Computing Applications Practice

WebRTC: Real-Time Communication for the Open Web Platform

What was once a way to bring audio and video to the Web has expanded into more use cases than we could ever imagine.
  1. Introduction
  2. Architecture and Functionality
  3. Standardization
  4. WebRTC in 2020
  5. Outlook
  6. References
  7. Authors
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In this time of pandemic, the world has turned to Internet-based, real-time communication (RTC) as never before. The number of RTC products has, over the past decade, exploded in large part because of cheaper high-speed network access and more powerful devices, but also because of an open, royalty-free platform called WebRTC.

In fact, over the past year, there has been a 100-fold increase of video minutes received via the WebRTC stack in the anonymous population that has opted into Google Chrome’s statistics. WebRTC can be found in most Internet meeting services, social networks, live-streaming experiences, and even cloud-based gaming products.

WebRTC provides RTC capabilities to browsers and native apps. An open source implementation and tutorials for this platform can be found at It includes audio and video codecs, and signal-processing functions such as bandwidth estimation, noise suppression, and echo cancellation.

This widely deployed communications platform powers audio/video calling, conferencing, and collaboration systems across all major browsers, both on desktop and mobile devices. This has enabled billions of users to interact. WebRTC has vastly expanded and facilitated the ability to create and deploy real-time, interactive services for startups and large-scale companies, and it can be found in commercial products and open source projects alike.

The idea for WebRTC originated in late 2009, more than a year after the launch of Google’s Chrome browser. The Chrome team looked for functionality gaps between the desktop and the Web. While most of the discrepancies were already being addressed by ongoing projects, no solution existed for real-time communications. At the time, only Adobe’s Flash and Netscape’s NPAPI (Netscape Plugin API) provided RTC. Flash’s offering was somewhat low quality and required a server license. Plug-ins are quite tricky for users to install, and few developers have the resources to handle deploying and updating plug-ins that work with three different browsers across several operating systems.

At about this time Google identified a company, Global IP Solutions (aka GIPS), that had the low-level components required for RTC. The GIPS components were licensed by several large customers and were present in products from Google, Skype, AOL, Yahoo!, Cisco, and others. By combining these audio and video components with a JavaScript interface, Google believed it could solve the big “hole” in its Web offerings and spur innovation in the RTC market. If a few lines of JavaScript code were all you needed to add RTC to a Web app—and with no licensing, integration of components, or deep knowledge of RTC required—who knew what could happen?

GIPS was based in Sweden and the U.S. and had engineers in both Stockholm and San Francisco. Luckily for Google, its audio and video Hangouts product was already being worked on in Stockholm, and having the GIPS engineers join in further reinforced the Stockholm office’s strength as an RTC specialist within Google.

When the acquisition was completed in January 2011, the newly formed Chrome WebRTC team focused on integrating the code into Chrome and open sourcing all the key components at From the beginning the plan was to build something open for the Web that would make RTC available for everyone.

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Architecture and Functionality

A WebRTC peer may be a user endpoint (Web browser, native app, and so on) or a server that acts as an intermediary between two or more endpoints. While many WebRTC services rely on a client-server architecture, many others are deployed in a peer-to-peer (P2P, aka connection-less) architecture.

WebRTC is both an API and a protocol. The WebRTC protocol is a set of rules for two WebRTC agents to negotiate bidirectional secure real-time communication. The WebRTC API23 then allows developers to use the WebRTC protocol.

The WebRTC API is specified only for JavaScript. The protocol to establish a connection between two WebRTC peers is a collection of other technologies, which can be split into signaling, connection management, security, and media transfer. These four steps usually happen sequentially. The prior step must be successful for the subsequent one to begin. Each step is actually made up of many other protocols.

As part of the WebRTC standards, many existing technologies that have been around since the early 2000s are combined and adapted for use in browsers and mobile applications.17

Figure 1 provides a high-level overview of the main components and technologies in WebRTC.

Figure 1. WebRTC library componenets.

Android and iOS APIs are implementation-specific and not part of the standard, but they follow the same principles as the JavaScript APIs ( open source implementation18). Audio and video capturing/rendering and network integration are specific to different operating systems.

PeerConnection API. The RTCPeer-Connection API21 is the central part of the WebRTC specification dealing with connecting two applications on different endpoints to communicate using a peer-to-peer protocol. The communication between peers can be video, audio, or arbitrary binary data (later we will discuss clients supporting the DataChannel API).

