Low latency Archives – Page 4 of 14 – The Broadcast Knowledge

Video: Low Latency, Real-Time Streaming & WebRTC

Posted on 24th August 2020 by Russell Trafford-Jones

Can any stream be too low-latency? For some matching broadcast latency, is all they need. But for others, particularly for gaming, gambling or more interactive services, sub-second is a must and they are happy to swap out parts of their technology stack to make that happen. WebRTC is often seen as the best choice for anyone wanting to go achieve an almost instant stream. Started by Google in 2011 for video conferencing applications, WebRTC hit a 1.0 release in 2018 and has been adopted by a number of companies catering to the broadcast market.

WebRTC stands out among the plethora of streaming protocols since it is an actual stream of data and not a series of files transferred just in time. Traditionally buffers have been heavily used in streaming because it was so hard to get data to the player when the mainstream internet was starting out in the 90s and as the mobile internet was establishing itself 10 years later. Whilst those buffers are very helpful in dealing with delayed data, they are a big set back in delivering a low-latency stream. With WebRTC, there is very little buffering, so when using the protocol you have to understand that you may not get all your data delivered and if packets are missing glitches will be seen. This is one significant difference since MPEG DASH and HLS will either show you a blank screen or a perfect rendition of the file chunk that was sent thanks to TCP. This is an example of the compromises of going to sub-second latency; there are no second chances to get the packet again. And whilst this compromise may be a great exchange for an auction site or betting service, for other streaming services, it may be better to use CMAF with 3-second latency.

In this talk, Limelight Networks Video Architect Andrew Crowe introduces WebRTC and explains how it can be deployed. He starts by talking about the video codecs it contains. VP9 has recently been added to the options and for a long time, it was a VP8 technology. Andrew explains how the codecs it carries does have a knock-on effect on its compatibility with browsers. UDP is the underlying technology to all low-latency technologies since the bureaucracy of TCP/IP gets in the way of real-time media streams. Andrew also explains how security pervades WebRTC from its use of DTLS (which is like HTTPS/TLS for UDP) to secure RTP and SRTCP.

The last part of the talk discusses the architectures that CDN LimeLight uses to enable large-scale WebRTC streams including the need to get through firewalls. Andrew discusses how some features of the technology suit small-scale events, but can’t be used with thousands of viewers. He also discusses how adaptive bitrate streams can be delivered, although not within WebRTC itself, there is enough information to implement ABR in addition to the standard stream.

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Speakers

Andrew Crowe
Video Architect,
Limelight Networks

Video: A State-of-the-Industry Webinar: Apple’s LL-HLS is finally here

Posted on 27th July 2020 by Russell Trafford-Jones

Even after restrictions are lifted, it’s estimated that overall streaming subscriptions will remain 10% higher than before the pandemic. We’ve known for a long time that streaming is here to stay and viewers want their live streams to arrive quickly and on-par with broadcast TV. There have been a number of attempts at this, the streaming community extended HLS to create LHLS which brought down latency quite a lot without making major changes to the defacto standard.

MPEG’s DASH also has created a standard for low-latency streaming allowing CMAF to be used to get the latency down even further than LHLS. Then Apple, the inventors of the original HLS, announced low-latency HLS (LL-HLS). We’ve looked at all of these previously here on The Broadcast Knowledge. This Online Streaming Primer is a great place to start. If you already know the basics, then there’s no better than Will Law to explain the details.

The big change that’s happened since Will Law’s talk above, is that Apple have revised their original plan. This talk from CTO and Founder of THEOplayer, Pieter-Jan Speelmans, explains how Apple’s modified its approach to low-latency. Starting with a reminder of the latency problem with HLS, Pieter-Jan explains how Apple originally wanted to implement LL-HLS with HTTP/2 push and the problems that caused. This has changed now, and this talk gives us the first glimpse of how well this works.

Pieter-Jan talks about how LL-DASH streams can be repurposed to LL-HLS, explains the protocol overheads and talks about the optimal settings regarding segment and part length. He explains how the segment length plays into both overall latency but also start-up latency and the ability to navigate the ABR ladder without buffering.

There was a lot of frustration initially within the community at the way Apple introduced LL-HLS both because of the way it was approached but also the problems implementing it. Now that the technical issues have been, at least partly, addressed, this is the first of hopefully many talks looking at the reality of the latest version. With an expected ‘GA’ date of September, it’s not long before nearly all Apple devices will be able to receive LL-HLS and using the protocol will need to be part of the playbook of many streaming services.

