This content originally appeared on web.dev and was authored by Huib Kleinhout
The process of defining a web standard is a lengthy process that ensures usefulness, consistency and compatibility across browsers. Today the W3C and IETF mark the completion of perhaps one of the most important standards during the pandemic: WebRTC.
Check out the Real-time communication with WebRTC codelab for a hands-on walkthrough of implementing WebRTC.
History
WebRTC is a platform giving browsers, mobile apps, and desktop apps real-time communication capabilities, typically used for video calling. The platform consists of a comprehensive set of technologies and standards. Google initiated the idea to create WebRTC in 2009, as an alternative to Adobe Flash and desktop applications that couldn't run in the browser. The previous generation of browser-based products were built on top of licensed proprietary technology. Various products were built with this technology, including Hangouts. Google then acquired the companies it had been licensing the technology from and made it available as the open source WebRTC project. This codebase is integrated in Chrome and used by the majority of applications using WebRTC. Together with other browser vendors and industry leaders such as Mozilla, Microsoft, Cisco, and Ericsson, the standardization of WebRTC was kicked off in both the W3C and IETF. In 2013, Mozilla and Google demonstrated video calling between their browsers. Through the evolution of the standard, many architectural discussions had led to implementation differences across browsers and challenged compatibility and interoperability. Most of these disagreements were ultimately settled as the standard became finalized in the past years. The WebRTC specification is now accompanied with a full set of platform tests and tools to address compatibility and browsers have largely adapted their implementations accordingly. This brings an end to a challenging period where web developers had to continuously adopt their services to different browser implementations and specification changes.
Architecture and functionality
The RTCPeerConnection API is
the central part of the WebRTC specification. RTCPeerConnection deals with connecting two
applications on different endpoints to communicate using a peer-to-peer protocol. The PeerConnection
API interacts closely with
getUserMedia for
accessing camera and microphone, and
getDisplayMedia for
capturing screen content. WebRTC allows you to send and receive streams that include audio and/or
video content, as well as arbitrary binary data through the DataChannel. The media functionality for
processing, encoding, and decoding audio and video provides the core of any WebRTC implementation.
WebRTC supports various audio codecs, with Opus being the most used and versatile. WebRTC
implementations are required to support both Google's free-to-use VP8 video codec and H.264 for
processing video. WebRTC connections are always encrypted, which is achieved through two existing
protocols: DTLS and SRTP. WebRTC leans heavily on existing standards and technologies, from video
codecs (VP8,H264), network traversal (ICE), transport (RTP, SCTP), to media description protocols
(SDP). This is tied together in over 50 RFCs.
Use cases: when it's a matter of milliseconds
WebRTC is widely used in time-critical applications such as remote surgery, system monitoring, and remote control of autonomous cars, and voice or video calls built on UDP where buffering is not possible. Nearly all browser-based video callings services from companies such as Google, Facebook, Cisco, RingCentral, and Jitsi use WebRTC. Google Stadia and NVIDIA GeForce NOW use WebRTC to get the stream of gameplay from the cloud to the web browser without perceivable delay.
Pandemic puts focus on video calling performance
Over the past year, WebRTC has seen a 100X increase of usage in Chrome due to increased video calling from within the browser. Recognizing that video calling has become a fundamental part of many people's lives during the pandemic, browser vendors have begun to optimize the technologies that video calling depends on. This was particularly important as resource demanding large meetings and video effects in video meetings became more common when employees and students started to work and study from home. In the past year Chrome has become up to 30% more battery friendly for video calling, with more optimizations to come for heavy usage scenarios. Mozilla, Apple, and Microsoft all have made significant improvements in their implementation of WebRTC through the pandemic, in particular in making sure they adhere to the now formalized standard.
The future of WebRTC
While WebRTC is now completed as a W3C standard, improvements continue. The new video codec AV1 which saves up to 50% of bandwidth is becoming available in WebRTC and web browsers. Continued improvements in the open source code base are expected to further reduce delay and improve the quality of video that can be streamed. WebRTC NV gathers the initiative to create supplementary APIs to enable new use cases. These consist of extensions to existing APIs to give more control over existing functionality such as Scalable Video Coding as well as APIs that give access to lower-level components. The latter gives more flexibility to web developers to innovate by integrating high-performance custom WebAssembly components. With emerging 5G networks and demand for more interactive services, we're expecting to see a continued increase of services building on top of WebRTC in the year to come.
This content originally appeared on web.dev and was authored by Huib Kleinhout
Huib Kleinhout | Sciencx (2021-01-26T00:00:00+00:00) WebRTC is now a W3C and IETF standard. Retrieved from https://www.scien.cx/2021/01/26/webrtc-is-now-a-w3c-and-ietf-standard/
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