Video: Creating Interoperable Hybrid Workflows with RIST

TV isn’t made in one place anymore. Throughout media and entertainment, workflows increasingly involve many third parties and being in the cloud. Content may be king, but getting it from place to place is foundational in our ability to do great work. RIST is a protocol that is able to move video very reliably and flexibly between buildings, into, out of and through the cloud. Leveraging its flexibility, there are many ways to use it. This video helps review where RIST is up to in its development and understand the many ways in which it can be used to solve your workflow problems.

Starting the RIST overview is Ciro Noronha, chair of the RIST Forum. Whilst we have delved in to the detail here before in talks like this from SMPTE and this talk also from Ciro, this is a good refresher on the main points that RIST is published in three parts, known as profiles. First was the Simple Profile which defined the basics, those being that it’s based on RTP and uses an ARQ technology to dynamically request any missing packets in a timely way which doesn’t trip the stream up if there are problems. The Main Profile was published second which includes encryption and authentication. Lastly is the Advanced Profile which will be released later this year.

 

 

Ciro outlines the importance of the Simple Profile. That it guarantees compatibility with RTP-only decoders, albeit without error correction. When you can use the error correction, you’ll benefit from correction even when 50% of the traffic is being lost unlike similar protocols such as SRT. Another useful feature for many is multi-link support allowing you to use RIST over bonded LTE modems as well as using SMPTE ST 2022-7

The Main Profile brings with it support for tunnelling meaning you can set up one connection between two locations and put multiple streams of data through. This is great for simplifying data connectivity because only one port needs to be opened in order to deliver many streams and it doesn’t matter in which direction you establish the tunnel. Once established, the tunnel is bi-directional. The tunnel provides the ability to carry general data such as control data or miscellaneous IT.

Encryption made its debut with the publishing of the Main Profile. RIST can use DTLS which is a version of the famous TLS security used in web sites that runs on UDP rather than TCP. The big advantage of using this is that it brings authentication as well as encryption. This ensures that the endpoint is allowed to receive your stream and is based on the strong encryption we are familiar with and which has been tested and hardened over the years. Certificate distribution can be difficult and disproportionate to the needs of the workflow, so RIST also allows encryption using pre-shared keys.

Handing over now to David Griggs and Tim Baldwin, we discuss the use cases which are enabled by RIST which is already found in encoders, decoders and gateways which are on the market. One use case which is on the rise is satellite replacement. There are many companies that have been using satellite for years and for whom the lack of operational agility hasn’t been a problem. In fact, they’ve also been able to make a business model work for occasional use even though, in a pure sense, satellite isn’t perfectly suited to occasional use satellites. However, with the ability to use C-band closing in many parts of the world, companies have been forced to look elsewhere for their links and RIST is one solution that works well.

David runs through a number of others including primary and secondary distribution, links aggregation, premium sports syndication with the handoff between the host broadcaster and the multiple rights-holding broadcasters being in the cloud and also a workflow for OTT where RIST is used for ingest.

RIST is available as an open source library called libRIST which can be downloaded from videolan and is documented in open specifications TR-06-1 and TR-06-2. LibRIST can be found in gstreamer, Upipe, VLC, Wireshark and FFmpeg.

The video finishes with questions about how RIST compares with SRT. RTMP, CMAF and WebRTC.

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Speakers

Tim Baldwin Tim Baldwin
Head of Product,
Zixi
David Griggs David Griggs
Senior Product Manager, Distribution Platforms
Disney Streaming Services
Ciro Noronha Ciro Noronha
President, RIST Forum
Executive Vice President of Engineering, Cobalt Digital

Video: Tweaking Error Correction Protocol Performance: A libRIST Deep Dive

There’s a false assumption that if you send video with these new error-correcting protocols like RIST or SRT that you just need to send the stream, it’ll get healed and everything will be good. But often people don’t consider what actually happens when things go wrong. To heal the stream, more data needs to be sent. Do you have enough headroom to cope with these resends? And what happens if part of your circuit becomes temporarily saturated, how will the feed cope? The reality is that it could kill it permanently due to re-request storms.

In this video from VidTrans21, Sergio Ammirata from SipRadius talks about how the error correcting protocol within RIST works and how it’s been improved to cope even better in a crisis. Joined by Adi Rozenberg they remind us of the key points of RIST and the libRIST. As a reminder, RIST is one of many protocols which allows the receiver to let the sender know which packets its missed and for them to be resent. For a proper overview of RIST and SRT, have a look at this talk explaining RIST and SRT or the multitude of talks here on The Broadcast Knowledge on RIST or SRT. Today’s video is not so much about why people use RIST, but how to make it performant with difficult circuits.

 
libRIST is an open-source, free, library which implements the RIST specification. The aim of libRIST is to allow companies to easily implement RIST within their own commercial and free programmes. Sergio points out that it’s an active project with over 675 commits in the last year bringing RIST to many platforms including ARM, AWS, Darwin, iOS, windows etc. and is now on version 0.2.0, plus is soon to be in VLC 4.0 and FFmpeg 4.3.