In order to discover how two peers can connect, both clients need to provide a STUN (session traversal utilities for NAT)9 or a TURN (traversal using relays around NAT) server3 configuration.11 Their role is to provide an ICE (interactive connectivity establishment) candidate to each client, which is then transferred to the remote peer. This transferring of ICE candidates and exchange of other configuration information, such as media capabilities, is commonly called signaling.

Audio/video processing. WebRTC allows you to send and receive streams that include audio and/or video content. Streams can be added and removed at any time during a call; they can be either independent or bundled together. A common collaboration use case for RTC is to capture a computer’s desktop content as a video feed and then include audio/video from the computer’s Webcam and microphone. The WebRTC protocol in general is codec agnostic. The underlying transport has been designed to support any codec format; however, the WebRTC user agent capabilities with regard to media codecs have been subject to standardardization and are well defined.

The media functionality for processing audio and video provides the core of any WebRTC implementation. For audio communications and recording, Opus, G.711μ-law/A-law algorithms, and DTMF (dual-tone multi-frequency) have been defined as mandatory codecs.16 The IETF standardization committees have agreed that WebRTC end-points need to support the VP8 video codec and H.264 Constrained Baseline for processing video.13

Buffers in WebRTC implementations manage variability in packet arrival times, also called jitter, over the connection between peers. The logic of the buffering, managing of retransmission requests, and concealing data packets that have been lost or timed out is at the core of the signal processing work in WebRTC. These algorithms are constantly being developed and have seen major improvements over the past 10 years. The work greatly contributes to obtaining the best possible media quality when communicating over the Internet, especially when peers are connected to networks with different throughput levels and quality.

Security and media transport. WebRTC connections must be encrypted. This is both a core part of the design and part of the standardization. Two existing protocols, DTLS12 (Datagram Transport Layer Security) and SRTP2 (Secure Real-time Transport Protocol), have been adopted for this.

DTLS allows you to negotiate a session and then exchange data securely between two peers. SRTP is designed for exchanging media; it does not have a handshake mechanism and is bootstrapped with the external keys exchanged via DTLS:

  1. DTLS does the handshake over the connection provided by ICE. During the DTLS handshake, both sides offer a certificate.
  2. The SRTP session is created from the keys generated by DTLS.
  3. With these steps completed successfully, SRTP-encrypted media can be exchanged between WebRTC peers.

Media flows between WebRTC peers are by default based on UDP (User Datagram Protocol), meaning that the protocol has to handle unreliable delivery. To achieve the highest possible quality, the stack needs to make trade-offs between latency and quality. Generally speaking, the more latency you are willing to tolerate, the higher-quality video you can expect. For real-time voice communication, ITU-T (International Telecommunication Union-Telecommunications Standardization Sector) has defined the E-model,7 which says that users start being dissatisfied when the mouth-to-ear delay becomes greater than 250 ms.

Congestion control is the mechanism by which WebRTC figures out what quality is achievable, given the latency constraints. Practically speaking, congestion control is being used by a bandwidth estimator adapting the media-encoding parameters for bit rate and video resolutions or audio frame sizes. This lowers the quality but assures that media keeps flowing when users have low or varying bandwidth available.

In the early days of WebRTC, it took, even under good conditions, on average 40 seconds or more to establish a connection and reach video quality of 720 pixels (HD) resolution. By setting aggressive goals, the time was pushed down to 100 ms, thanks to a collaboration with researchers from the Polytechnic University of Bari. This collaboration led to a new congestion-controller design;4 Figure 2 shows the result of launching the congestion-control algorithm.

Figure 2. Ramp-up time to 1Mbps video bit rate.

Data channels. In addition to sending real-time audio and video data, WebRTC allows sending and receiving arbitrary data via so-called data channels. Use cases for data channels range from file transfer, gaming, and IoT (Internet of Things) services to P2P CDNs (content delivery networks). The peer-to-peer data API20 allows the creation of data channels. It extends the RTCPeerConnection API. SCTP (Stream Control Transmission Protocol)15 is used as the underlying protocol to transport data channels. It includes channel multiplexing, reliable delivery with TCP-like retransmission mechanism, congestion avoidance, and flow control.

