Alarm Services
10 December, 2020
For over 15 years, industry analyst and futurist Dean Bubley has tracked the evolution of voice and video communications. 10 years ago, he started running public master-classes on “The Future of Voice” together with his colleague Martin Geddes. That same year a new standard designed to embed realtime voice and video into web browsers emerged. It was WebRTC. In this guest article, Dean walks us through what has happened these past 10 years, and what’s next for WebRTC.
For almost 20 years, it has been clear that the “The Future of Voice”, would evolve “beyond basic phone calls” into a far more diverse set of applications and use-cases. Early corporate softphones, IP contact-centres, audio/video conferencing and collaboration tools were often clunky and had poor user-experience.
VoIP applications or video browser extensions was often hit-and-miss, leading to frustration and ineffective conversations. VoIP signalling, acoustics and image-processing skills were rare and specialised “dark arts”.
Ten years ago in June 2011 a new proposed standard from W3C and IETF emerged – WebRTC. It was designed to embed realtime voice and video communications into web browsers. It brought three key innovations:
Cloud-based providers started to offer these capabilities as a service. WebRTC-based video CPaaS (Communications Platform as a Service) players enabled embedded video-chat for the web, or early forms of video contact-centre.
Over the last ten years, WebRTC has taken a sometimes slow and winding journey, but has already had a huge impact in both consumer and business spheres. It has democratised voice and video capability. It is now much easier to create a new communications application/experience, or add web/app communications into an existing system as a secondary feature.
It is used in billions of devices, as it is supported in every modern browser and other platforms – notably Android, directly in the OS. It has also been “baked in” to thousands of mobile apps with SDKs and libraries. Numerous infrastructure vendors have supplied gateways, tools, testing platforms and many other functions.
That said, WebRTC is not universal. There are plenty of standalone voice applications and soft-phones, as well as separate video applications. Zoom in particular has its own approach and technology, while Microsoft Teams uses WebRTC for browser access, but not its native client.
WebRTC is excellent as a choice for most voice/video developers, but some will either have unique niche requirements, or such deep technical domain expertise or IPR, that they can “roll their own” infrastructure and optimisations.
There are huge numbers of use-cases for WebRTC, across the consumer and enterprise communications landscape. They can be broadly classified into two groups:
These are applications, or user-scenarios which have been designed to use WebRTC throughout. That is, both ends of a connection use WebRTC in the browser or built into a dedicated app. A new standalone video-conference service, or video-chat integrated into a social media app, would typically fit into this category. This may involve a specialist platform provider (CPaaS) or could just be engineered directly by the app developer using WebRTC “libraries” (software components).
This is where WebRTC is used on one end of a connection, but not the other, requiring some sort of gateway or border function. A common example could be a user with a web browser, connecting to an enterprise platform such as a contact centre or cloud-communications UCaaS platform. Often this will involve conversion of signalling to and from the common SIP protocols used in commercial telephony or videoconferencing systems, and may also involve transcoding between different audio/video formats (codecs). A service provider may run the gateway and offer the interconnection functions as a cloud-based service – perhaps as an extra function, if they are also delivering the UCaaS or CCaaS themselves.
Some applications use both models – for instance a conferencing platform which uses SIP between “on-net” users, but also needs to interconnect with the “outside world”. Both of these use-case groups have seen sharp growth during the pandemic, discussed further in the section below.
General business UCaaS users, especially those with desktops.
Retail and travel sectors have used WebRTC for some click-to-call functions, or occasionally “co-browsing” where a sales representative talks a customer through options displayed on an app or web-page.
The telecoms / service provider sector has been fairly slow on WebRTC. In some cases it has formed the basis of niche voice/video applications, or as an extra on-ramp for guest access to hosted telephony and UCaaS services. While various gateways have extended normal “on net” telephony or video functions, the interaction between WebRTC and IMS worlds has been fairly patchy in deployment and uptake.
WebRTC has seen huge changes in usage volumes and diversity of applications. 2020 also saw something of a move away from the use of mobile devices and back towards laptop and desktop PCs, especially for WFH (work from home) interactions – but also consumers preferring large-screen devices more broadly during lockdowns.
Importantly, there has been a huge shift in acceptance of two-way video communications – people are much more comfortable with video in many settings. They have cameras and microphones set up, plugged in and ready to use. They are familiar with how to manage privacy, muting, background filters – and in some cases proper lighting.
There has also been a shift away from room-based conferencing systems, as fewer people are in offices. The same is true for voice-only communications, with few employees using corporate desk-phones, or speaking into dedicated setups in large contact centres.
While in theory these could have been replaced with “cloud native” UCaaS and CCaaS services, in the real world that transition is likely to be quite a slow process – there has been an immediate requirement to repurpose and extend existing “legacy” platforms. Software clients using WebRTC presents an important class of solution.
In other words, people at both/all ends of a conversation are now more likely to be in front of a PC and browser than in 2019. And at the same time, smartphone / tablet users have also expanded their communications use, especially in countries where in-person social events have been closed or limited in scale.
A huge increase in healthcare, telecare and telemedicine usage, across numerous different application scenarios and user contexts. These range from regular video consultations with doctors through to more specialist applications and tools for virtual telecare visits for “shielded” vulnerable patients.
New consumer communications apps and experiences, from home fitness solutions (Peloton was an early user for its exercise-bike classes) to group audio-chat and “collaborative podcasts”.
One notable trend has been a resurgence of some of the original “simple” WebRTC use-cases that were easy to describe 6 or 7 years ago, but which were tricky to implement or which did not fit users’ behaviour and preferences.
While “click to call” buttons have been common in websites for a long time, most users preferred web text-chat windows as they were not familiar with realtime voice/video in that context. That has now changed, and as a result the original vision has been made real – often enabled by third party cloud-based enablers (often linked to broader CPaaS provision).
Many of the trends seen in 2020 will continue in 2021 and beyond, but there will also be continued evolution in both the technology and use-cases. In many ways, WebRTC will mirror broader trends in communications, providing application and developers with an easier way to embed voice/video functions or create new experiences.
Continued work on new video codecs such as VP9 and AV1, allowing better trade-offs between network requirements and processing performance. These may enable future applications such as AR/VR, especially on devices with access to GPUs and hardware accelerators.
In summary – WebRTC has already been on a 10-year journey in democratising voice and video communications. It has enabled a vast array of new applications – and has enabled existing communications services (especially SIP-based) to be extended to wider desktop and mobile audiences, via browsers and smartphone apps.
Dean Bubley is a global outspoken industry analyst and futurist, with huge experience in areas such as CPaaS, WebRTC, 5G and telecom strategy. He is known for his visionary but challenging opinions, his online presence as @disruptivedean, and is regularly seen at live and virtual conferences around the world and quoted in publications such as The Economist, FT and Wall Street Journal.
Mr. Bubley’s clients include many of the world’s leading and most innovative telecom operators. Make sure to follow him on Linkedin and Twitter.
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