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!
Speaker

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.

Speaker

Alex Converse Alex Converse
Streaming Video Software Engineer,
Twitch

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!
Speakers

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

Video: SRT Protocol Overview

SRT’s ability to make lossy networks seem like perfect video circuits is increasingly well known, testified to by the SRT Alliance having just surpassed 400 member companies. But this isn’t your average ‘overview’, it dispenses with the technology introductions and goes straight into the detail so is ideal for people who already know the basics and want some deeper knowledge plus a look at the new features to come.

For those wanting an introduction, this article What is SRT? is a good starter which also links to two other intro videos. But today we’re going to join Haivision’s Maxim Sharabayko to look below the surface of SRT.

Maxim starts by introducing the open-source Git repository and the open-source integrations available before heading into the feature matrix. This shows what is and isn’t in SRT. We see that on top of ARQ, it has FEC, encryption, stream multiplexing and, soon, connection bonding. Addressing the major feature areas one by one, we start with connectivity.

SRT has two modes to establish a connection which Maixm shows on handshake diagrams. We can see that establishment need only take 2x round trips so is quick to establish. This allows Maxim to show how firewall traversal is accomplished, though NAT traversal is not yet implemented.

Next on the list of topics is access control whereby we need to ensure that only authorised users can gain access. This is achieved using the Stream ID field within SRT control packets which can contain up to 512 characters meaning it can be used to transfer usernames, passwords (in the form of keys) and requests. Maxim then explains the AES PSK encryption function and discusses the potential implementation of TLS and DTLS.

Content delivery is next under the magnifying glass starting with the structure of SRT packets and the difference between the two types: Data and Control, the former being restricted to only containing payload or FEC data. Maxim covers the positive acknowledgement which is contained with SRT with the range of received packets being acknowledged every 10ms and, where 64 packets come in less than 10ms, a low-overhead acknowledgement being sent for each group of 64 data packets. But of course, it’s the NAK packets which are the most important part of the protocol. Maxim explains they are able to send back one sequence number or a range of lost packets and talks about when they are sent. We see how this then fits into the Timestamp Based Packet Delivery (TSBPD) mechanism which itself is a feature of SRT which delivers packets to the receiver with the same timing as they arrived at the sender. The last thing we look at in the section is a worked example of Too-Late Packet Drop which explains when and why packets are dropped.

ARQ isn’t the only recovery mechanism in SRT, it also provides FEC and, soon, channel bonding. FEC’s can be useful but do have downsides which should be understood. There is a permanent bandwidth overhead, even when the circuit is working well, and a further latency is needed in order to generate the necessary recovery packets. Bonding allows you to stream the same stream over more than one circuit and use data from circuit B to fill in any gaps in circuit A, this technique is used in SMPTE ST 2022-7. Connection bonding, though, can also be used with multiple connections at once and having dynamic balancing across them. Maxim sums up the pros and cons of the different techniques in the table below.

Pros and cons of different packet recovery techniques. Source: Haivision

The talk finishes with a look at stream multiplexing, congestion control and ways in which you can use the SRT statistics which are constantly updated to manage your connectivity.

Watch now!
Speakers

Maxim Sharabayko Maxim Sharabayko
Senior Software Developer,
Havision