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At IETF 78 (summer 2010) in Maastricht, Google’s nascent WebRTC team had an informal lunch with engineers from Microsoft, Apple, Mozilla, Skype, Ericsson, and others to gauge the interest in building such an RTC platform for the Web. A quickly organized one-day workshop14 was held with the goal of understanding how such a standard should be written and defined. This led to intense activity in the W3C (World Wide Web Consortium) and the IETF, resulting in the formation of two working groups in May 2011: the IETF’s RTCWeb5 and the W3C’s WebRTC,27 both with participation from across the industry.

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WebRTC in 2020

The adoption of WebRTC has come a long way. Most modern services that use voice or video are either based on the WebRTC protocols or have the ability to use them in addition to the native protocols the service originally deployed with. Cisco’s Webex service, for example, has a WebRTC client that lets people participate in conferences directly from their browsers without downloading additional software. Newer services, such as and Jitsi, have been natively based on WebRTC from the outset. Even when no Web browser is involved, major services use WebRTC for video transmission. For example, WebRTC enables the Amazon Ring product to view security camera and doorbell footage. Increasingly, new IoT products that stream voice and/or video are basing their network stacks on the WebRTC protocols.3

2020 was a year unlike any before. The need for RTC has been highlighted by Covid-19, as people across the globe have found new ways to work, educate, and connect with loved ones via video chat. WebRTC has suddenly become one of the most important sets of technologies allowing Web browsers to make voice, video, and real-time data calls. It has allowed for an ecosystem of interoperable communications apps to flourish: Since the beginning of March 2020, Chrome has seen a 100-fold increase in received video streams via WebRTC, excluding incognito and users opted by default out of sharing stats (see Figure 3).

Figure 3. Video minutes in Chrome over WebRTC in 2020.

These successes would not have been possible without all the supporters that make an open source community. An important element of this success is all the code contributors, testers, bug filers, and corporate partners who helped make this ecosystem a reality.

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Google is a founding member of AO-Media (Alliance for Open Media) and has been active in defining the AV1 video bitstream for the RTC use case. As AV1 has become a standard, the video codec is being integrated into WebRTC. Chrome version 89 is shipping an AV1 software encoder providing AV1-to-Web applications for RTC. AV1 provides another 30%–50% bit-rate savings at the same quality compared with VP9, and is expected to offer another level of bandwidth efficiency and quality for video-calling services. Because of the complexity of the codec, hardware support will be of great importance to make it ubiquitously available. AV1 will be critical in facilitating RTC services to scale further and in allowing for higher-quality video experiences in the future.

WebRTC goes beyond voice and video communication. Emerging gaming, low-latency video streaming, AR/VR (augmented reality/virtual reality), and mixed-reality services are equally benefiting from and demanding low-latency media. For example, WebRTC enables the Stadia gaming service to bring cloud-based, low-latency, high-quality experiences to Web browsers and televisions.

These use cases push the latency barrier, resulting in the need for further transport protocol optimizations. The corresponding standardization effort to cover this need is WebTransport,6,26 focusing on optimizing for super-low-latency client-server media streaming via the QUIC protocol.

As new use cases for WebRTC emerge, the WebRTC standardization is evolving into what is called WebRTC NV (Next Version).25 NV will not be a completely new API but will allow access to the lower-level media pipeline inside PeerConnections. Media will become accessible using the Streams19 and WebCodecs APIs.22 A first step in this direction is the already implemented Insertable Streams API24 that provides the foundation for full E2EE (end-to-end encryption) multiparty conferencing in browsers.8

WebRTC’s reach into mobile devices started through the native (that is, non-Web) integration into mobile social media, messaging, and video calling apps. With emerging 5G networks, video calling will become even more of a commodity.

WebRTC’s open architecture also allows for interesting innovations using machine learning and artificial intelligence to augment call quality and hide the effects of noise10 or network disruptions.1

What started as a way to bring audio and video to the Web has expanded into more use cases than could be imagined—from simple video calling to AR/VR experiences, cloud-based gaming, and massively scalable live streaming services; and from simple point-to-point video chat to multiuser conversations where quality is augmented through advanced machine-learning models. Most importantly, WebRTC is growing from enabling useful experiences to being essential in allowing billions to continue their work and education, and keep vital human contact during a pandemic. The opportunities and impact that lie ahead for WebRTC are intriguing indeed.

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