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.

Video visit using WebRTC
Virtual video visit using a regular web browser

What is WebRTC and how did it start?

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:

  • The ability to run low-latency real-time communications (RTC) channels over the Internet, automating deep technical issues like codec selection and handling firewalls. It is also fully encrypted.

  • A JavaScript API, enabling these RTC capabilities to be embedded into the web easily – enabling voice and video functions to be “native” parts of a page, without the need for plug-ins.

  • Allowing these capabilities to be used independently of the browser, by creating libraries and SDKs (software development kits) that meant WebRTC voice and video could be embedded into mobile or desktop apps.

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.

Video enabled door intercom with the help of a WebRTC gateway
Video enabled door intercom with the help of a WebRTC gateway

Use-cases and verticals for WebRTC

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).

WebRTC gateway

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.

In terms of industry verticals, some of the main users of WebRTC have been:

Business UCaaS

General business UCaaS users, especially those with desktops.

Contact centres

Contact centres across many verticals, especially those with remote agents (eg outsourced customer service and support).

Social media

Consumers using social media apps, which use WebRTC for streaming, broadcasting or in-app video chat. Dating has also been a major use-case.

Healthcare and telemedicine

Healthcare and telemedicine, especially for “virtual visits”.

Financial services

Financial services applications such as identity-verification via video, or for insurance claims-assessment via video in a mobile app.

Retail and travel sectors

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 telemedicine application
WebRTC application for video consultations with healthcare workers

What has changed during the pandemic?

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.

Some of the ways this has translated to extra WebRTC use include:

Conferencing and collaboration

A massive growth in “pure” WebRTC conferencing/collaboration systems such as Google Meet and Jitsi Meet.

Guests and desktop users

Wide use of WebRTC access for guests and desktop/browser users, gatewayed into traditional UC/UCaaS platforms such as WebEx, avoiding the need for plug-in or separate application downloads.

Customer-case and sales contexts

Extensive use of WebRTC in customer-case and sales contexts, as both call-centre operators and customers are now more likely to be PC-based.

New types of contact-centre interactions

Certain types of contact-centre interactions have seen huge rises in call complexity and duration, for which video may be appropriate, such as retail “remote shopping” for click-and-collect, social care and employment/benefits discussions. The ability to use video inside the web page (for example next to product descriptions or web-forms) rather than in a separate application is a huge benefit.

Healthcare, telecare & telemedicine

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.

Flexible workforces

Growing needs for flexible workforces, including freelancers, consultants and subject-matter experts, who may be brought on-board into companies’ communications systems. This can be important for rapid scaling up/down, or for compliance reasons such as recording, which is much easier where external users are funneled through the company’s platform, rather than “peer to peer” with customers.

Remote viewing & inspection

Wide use of video “remote viewing” and “remote inspection” applications, ranging from house-buying to “remote hands” technicians repairing aircraft engines or installing servers in data-centres.

Social, education & training sessions

Growing use of browser-based video interactions and chat for social, education, training and similar sessions. While these sometimes use commercial conferencing options such as Zoom (which is not based on WebRTC) others have dedicated audio and video interaction built into websites and mobile apps.

Streaming-type applications

More use of WebRTC for streaming-type applications, notably in gaming, where Google’s Stadia platform uses the technology.

IoT Use Cases

Increasing coupling of WebRTC to IoT use-cases – for instance, remote video doorbells and locks, enabling deliveries to be safely made while someone is out of the house, or unable to come to the door.

Interactive screens in public places

Growing use of interactive screens in public places – such as a virtual “reception” at a building or office, with a remote video attendant and perhaps displays/captures of QR codes for permissions.

Consumer communications apps

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”.

Source: Google presentation at the 2020 Krankygeek Event,

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).

Virtual “reception” with a remote video attendant
Virtual reception with a remote video attendant

What’s next for WebRTC?

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.

Some of the things to watch out for in future include:

Simultaneous users

Greater numbers of simultaneous users, for instance for 10s or 100s of people on a video meeting or event.


General improvements in WebRTC performance, for instance on power-efficiency.

Hybrid events

Creation of hybrid events, leading to new opportunities for differentiated communications applications and CPaaS / other cloud-based enablers.

From legacy PBX to cloud

Shifts away from legacy PBX and call-centre platforms towards more flexible cloud.

Health & Social care

Continued emphasis on health and social care applications involving voice, video and integrated sensors.

Awareness around trade-offs

More awareness of the trade-offs between security, privacy, compliance and utility – for instance, how can multiway conversations encrypted end-to-end, but also recorded centrally?


More use of audio/video-processing – most notably background blurring, but also customised versions of noise-suppression suitable for particular applications (eg music lessons vs. a virtual party). Decomposition of some of the internal capabilities of WebRTC, using new standards like WebAssembly, which should enable this more easily.

IoT use-cases

More IoT-centric use-cases, especially as cameras, displays and microphones become ubiquitous in smart-home devices, industrial and smart-building systems, and new forms of interactive displays.

New video codecs for future applications

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.

It is not the only approach to creating video experiences and services, but is now a permanent part of the landscape and a core to a wide range of consumer, business and service-provider innovations.

Enable browser-based calling with the WebRTC gateway

Add WebRTC capabilities to enable browser-based calling with our easy-to-use APIs! Use the gateway functionalities to bridge voice & video communication between WebRTC and SIP​.

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.

Dean Bubley

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