From WebRTC to RTMP

Continuing our look at the most popular videos of 2020, in common with the previous post on SRT, today we look at replacing RTMP for ingest. This time, WebRTC is demonstrated as an option. With sub-second latency, WebRTC is a compelling replacement for RTMP.

Read what we said about it the first time in the original article, but you’ll see that Nick Chadwick from Mux takes us through the how RTMP works and where the gaps are as it’s phased out. He steps through the alternatives showing how even the low-latency delivery formats don’t fit the bill for contribution and shows how WebRTC can be a sub-second solution.

RIST and SRT saw significant and continued growth in use throughout 2020 as delivery formats and appear to be more commonly used than WebRTC, though that’s not to say that WebRTC isn’t continuing to grow within the broadcast community. SRT and RIST are both designed for contribution in that they actively manage packet loss, allow any codecs to be used and provide for other data to be sent, too. Overall, this tends to give them the edge, particularly for hardware products. But WebRTC’s wide availability on computers can be a bonus in some circumstances. Have a listen and come to your own conclusion.

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Nick Chadwick Nick Chadwick
Software Engineer,

Video: Line by Line Processing of Video on IT Hardware

If the tyranny of frame buffers is let to continue, line-latency I/O is rendered impossible without increasing frame-rate to 60fps or, preferably, beyond. In SDI, hardware was able to process video line-by-line. Now, with uncompressed SDI, is the same possible with IT hardware?

Kieran Kunhya from Open Broadcast Systems explains how he has been able to develop line-latency video I/O with SMPTE 2110, how he’s coupled that with low-latency AVC and HEVC encoding and the challenges his company has had to overcome.

The commercial drivers are fairly well known for reducing the latency. Firstly, for standard 1080i50, typically treated as 25fps, if you have a single frame buffer, you are treated to a 40ms delay. If you need multiple buffers for a workflow, this soon stacks up so whatever the latency of your codec – uncompressed or JPEG XS, for example – the latency will be far above it. In today’s covid world, companies are looking for cutting the latency so people can work remotely. This has only intensified the interest that was already there for the purposes of remote production (REMIs) in having low-latency feeds. In the Covid world, low latency allows full engagement in conversations which is vital for news anchors to conduct interviews as well as they would in person.

IP, itself, has come into its own during recent times where there has been no-one around to move an SDI cable, being able to log in and scale up, or down, SMPTE ST 2110 infrastructure remotely is a major benefit. IT equipment has been shown to be fairly resilient to supply chain disruption during the pandemic, says Kieran, due to the industry being larger and being used to scaling up.

Kieran’s approach to receiving ST 2110 deals in chunks of 5 to 10 lines. This gives you time to process the last few lines whilst you are waiting for the next to arrive. This processing can be de-encapsulation, processing the pixel values to translate to another format or to modify the values to key on graphics.

As the world is focussed on delivering in and out of unusual and residential places, low-bitrate is the name of the game. So Kieran looks at low-latency HEVC/AVC encoding as part of an example workflow which takes in ST 2110 video at the broadcaster and encodes to MPEG to deliver to the home. In the home, the video is likely to be decoded natively on a computer, but Kieran shows an SDI card which can be used to deliver in traditional baseband if necessary.

Kieran talks about the dos and don’ts of encoding and decoding with AVC and HEVC with low latency targetting an end-to-end budget of 100ms. The name of the game is to avoid waiting for whole frames, so refreshing the screen with I-frame information in small slices, is one way of keeping the decoder supplied with fresh information without having to take the full-frame hit of 40ms (for 1080i50). Audio is best sent uncompressed to ensure its latency is lower than that of the video.

Decoding requires carefully handling the slice boundaries, ensuring deblocking i used so there are no artefacts seen. Compressed video is often not PTP locked which does mean that delivery into most ST 2110 infrastructures requires frame synchronising and resampling audio.

Kieran foresees increasing use of 2110 to MPEG Transport Stream back to 2110 workflows during the pandemic and finishes by discussing the tradeoffs in delivering during Covid.

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Kieran Kunhya Kieran Kunhya
CEO & Founder, Open Broadcast Systems

Video: Delivering Quality Video Over IP with RIST

RIST continues to gain traction as a way to deliver video reliably over the internet. Reliable Internet Stream Transport continues to find uses both as part of the on-air signal chain and to enable broadcast workflows by ensuring that any packet loss is mitigated before a decoder gets around to decoding the stream.

In this video, AWS Elemental’s David Griggs explains why AWS use RIST and how RIST works. Introduced by’s Will Simpson who is also the co-chair of the RIST Activity Group at the VSF. Wes starts off by explaining the difference between consumer and business use-cases for video streaming against broadcast workflows. Two of the pertinent differences being one-directional video and needing a fixed delay. David explains that one motivator of broadcasters looking to the internet is the need to replace C-Band satellite links.

