WebRTC @5
Harald Alvestrand
Five years ago, this announcement went to the public...@w3.org mailing list:
Today, Google made available WebRTC, an open source software package
for real-time voice and video on the web that we will be integrating
in Chrome. This is our initial contribution to achieve the mission of
making audio and video available in all browsers, through a uniform
standard set of APIs. This initial release will provide the
functionality we envision for WebRTC/RTCWEB, as detailed at
https://sites.google.com/site/webrtc/. Working with the browser
community and working groups like this, our goal is to expand the
available APIs over the next few months for developers to create voice
and video applications on the web.
The underlying components we're releasing are stable and the
interfaces for this initial release are consistent with the
discussions in this working group. We will continue to provide working
implementations for consideration and feedback to collectively ensure
stable standards are finalized. Google is committed to fully
supporting these standards and we look forward to your input in the
coming months.
Harald, speaking for Google.
Today, we can look back and say “yes, we achieved what we set out to do”.
Interactive audio and video is now seen as a commonplace staple of internet applications, both on the Web and in native applications. Many products and services are delivering functionality that crosses seamlessly between Web and native — based on standard APIs that are developed in the open community — and very often powered by the code built in Google’s WebRTC project.
The existence of a common set of accepted standards has fostered adoption across the industry with Firefox, Opera and recently Microsoft Edge supporting interoperable WebRTC. This has led to over 2 billion WebRTC-enabled browsers today, transmitting over an estimated 1 billion audio and video minutes per week in Chrome alone, and over 500 TB of data per week through the WebRTC Data Channel.
At the point of writing over 950 projects and companies base their business on WebRTC with a 70% year-to-year growth rate.
Beyond the web, our native libraries are powering some of the most popular RTC apps, with billions of WebRTC app downloads for Android and iOS, including Facebook Messenger, Hangouts, Snapchat and many more.
And finally, a large set of platforms are available allowing developers to easily create new experiences from vendors such as Twilio, Tokbox, AT&T, Cisco and more.
WebRTC has from the very beginning supported open, royalty-free media codecs as part of the common platform of the Web. 80% of audio communication in WebRTC is based on Opus, and VP9 has been recently launched supporting video calling with 30% less bits than VP8.
The Alliance for Open Media continues to work on improving the royalty-free codecs even further, with a wide selection of industry partners.
Today, WebRTC is a robust suite of protocols, codecs and APIs that have proven themselves in the field. In the next few years we expect to see even more powerful functions that build upon this success.
Continued improvements to audio and video quality
We've agreed on a common set of codecs that WebRTC implementations can use to communicate, but there are still a few interoperability wrinkles to iron out. Looking forward, VP9 and future video codecs such as AV1 promise improved compression efficiency, improved conferencing support via scalability, and better support for low-bandwidth situations.A fresh look at networking
Today's communication is occurring on many different network types, from EDGE to LTE and beyond. WebRTC increasingly faces a diverse set of network conditions, and must be able to adapt accordingly. While we are working on new congestion control algorithms and various ways of configuring media transmission to handle these scenarios, there's an opportunity to rethink and simplify the media protocols that underlie of all of WebRTC to match the demands of today's networks.Beyond the Web: Universal RTC
Five years ago, the majority of communication occurred on the desktop. That's all changed, and WebRTC has evolved to meet the needs of mobile communications apps. Looking forward, there are an increasing number of opportunities in other forms of communication, including communication-focused devices, and communication in VR.
The community has been a vital part of the development effort, and has proven the value of doing development in the open. Feature requests, bug reports and code submissions have proved of enormous value for bringing the project forward and making it useful for everyone.
The value of having this platform available will only grow over time. This is just the beginning.
Harald, still working on delivering WebRTC
Bilal Korir
韩龙
在 2016年6月9日星期四 UTC+8下午5:43:01，Harald Alvestrand写道：
