick to point out that WebRTC is in daily use not only in many of the big video call apps but also in Clubhouse, the high-scale WebRTC-based interactive audio platform. He also establishes that it’s commonly in use on CDNs such as Limelight and Millicast to deliver ultra-low-latency streams to end-users for auctions, gambling and interactive streams, but also as part of broadcast workflows. NFL, for instance, used WebRTC for low-latency monitoring of 122 cameras for the Super Bowl. As far as end-users are concerned, Ryan sees the ‘interactivity’ market as a way, as yet untapped, to release value in many verticals and will be the fastest-growing sector of the streaming industry over the next few years.

 

 

Looking back at Flash, Ryan explains that we came from a point where we had a low-latency protocol in the name of RTMP. Its latency was in the realms of 1 to 3 seconds, it had end-to-end security, encoder control and interactivity. RTMP was displaced due to three main factors, security concerns, rejection of the proprietary nature of the protocol and the move to HLS which provided improved scalability and was enthusiastically adopted by CDNs.

WebRTC, Ryan contends, learns from the mistakes of RTMP. WebRTC has ways to recover lost packets, is content agnostic, has a solution for NAT traversal, is non-proprietary and has no plugins. These latter two points address many of the security concerns of RTMP. Now a standard, the W3C has documented many upcoming use cases for this free, Open Source, technology.

Why, then, do we not see WebRTC much more prevalent in video streaming such as Netflix or Peacock? This is a question that Russell Trafford-Jones discussed in this IBC panel with nanocosmos, M2A and VisualOn. One view from that panel is that sub-second is lower than needed for some services. For instance, a public broadcaster may not wish to deliver online faster than it does over the air. Also, there’s a quality issue to contend with. One strength of WebRTC is that it prioritises latency over quality, always. This is great for face-to-face communication, but tier-1 broadcasters want people to see video in the same quality that left their encoders and if that means waiting for packets to be recovered instead of showing an impaired signal, that’s what they will do. As ever, therefore, this is a business decision that has to pay careful attention to the needs of the viewers, the quality aspirations of the viewers and broadcaster/provider as well as the technical pros and cons of each approach.

Ryan tlks about Real-time AV1 in WebRTC covered also in this talk

Moving on to AV1, Ryan explains that this royalty-free codec has been sped up significantly since the early days when it required thousands of CPUs for real-time encoding. Using AV1 is a boon for WebRTC for two reasons: screen content and scalable video coding. Screen Content Coding is a set of techniques to adapt encoding specifically for screen content meaning computer graphics whether that be in games or just sending a computer desktop. With straighter lines and the possibility for many parts of the screen to be identical or close to identical to other parts, it’s possible to get much better encoding for screen content if you can detect it and optimise for it.

Ryan moves on to AV1’s use in shoring up security. Although a codec and not a security measure in and of itself, AV1’s ability to send multiple resolutions in one stream is a big deal for securing communications. Scalable video coding, SVC, is not a new technology, but AV1 is one of the first mainstream, modern codecs which has it by default. This enables an encoder to encode to, say, sub-SD, SD and HD resolution and send these all at once in one stream. These are not simply 3 encodes squeezed down the same pipe, but they encode that build on top of each other. The sub-HD provides a foundation on which the SD feed provides enhancement information. You need both the sub-SD and SD layer to get SD. Adding on the HD layer to those two gives you that full-resolution HD. By only delivering the extra information needed for HD rather than all the underlying data again, a lot of bitrate can be saved. Importantly, by generating all the encoding at the source, you can encrypt at the source for an end-to-end encrypted workflow and also deliver multiple bitrates. Ryan explains that the move to ABR streaming, whether HLS, DASH or otherwise breaks the end-to-end security model as the need to transcode the media necessitates being able to view it. Using AV1’s SVC is one way around the need for mid-workflow transcoding.

One aspect is missing, though, for modern streaming workflows. If you don’t want to do peer-to-peer networking, some form of traffic manipulation will be needed in your CDN and/or delivery infrastructure. This is why Ryan says that Millicast has proposed that ‘secure frames’ are added to the WebRTC spec. Whilst this talk doesn’t detail their functionality they add a way of encrypting data twice such that the media can be encrypted for end-to-end workflows, but also each hop can be separately encrypted. This provides just enough access to the metadata of the stream for traffic manipulation, but without allowing access to the underlying media.

As the video comes to end, Ryan gives us a glimpse into one other upcoming technology that may be added to WebRTC called WHIP. The RFC explains the intention of WHIP:

The WebRTC-HTTP ingest protocol (WHIP) uses an HTTP POST request to
perform a single shot SDP offer/answer so an ICE/DTLS session can be
established between the encoder/media producer and the broadcasting
ingestion endpoint.

Once the ICE/DTLS session is set up, the media will flow
unidirectionally from the encoder/media producer broadcasting
ingestion endpoint. In order to reduce complexity, no SDP
renegotiation is supported, so no tracks or streams can be added or
removed once the initial SDP O/A over HTTP is completed.

Ryan closes his video with a demonstration of the Millicast platform and looks at how other use cases might be architected such as watch parties.

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Download the slide deck

Speaker

Ryan Jespersen Ryan Jespersen
Head of Sales and Marketing
Millicast

Video: Synchronising Geo-Redundant Origins

Posted on by Russell Trafford-Jones

Synchronised origins in streaming means that a player can switch from one origin to another without any errors or having to restart decoding allowing a much more seamless viewing experience. Adam Ross, speaking from his experience on the Comcast linear video packing team, takes us through the pros and cons of two approaches to synchronisation. This discussion centres around video going into an encoder, transcoder and then packager. This video is either split from a single source which helps keep the video and audio clocks aligned or the clocks are aligned in the encoder or transcoder through communication site A and B.

Keeping segments aligned isn’t too difficult as we just need to keep naming the same and keep them timed together. Whilst not trivial, manifests have many more layers of metadata to synchronised in the form of short-term metadata like content currently present in the manifest and long-term metadata like the dash period. For DASH streams, the Period@ID and Period@Start need to be the same. SegmentTimelines need to have the same start number mapping to the same content. For HLS, variant playlists need to be the same as well as the sequence numbering.

 

 

Adam proposes two methods of doing this. the first is Co-operative Packaging where each site sends metadata between the packagers so that they each make the same, more informed decisions. However, this is complicated to implement and produces a lot of cross-site traffic which can live-point introduce latency. The alternative is a Minimal Synchronisation strategy which relies much more on determinism. Given the same output from the transcoder, the packagers should make the same decisions. Each packager does still need to look at the other’s manifest to ensure it stays in sync and it can resync if not deemed impactful. Overall this second method is much simpler.

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Speaker

Adam Ross Adam Ross
Formerly Software Engineer, Comcast