RIST’s original goals were to deliver video reliably over the internet but to ensure interoperability between vendors which has been missing to date in the purest sense of the word. Along with this, RIST also aimed to have a low, deterministic latency which is vital to make most broadcast workflows practical. RIST was also designed to be agnostic to the carrier type being internet, satellite or cellular.

Wes outlines how important it is to compensate for packet loss showing that even for what might seem low packet loss situations, you’ll still observe a glitch on the audio or video every twenty minutes. But RIST is more than just a way of ensuring your video/audio arrives without gaps, it. can also support other control signals such as PTZ for cameras, intercom feeds, ad insertion such as SCTE 35, subtitling and timecode. This is one strength which makes RIST ideal for broadcast over using, say RTMP for delivering a live stream.

Wes covers the main and simple profile which are also explained in more detail in this video from SMPTE and this article. One way in which RIST is different from other technologies is GRE tunnelling which allows the carriage of any data type alongside RIST and also allows bundling of RIST streams down a single connecting. This provides a great amount of flexibility to support new workflows as they arise.

David closes the video by explaining why RIST is important to AWS. It allows for a single protocol to support media transfers to, from and within the AWS network. Also important, David explains, is RIST’s standards-based approach. RIST is created out of many standards and RFC with very little bespoke technology. Moreover, the RIST specification is being formally created by the VSF and many VSF specifications have gone on to be standardised by bodies such as SMPTE, ST 2110 being a good example. AWS offer RIST simple profile within MediaConnect with plans to implement the main profile in the near future.

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David Griggs David Griggs
Senior Product Manager, Media Services,
AWS Elemental
Wes Simpson Wes Simpson
RIST AG Co-Chair,
President & Founder,

Video: A video transport protocol for content that matters

What is RIST and why’s it useful? The Reliable Internet Stream Protocol was seeing as strong uptake by broadcasters and other users wanting to use the internet to get their video from A to B over the internet even before the pandemic hit.

Kieran Kunhya from Open Broadcast Systems explains what RIST is trying to do. It comes from a history of expensive links between businesses, with fixed lines or satellite and recognises the increased use of cloud. With cloud computing increasingly forming a key part of many companies’ workflows, media needs to be sent over the internet to get into the workflow. Cloud technology, he explains, allows broadcasters to get away from the traditional on-prem model where systems need to be created to handle peak workload meaning there could be a lot of underutilised equipment.

Whilst the inclination to use the internet seems only too natural given this backdrop, RIST exists to fix the problems that the internet brings with it. It’s not controversial to say that it loses packets and adds jitter to signals. On top of that, using common file transfer technologies like HTTP on TCP leaves you susceptible to drops and variable latency. For broadcasters, it’s also important to know what your latency will be, and know it won’t change. This isn’t something that typical TCP-based technologies offer. On top of solving these problems, RIST also sets out to provide an authenticated, encrypted link.

Ways of doing this have been done before, with Zixi and VideoFlow being two examples that Kieran cites. RIST was created in order to allow interoperability between equipment in a vendor-neutral way. To underline it’s open nature, Kieran shows a table of the IETF RFCs used as part of the protocol.

RIST has two groups of features, those in the ‘Simple Profile’ such as use of RTP, packet loss recovery, bonding and hitless switching. Whereas the ‘Main Profile’ adds on top of that tunnelling (including the ability to choose which direction you set up your connection), encryption, authentication and null packets removal. Both of these are available as published specifications today. A third group of features is being planned under the ‘enhanced profile’ to be released around the beginning of Q2 2021.

Kieran discusses real-world proof points such as a 10-month link which had lost zero packets, though had needed to correct for millions of lost packets. He discusses deployments and moves on to SRT. SRT, Secure Reliable Transport, is a very popular technology which achieves a lot of what RIST does. Although it is an open-source project, it is controlled by one vendor, Haivision. It’s easy to use and has seen very wide deployment and it has done much to educate the market so people understand why they need a protocol such as RIST and SRT so has left a thirst in the market. Kieran sees benefit in RIST having brought together a whole range of industry experts, including Haivision, to develop this protocol and that it already has multipath support, unlike SRT. Furthermore, at 15% packet loss, SRT doesn’t work effectively whereas RIST can achieve full effectiveness with 40% packet loss, as long as you have enough bandwidth for a 200% overhead.

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Kieran Kunhya Kieran Kunhya
Director, RIST Forum
Founder & CEO, Open Broadcast Systems