To understand why getting error correction is important, we can look at the effects of a simplistic implementation of the negative acknowledgement error recovery method. When the receiver doesn’t receive a packet it sends back a request for a resend of that packet. The sender will send that and, hopefully, it will be received. Let’s imagine, though, that you’re in a data centre sending to someone on a 100Mbps leased line. If the incoming bitrate of your receiver’s internet connection started getting close to 100Mbps due to the aggregate traffic coming into the site, the receiver may start missing out on occasional packets leading it to ask for more packets from the sender. The sender’s bitrate then increases which reduces the margin available in the incoming circuit resulting in more lost packets. This cycle continues until the line is saturated. It’s important to remember that saturating an incoming link doesn’t mean traffic can’t get out. It’s quite possible there are hundreds of megabits available outgoing so there’s plenty of bandwidth to shout for more and more re-requests. The sender is quite happy to send these re-requests as it’s on a 10Gbe link and has plenty of headroom left. Only by stopping the receiver would you be able to break this positive-feedback loop.

Now, all protocols deliver some form of control over what’s re-requested to try to manage difficult situations. Sergio agrees that other implementations of RIST work well in normal situations with less than 10% packet loss, for example. But where bursts of packet loss exceed 20% or the circuit headroom dips below 20%, Sergio says implementations tend to struggle.

As a lead-up up to the recent improvements made in congestion management, Sergio outlines how libRIST uses internal QOS to maintain a bandwidth cap. It will also monitor the RTT every tenth of a second to help spread retries over time. By checking how the RTT is changing in these extreme conditions, libRIST is able to throw away redundant re-requests leaving more bandwidth for useful requests. The fact that the sender is doing this work means that even if the receiver is on an older version of libRIST or on another implementation, the link can still benefit from the checking the libRIST 0.2.0 is doing. The upshot of all this work is that no longer can libRIST deal with 50% packet loss, it can now deliver an unblemished stream up to just shy of 70% packet loss.

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Speaker

Sergio Ammirata Sergio Ammirata Ph.D.
Chief Scientist,
SipRadius LLC
Adi Rozenberg Adi Rozenberg
CTO & Co-founder,
VideoFlow

Video: I know X, what does WebRTC get me?

WebRTC is now a W3C standard providing sub-second peer-to-peer video and audio streaming with NAT traversal. Widely used for video conferencing, its sub-second latency has also been the focus of video streaming companies such as Millicast and Limelight (to name but two) who aim to deliver this otherwise peer-to-peer technology to thousands or millions of people in under a second enabling interactive video, gamefied streams, auctions and ultra-low-latency sports.

Addressing directly people using other streaming protocols, Pion creator Sean DuBois spoke at SF Video Tech about what WebRTC brings over and above protocols like RTMP, SRT and RIST. At the heart of it, WebRTC, like SRT and RIST, creates a connection over which it can send a variety of data. Whilst we expect media to be sent, actually, file transfer can be easily achieved – let’s not forget the whole of SRT is build upon UDT which is specifically a file delivery utility. Where file transfer can be achieved, so can real-time data & metadata transfer.

Sean quickly summarises WebRTC as a Protocol between (typically) browsers, an peer-to-peer secure connection over which multiple audio & video streams can flow. In common with RIST and other recent protocols, it’s based on many pre-existing
technologies such as SRTP, DTLS, ICE and SDP to deliver signalling, connection management, encryption and communication.

 

 

The list of improvements over RTMP is very long. They’re spelt out concisely in the video so we will highlight just a few here. Importantly, low-latency is key. RTMP was low-latency for its time, but not by today’s standards. Google’s Stadia can boast 125ms video latency for a keypress, explains Sean. DTLS and SRTP are essential for security but are well understood, trusted methods of securing your data. DTLS is pretty much exactly the same as the TLS which secures your bank transfers, just moved into UDP instead of TCP. However, WebRTC can work by exchanging ‘fingerprints’ (DTLS-SRTP) instead of the full trusted certificate infrastructure that underpins TLS on the web. Removing the requirement for certs is a big boost for flexibility and agility as long as you are confident you can exchange fingerprints securely ahead of time.

NAT traversal is also a big boon where, even with both endpoints behind a firewall, endpoints can always find a way to communicate although this does mean that ICE servers are needed to facilitate connectivity. Within broadcasting, however, it’s more likely that you’ll have control of one end so this is less needed. Sean highlights the ability to send multiple quality levels within the same stream using the ‘simulcast’ ability of WebRTC.

Sean then looks at SRT and RIST. Both of these are low-latency streaming protocols which can, both, also provide sub-second streaming for good connections with a relatively low RTT. Sean highlights the lack of SRT and RIST to negotiate the codec in use and their optional security. Being focused more on delivering contribution feeds, they tend to have a more static configuration often created after a programme of testing to ensure the quality will be acceptable to the broadcaster/streaming provider.

To finish, Sean highlights a whole series of interesting, innovative uses of WebRTC from informal group streaming to drones to shared online games to file transfers and more.

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Speaker

Sean DuBois Sean DuBois
Developer, Apple
Creator of Pion WebRTC

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 LearnIPvideo.com’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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Speakers

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