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Video: Low Latency Live Streaming At Scale

Posted on 1st September 2020 by Russell Trafford-Jones

Low latency can be a differentiator for a live streaming service, or just a way to ensure you’re not beaten to the punch by social media or broadcast TV. Either way, it’s seen as increasingly important for live streaming to be punctual breaking from the past where latencies of thirty to sixty seconds were not uncommon. As the industry has matured and connectivity has enough capacity for video, simply getting motion on the screen isn’t enough anymore.

Steve Heffernan from MUX takes us through the thinking about how we can deliver low latency video both into the cloud and out to the viewers. He starts by talking about the use cases for sub-second latency – anything with interaction/conversations – and how that’s different from low-latency streaming which is one to many, potentially very large scale distribution. If you’re on a video call with ten people, then you need sub-second latency else the conversation will suffer. But distributing to thousands or millions of people, the sacrifice in potential rebuffering of operating sub-second, isn’t worth it, and usually 3 seconds is perfectly fine.

Steve talks through the low-latency delivery chain starting with the camera and encoder then looking at the contribution protocol. RTMP is still often the only option, but increasingly it’s possible to use WebRTC or SRT, the latter usually being the best for streaming contribution. Once the video has hit the streaming infrastructure, be that in the cloud or otherwise, it’s time to look at how to build the manifest and send the video out. Steve talks us through the options of Low-Latency HLS (LHLS) CMAF DASH and Apple’s LL-HLS. Do note that since the talk, Apple removed the requirement for HTTP/2 push.

The talk finishes off with Steve looking at the players. If you don’t get the players logic right, you can start off much farther behind than necessary. This is becoming less of a problem now as players are starting to ‘bend time’ by speeding up and slowing down to bring their latency within a certain target range. But this only underlines the importance of the quality of your player implementation.

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Speaker

Steve Heffernan
Founder & Head of Product, MUX
Creator of video.js

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: Demystifying Video Delivery Protocols

Posted on 30th July 2020 by Russell Trafford-Jones

Let’s face it, there are a lot of streaming protocols out there both for contribution and distribution. Internet ingest in RTMP is being displaced by RIST and SRT, whilst low-latency players such as CMAF and LL-HLS are vying for position as they try to oust HLS and DASH in existing services streaming to the viewer.

This panel, hosted by Jason Thibeault from the Streaming Video Alliance, talks about all these protocols and attempts to put each in context, both in the broadcast chain and in terms of its features. Two of the main contribution technologies are RIST and SRT which are both UDP-based protocols which implement a method of recovering lost packets whereby packets which are lost are re-requested from the sender. This results in a very high resilience to packet loss – ideal for internet deployments.

First, we hear about SRT from Maxim Sharabayko. He lists some of the 350 members of the SRT Alliance, a group of companies who are delivering SRT in their products and collaborating to ensure interoperability. Maxim explains that, based on the UDT protocol, it’s able to do live streaming for contribution as well as optimised file transfer. He also explains that it’s free for commercial use and can be found on github. SRT has been featured a number of times on The Broadcast Knowledge. For a deeper dive into SRT, have a look at videos such as this one, or the ones under the SRT tag.

Next Kieran Kunhya explains that RIST was a response to an industry request to have a vendor-neutral protocol for reliable delivery over the internet or other dedicated links. Not only does vendor-neutrality help remove reticence for users or vendors to adopt the technology, but interoperability is also a key benefit. Kieran calls out hitless switching across multiple ISPs and cellular. bonding as important features of RIST. For a summary of all of RIST’s features, read this article. For videos with a deeper dive, have a look at the RIST tag here on The Broadcast Knowledge.

Demystifying Video Delivery Protocols from Streaming Video Alliance on Vimeo.

Barry Owen represents WebRTC in this webinar, though Wowza deal with many protocols in their products. WebRTC’s big advantage is sub-second delivery which is not possible with either CMAF or LL-HLS. Whilst it’s heavily used for video conferencing, for which it was invented, there are a number of companies in the streaming space using this for delivery to the user because of it’s almost instantaneous delivery speed. Whilst a perfect rendition of the video isn’t guaranteed, unlike CMAF and LL-HLS, for auctions, gambling and interactive services, latency is always king. For contribution, Barry explains, the flexibility of being able to contribute from a browser can be enough to make this a compelling technology although it does bring with it quality/profile/codec restrictions.

Josh Pressnell and Ali C Begen talk about the protocols which are for delivery to the user. Josh explains how smoothstreaming has excited to leave the ground to DASH, CMAF and HLS. They discuss the lack of a true CENC – Common Encryption – mechanism leading to duplication of assets. Similarly, the discussion moves to the fact that many streaming services have to have duplicate assets due to target device support.

Looking ahead, the panel is buoyed by the promise of QUIC. There is concern that QUIC, the Google-invented protocol for HTTP delivery over UDP, is both under standardisation proceedings in the IETF and is also being modified by Google separately and at the same time. But the prospect of a UDP-style mode and the higher efficiency seems to instil hope across all the participants of the panel.

Watch now to hear all the details!
Speakers

	Ali C. Begen Technical Consultant, Comcast
	Kieran Kunhya Founder & CEO, Open Broadcast Systems Director, RIST Forum
	Barry Owen VP, Solutions Engineering Wowza Media Systems
	Josh Pressnell CTO, Penthera Technologies
	Maxim Sharabayko Senior Software Developer, Haivision
	Moderator: Jason Thibeault Executive Director, Streaming Video Alliance

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