Video: Public Internet Transport of Live Broadcast Video – SRT, NDI and RIST for Compressed Video

Getting video over the internet and around the cloud has well-established solutions, but not only are they continuing to evolve, they are still new to some. This video looks at workflows that are possible teaming up SRT, RIST and NDI by getting a glimpse into projects that have gone live in 2020. We also get a deeper look at RIST’s features with a Q&A.

This video from SMPTE’s New York section starts with Bryan Nelson from Alpha Video who’s been involved in many cloud-based NDI projects many of which also use SRT to get in and out of the cloud. NDI’s a lightly compressed, low-delay codec suitable for production and works well on 1GbE networks. Not dependant on multicast, it’s a technology that lends itself to cloud-based production where it’s found many uses. Bryan looks at a number of workflows that are also enabled by the Sienna production system which can use many video formats including NDI.

For more information on SRT and RIST, have a look at this SMPTE video outlining how they work and the differences. For a deeper dive into NDI, this SMPTE webinar with VizRT explains how its works and also gives demos of the same software that Bryan uses. To get a feel for how NDI fits in with live production compared to SMPTE’s uncompressed ST 2110, this IBC Panel discussion ‘Where can SMPTE ST 2110 and NDI Co-exist’? explores the topic further.

Bryan’s first example is the 2020 NFL draft is first up which used remote contribution on iPhones streaming using SRT. All streams were aggregated in AWS and converted to NDI feeding NDI multiviewers and routed. These were passed down to on-prem NDI processors which used HP ProLiant servers to output as SDI for handoff to other broadcast workflows. The router could be controlled by soft panels but also hardware panels on-prem. Bryan explores an extension to this idea where multiple cloud domains can be used, with NDI being the handoff between them. In one cloud system, VizRT vision mixing and graphics can be added with multiviewers and other outputs being sent via SRT to remote directors, producers etc. Another cloud system could be controlled by a third party with other processing ahead of then being sent to side and being decoded to SDI on-prem. This can be totally separate to acquisition from SDI & NDI with cameras located elsewhere. SRT & NDI become the mediators between this decentralised production environment.

Bryan finishes off by talking about remote NLE monitoring and various types of MCR monitoring. NLE editing is made easy through NDI integration within Adobe Premiere and Avid Media Composer. It’s possible to bring all of these into a processing engine and move them over the public internet for viewing elsewhere via Apple TV or otherwise.



Ciro Noronha from Cobalt Digital takes the last half of the video to talk about RIST. In addition to the talks mentioned above, Ciro recently gave a talk exploring the many RIST use cases. A good written overview of RIST can be found here.

Ciro looks at the two published profiles that form RIST, the simple and main profile. The simple profile defines RTP interoperability with error correction, using re-requested packets with the option of bonding links. Ciro covers its use of RTCP for maintaining the channel and handling the negative acknowledgements (NACKs) which are based on RFC 4585. RIST can bond multiple links or use 2022-7 seamless switching.

The Main profile builds on the simple profile by adding encryption, authentication and tunnelling. Tunnels allow multiple flows down one connection which simplifies firewall configuration, encryption and allows either end to initiate the bi-directional link. The tunnel can also carry non-RIST traffic for any other purpose. The tunnels are FRE over UDP (RFC 8086). DTLS is used for encryption which is almost identical to TLS used to secure websites. DTLS uses certificates meaning you get to authenticate the other end, not just encrypt the data. Alternatively, you can send a password that avoids the need for certificates when that’s not needed or for one-to-many distribution. Ciro concludes by showing that it can work with up to 50% packet loss and answers many questions in the Q&A.

Watch now!

Byran Nelson Bryan Nelson
Sales Account Executive,
Alpha Video
Ciro Noronha Ciro Noronha
President, RIST Forum
Executive Vice President of Engineering, Cobalt Digital

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.

Watch now!

Sean DuBois Sean DuBois
Developer, Apple
Creator of Pion WebRTC

SRT – How the hot new UDP video protocol actually works under the hood

It’s been a great year at The Broadcast Knowledge growing to over four thousand followers on social media and packing in 250 new articles. So what better time to look back at 2020’s most popular articles as we head into the new year?

It’s fair to say that SRT has seen a lot of interest this year. This was always going to be the case as top-tier broadcasters are now adopting a ‘code as infrastructure’ approach. whereby transmission chains, post-production and live-production workflows are generated via APIs in the cloud, ready for temporary or permanent use. Seen before as the perfect place to put your streaming service, the cloud is increasingly viewed as a viable option for nearly all parts of the production chain.

Getting video in and out of the cloud can be done without SRT, but SRT is a great option as it seamlessly corrects for missing packets which can get lost on the route. How it does this, is the topic of this talk from Alex Converse from Twitch. In the original article on this site, one of the highest-ranking this year, it’s also pitched as an RTMP replacement.

RTMP is still heavily used around the world and like many established technologies, there’s an element of ‘better the devil you know’ mixed in with a reality that much equipment out there will never be updated to do anything else. However, new equipment is being delivered with technologies such as SRT which means that getting from your encoder to the cloud, can now be done with less latency, with better reliability and with a wider choice of codecs than RTMP.

SRT, along with RIST, is helping transform the broadcast industry. To learn more, watch Alex’s video and then look at our other articles and videos on the topic.


Alex Converse Alex Converse
Streaming Video Software Engineer,

Video: Reliable, Live Contribution over the Internet

For so long we’ve been desperate for a cheap and reliable way to contribute programmes into broadcasters, but it’s only in recent years that using the internet for live-to-air streams has been practical for anyone who cares about staying on-air. Add to that an increasing need to contribute live video into, and out of, cloud workflows, it’s easy to see why there’s so much energy going into making the internet a reliable part of the broadcast chain.

This free on-demand webcast co-produced by The Broadcast Knowledge and SMPTE explores the two popular open technologies for contribution over the internet, RIST and SRT. There are many technologies that pre-date those, including Zixi, Dozer and QVidium’s ARQ to name but 3. However, as the talk covers, it’s only in the last couple of years that the proprietary players have come together with other industry members to work on an open and interoperable way of doing this.

Russell Trafford-Jones, from UK video-over-IP specialist Techex, explores this topic starting from why we need anything more than a bit of forward error correction (FEC) moving on to understanding how these technologies apply to networks other than the internet.

This webcast looks at how SRT and RIST work, their differences and similarities. SRT is a well known protocol created and open sourced by Haivision which predates RIST by a number of years. Haivision have done a remarkable job of explaining to the industry the benefits of using the internet for contibution as well as proving that top-tier broadcasters can rely on it.

RIST is more recent on the scene. A group effort from companies including Haivision, Cobalt, Zixi and AWS elemental to name just a few of the main members, with the aim of making a vendor-agnostic, interoperable protocol. Despite, being only 3 years old, Russell explains the 2 specifications they have already delivered which brings them broadly up to feature parity with SRT and are closing in on 100 members.

Delving into the technical detail, Russell looks at how ARQ, the technology fundamental to all these protocols works, how to navigate firewalls, the benefits of GRE tunnels and much more!

The webcast is free to watch with no registration required.

Watch now!

Russell Trafford-Jones Russell Trafford-Jones
Manager, Support & Services, Techex
Director of Education, Emerging Technologies, SMPTE
Editor, The Broadcast Knowledge