Watch now to get the full detail

Speaker

Pieter-Jan Speelmans
CTO & Founder
THEOplayer

Video: Bandwidth Prediction for Multi-Bitrate Streaming at Low Latency

Posted on 24th July 2020 by Russell Trafford-Jones

Low latency protocols like CMAF are wreaking havoc with traditional ABR algorithms. We’re having to come up with new ways of assessing if we’re running out of bandwidth. Traditionally, this is done by looking at how long a video chunk takes to download and comparing that with its playback duration. If you’re downloading at the same speed it’s playing, it’s time consider changing stream to a lower-bandwidth one.

As latencies have come down, servers will now start sending data from the beginning of a chunk as it’s being written which means it’s can’t be downloaded any quicker. To learn more about this, look at our article on ISO BMFF and this streaming primer. Since the file can’t be downloaded any quicker, we can’t ascertain if we should move up in bitrate to a better quality stream, so while we can switch down if we start running out of bandwidth, we can’t find a time to go up.

Ali C. Begen and team have been working on a way around this. The problem is that with the newer protocols, you pre-request files which start getting sent when they are ready. As such you don’t actually know the time the chunk starts downloading to you. Whilst you know when it’s finished, you don’t have access, via javascript, to when the file started being sent to you robbing you of a way of determining the download time.

Ali’s algorithm uses the time the last chunk finished downloading in place of the missing timestamp figuring that the new chunk is going to load pretty soon after the old. Now, looking at the data, we see that the gap between one chunk finishing and the next one starting does vary. This lead Ali’s team to move to a sliding window moving average taking the last 3 download durations into consideration. This is assumed to be enough to smooth out some of those variances and provides the data to allow them to predict future bandwidth and make a decision to change bitrate or not. There have been a number of alternative suggestions over the last year or so, all of which perform worse than this technique called ACTE.

In the last section of this talk, Ali explores the entry he was part of into a Twitch-sponsored competition to keep playback latency close to a second in test conditions with varying bitrate. Playback speed is key to much work in low-latency streaming as it’s the best way to trim off a little bit of latency when things are going well and allows you to buy time if you’re waiting for data; the big challenge is doing it without the viewer noticing. The entry used a heuristics and a machine learning approach which worked so well, they were runners up in the contest.

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Speaker

Ali C. Begen,
Technical Consultant, Comcast
Professor, Computer Science, Özyeğin University

Video: RIST for high-end live media workflows

Posted on 17th July 2020 by Russell Trafford-Jones

RIST overcomes the propensity of the internet to lose packets. It makes possible very-high-bandwidth, low-latency contribution over the internet into a studio or directly into the cloud as part of a streaming workflow. Broadcasters have long dreamed of using the increasingly ubiquitous internet to deliver programmes at a lower cost than fixed lines, satellite or microwave. Back in the day, FEC tended to save the day but it had limits meaning the internet was still not so appetising. Now with RIST, the internet is a safe medium for contribution. As ever, two paths are advised!

In this talk, Love Thyresson explains how NetInsight use RIST to deliver high bandwidth contribution for their customers. Love focusses on the lower-tier sports events which would attract an audience, but when the audience is small, the budgets are also small meaning that if you can’t use the internet to get the sports game back to your production centre, the costs – often just on connectivity – are too high to make the programme viable. So whether we are trying to cut costs on a big production or make new programming viable (which might even be the catalyst for a whole new business model or channel), internet contribution is the only way to go.

Love talks about the extension done in RIST to the standard RTP timestamp which, when using high bandwidth streams, quickly runs out of numbers. Expanding it from 16 to 32 bits was the way to allow for more packets to be delivered before having to start the timer from zero again. Indeed, it’s this extra capacity which allows the RIST main profile to deliver JPEG 2000 or JPEG XS. JPEG XS, in particular, is key to modern remote-production workflows. Ingest into the cloud may end up being the most common use for RIST despite the high-value use cases for delivering from events to broadcasters or between broadcasters’ buildings.

After a quick retransmission 101, Love Thyresson closes by looking at the features available now in the simple and main profile of RIST.

For more information, have a look at this article or these videos

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Speakers